aboutsummaryrefslogtreecommitdiff
diff options
context:
space:
mode:
Diffstat
-rw-r--r--Changelog7
-rw-r--r--Makefile.target43
-rw-r--r--audio/alsaaudio.c926
-rw-r--r--audio/audio.c1850
-rw-r--r--audio/audio.h90
-rw-r--r--audio/audio_int.h310
-rw-r--r--audio/audio_template.h401
-rw-r--r--audio/coreaudio.c513
-rw-r--r--audio/dsound_template.h298
-rw-r--r--audio/dsoundaudio.c1082
-rw-r--r--audio/fmodaudio.c526
-rw-r--r--audio/mixeng.c231
-rw-r--r--audio/mixeng.h26
-rw-r--r--audio/mixeng_template.h136
-rw-r--r--audio/noaudio.c164
-rw-r--r--audio/ossaudio.c588
-rw-r--r--audio/rate_template.h111
-rw-r--r--audio/sdlaudio.c327
-rw-r--r--audio/sys-queue.h241
-rw-r--r--audio/wavaudio.c160
-rwxr-xr-xconfigure74
-rw-r--r--hw/adlib.c285
-rw-r--r--hw/es1370.c1007
-rw-r--r--hw/pc.c6
-rw-r--r--hw/sb16.c266
-rw-r--r--qemu-doc.texi19
-rw-r--r--vl.c109
-rw-r--r--vl.h4
28 files changed, 8176 insertions, 1624 deletions
diff --git a/Changelog b/Changelog
index 3ba685bc21..ea7806544a 100644
--- a/Changelog
+++ b/Changelog
@@ -1,3 +1,10 @@
+version 0.7.3:
+
+ - Mac OS X cocoa improvements (Mike Kronenberg)
+ - DirectSound driver (malc)
+ - new audio options: '-soundhw' and 'audio-help' (malc)
+ - ES1370 PCI audio device (malc)
+
version 0.7.2:
- x86_64 fixes (Win2000 and Linux 2.6 boot in 32 bit)
diff --git a/Makefile.target b/Makefile.target
index 1ca4e3a4c4..b1773f4ddb 100644
--- a/Makefile.target
+++ b/Makefile.target
@@ -262,7 +262,7 @@ endif
VL_OBJS=vl.o osdep.o block.o readline.o monitor.o pci.o console.o
VL_OBJS+=block-cow.o block-qcow.o aes.o block-vmdk.o block-cloop.o block-dmg.o block-bochs.o block-vpc.o block-vvfat.o
-SOUND_HW = sb16.o
+SOUND_HW = sb16.o es1370.o
AUDIODRV = audio.o noaudio.o wavaudio.o
ifdef CONFIG_SDL
AUDIODRV += sdlaudio.o
@@ -270,29 +270,38 @@ endif
ifdef CONFIG_OSS
AUDIODRV += ossaudio.o
endif
-
-pc.o: DEFINES := -DUSE_SB16 $(DEFINES)
-
-ifdef CONFIG_ADLIB
-SOUND_HW += fmopl.o adlib.o
+ifdef CONFIG_COREAUDIO
+AUDIODRV += coreaudio.o
+endif
+ifdef CONFIG_ALSA
+AUDIODRV += alsaaudio.o
+LIBS += -lasound
+endif
+ifdef CONFIG_DSOUND
+AUDIODRV += dsoundaudio.o
+LIBS += -lole32 -ldxguid
endif
-
ifdef CONFIG_FMOD
AUDIODRV += fmodaudio.o
audio.o fmodaudio.o: DEFINES := -I$(CONFIG_FMOD_INC) $(DEFINES)
LIBS += $(CONFIG_FMOD_LIB)
endif
+ifdef CONFIG_ADLIB
+SOUND_HW += fmopl.o adlib.o
+endif
ifeq ($(TARGET_BASE_ARCH), i386)
# Hardware support
VL_OBJS+= ide.o ne2000.o pckbd.o vga.o $(SOUND_HW) dma.o $(AUDIODRV)
VL_OBJS+= fdc.o mc146818rtc.o serial.o i8259.o i8254.o pc.o
VL_OBJS+= cirrus_vga.o mixeng.o apic.o parallel.o
+DEFINES += -DHAS_AUDIO
endif
ifeq ($(TARGET_BASE_ARCH), ppc)
VL_OBJS+= ppc.o ide.o ne2000.o pckbd.o vga.o $(SOUND_HW) dma.o $(AUDIODRV)
VL_OBJS+= mc146818rtc.o serial.o i8259.o i8254.o fdc.o m48t59.o
VL_OBJS+= ppc_prep.o ppc_chrp.o cuda.o adb.o openpic.o heathrow_pic.o mixeng.o
+DEFINES += -DHAS_AUDIO
endif
ifeq ($(TARGET_ARCH), mips)
VL_OBJS+= mips_r4k.o dma.o vga.o serial.o ne2000.o i8254.o i8259.o
@@ -317,7 +326,10 @@ VL_OBJS+=sdl.o
endif
ifdef CONFIG_COCOA
VL_OBJS+=cocoa.o
-COCOA_LIBS=-F/System/Library/Frameworks -framework Cocoa
+COCOA_LIBS=-F/System/Library/Frameworks -framework Cocoa -framework IOKit
+ifdef CONFIG_COREAUDIO
+COCOA_LIBS+=-framework CoreAudio
+endif
endif
ifdef CONFIG_SLIRP
DEFINES+=-I$(SRC_PATH)/slirp
@@ -349,6 +361,10 @@ ifeq ($(ARCH),ia64)
VL_LDFLAGS+=-Wl,-G0 -Wl,-T,$(SRC_PATH)/ia64.ld
endif
+ifdef CONFIG_WIN32
+SDL_LIBS := $(filter-out -mwindows, $(SDL_LIBS)) -mconsole
+endif
+
$(QEMU_SYSTEM): $(VL_OBJS) libqemu.a
$(CC) $(VL_LDFLAGS) -o $@ $^ $(LIBS) $(SDL_LIBS) $(COCOA_LIBS) $(VL_LIBS)
@@ -364,6 +380,9 @@ sdlaudio.o: sdlaudio.c
depend: $(SRCS)
$(CC) -MM $(CFLAGS) $(DEFINES) $^ 1>.depend
+vldepend: $(VL_OBJS:.o=.c)
+ $(CC) -MM $(CFLAGS) $(DEFINES) $^ 1>.depend
+
# libqemu
libqemu.a: $(LIBOBJS)
@@ -415,8 +434,6 @@ op.o: op.c op_template.c op_mem.c
op_helper.o: op_helper_mem.c
endif
-mixeng.o: mixeng.c mixeng.h mixeng_template.h
-
%.o: %.c
$(CC) $(CFLAGS) $(DEFINES) -c -o $@ $<
@@ -434,3 +451,9 @@ endif
ifneq ($(wildcard .depend),)
include .depend
endif
+
+ifeq (0, 1)
+audio.o sdlaudio.o dsoundaudio.o ossaudio.o wavaudio.o noaudio.o \
+fmodaudio.o alsaaudio.o mixeng.o: \
+CFLAGS := $(CFLAGS) -Wall -Werror -W -Wsign-compare
+endif
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
new file mode 100644
index 0000000000..133690576e
--- /dev/null
+++ b/audio/alsaaudio.c
@@ -0,0 +1,926 @@
+/*
+ * QEMU ALSA audio driver
+ *
+ * Copyright (c) 2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#include <alsa/asoundlib.h>
+#include "vl.h"
+
+#define AUDIO_CAP "alsa"
+#include "audio_int.h"
+
+typedef struct ALSAVoiceOut {
+ HWVoiceOut hw;
+ void *pcm_buf;
+ snd_pcm_t *handle;
+ int can_pause;
+ int was_enabled;
+} ALSAVoiceOut;
+
+typedef struct ALSAVoiceIn {
+ HWVoiceIn hw;
+ snd_pcm_t *handle;
+ void *pcm_buf;
+ int can_pause;
+} ALSAVoiceIn;
+
+static struct {
+ int size_in_usec_in;
+ int size_in_usec_out;
+ const char *pcm_name_in;
+ const char *pcm_name_out;
+ unsigned int buffer_size_in;
+ unsigned int period_size_in;
+ unsigned int buffer_size_out;
+ unsigned int period_size_out;
+ unsigned int threshold;
+
+ int buffer_size_in_overriden;
+ int period_size_in_overriden;
+
+ int buffer_size_out_overriden;
+ int period_size_out_overriden;
+} conf = {
+#ifdef HIGH_LATENCY
+ .size_in_usec_in = 1,
+ .size_in_usec_out = 1,
+#endif
+ .pcm_name_out = "hw:0,0",
+ .pcm_name_in = "hw:0,0",
+#ifdef HIGH_LATENCY
+ .buffer_size_in = 400000,
+ .period_size_in = 400000 / 4,
+ .buffer_size_out = 400000,
+ .period_size_out = 400000 / 4,
+#else
+#define DEFAULT_BUFFER_SIZE 1024
+#define DEFAULT_PERIOD_SIZE 256
+ .buffer_size_in = DEFAULT_BUFFER_SIZE,
+ .period_size_in = DEFAULT_PERIOD_SIZE,
+ .buffer_size_out = DEFAULT_BUFFER_SIZE,
+ .period_size_out = DEFAULT_PERIOD_SIZE,
+ .buffer_size_in_overriden = 0,
+ .buffer_size_out_overriden = 0,
+ .period_size_in_overriden = 0,
+ .period_size_out_overriden = 0,
+#endif
+ .threshold = 0
+};
+
+struct alsa_params_req {
+ int freq;
+ audfmt_e fmt;
+ int nchannels;
+ unsigned int buffer_size;
+ unsigned int period_size;
+};
+
+struct alsa_params_obt {
+ int freq;
+ audfmt_e fmt;
+ int nchannels;
+ int can_pause;
+ snd_pcm_uframes_t buffer_size;
+};
+
+static void GCC_FMT_ATTR (2, 3) alsa_logerr (int err, const char *fmt, ...)
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
+}
+
+static void GCC_FMT_ATTR (3, 4) alsa_logerr2 (
+ int err,
+ const char *typ,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", snd_strerror (err));
+}
+
+static void alsa_anal_close (snd_pcm_t **handlep)
+{
+ int err = snd_pcm_close (*handlep);
+ if (err) {
+ alsa_logerr (err, "Failed to close PCM handle %p\n", *handlep);
+ }
+ *handlep = NULL;
+}
+
+static int alsa_write (SWVoiceOut *sw, void *buf, int len)
+{
+ return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int aud_to_alsafmt (audfmt_e fmt)
+{
+ switch (fmt) {
+ case AUD_FMT_S8:
+ return SND_PCM_FORMAT_S8;
+
+ case AUD_FMT_U8:
+ return SND_PCM_FORMAT_U8;
+
+ case AUD_FMT_S16:
+ return SND_PCM_FORMAT_S16_LE;
+
+ case AUD_FMT_U16:
+ return SND_PCM_FORMAT_U16_LE;
+
+ default:
+ dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+ abort ();
+#endif
+ return SND_PCM_FORMAT_U8;
+ }
+}
+
+static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness)
+{
+ switch (alsafmt) {
+ case SND_PCM_FORMAT_S8:
+ *endianness = 0;
+ *fmt = AUD_FMT_S8;
+ break;
+
+ case SND_PCM_FORMAT_U8:
+ *endianness = 0;
+ *fmt = AUD_FMT_U8;
+ break;
+
+ case SND_PCM_FORMAT_S16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case SND_PCM_FORMAT_U16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ case SND_PCM_FORMAT_S16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case SND_PCM_FORMAT_U16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ default:
+ dolog ("Unrecognized audio format %d\n", alsafmt);
+ return -1;
+ }
+
+ return 0;
+}
+
+#ifdef DEBUG_MISMATCHES
+static void alsa_dump_info (struct alsa_params_req *req,
+ struct alsa_params_obt *obt)
+{
+ dolog ("parameter | requested value | obtained value\n");
+ dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
+ dolog ("channels | %10d | %10d\n",
+ req->nchannels, obt->nchannels);
+ dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
+ dolog ("============================================\n");
+ dolog ("requested: buffer size %d period size %d\n",
+ req->buffer_size, req->period_size);
+ dolog ("obtained: buffer size %ld\n", obt->buffer_size);
+}
+#endif
+
+static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold)
+{
+ int err;
+ snd_pcm_sw_params_t *sw_params;
+
+ snd_pcm_sw_params_alloca (&sw_params);
+
+ err = snd_pcm_sw_params_current (handle, sw_params);
+ if (err < 0) {
+ dolog ("Can not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to get current software parameters\n");
+ return;
+ }
+
+ err = snd_pcm_sw_params_set_start_threshold (handle, sw_params, threshold);
+ if (err < 0) {
+ dolog ("Can not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to set software threshold to %ld\n",
+ threshold);
+ return;
+ }
+
+ err = snd_pcm_sw_params (handle, sw_params);
+ if (err < 0) {
+ dolog ("Can not fully initialize DAC\n");
+ alsa_logerr (err, "Failed to set software parameters\n");
+ return;
+ }
+}
+
+static int alsa_open (int in, struct alsa_params_req *req,
+ struct alsa_params_obt *obt, snd_pcm_t **handlep)
+{
+ snd_pcm_t *handle;
+ snd_pcm_hw_params_t *hw_params;
+ int err, freq, nchannels;
+ const char *pcm_name = in ? conf.pcm_name_in : conf.pcm_name_out;
+ unsigned int period_size, buffer_size;
+ snd_pcm_uframes_t obt_buffer_size;
+ const char *typ = in ? "ADC" : "DAC";
+
+ freq = req->freq;
+ period_size = req->period_size;
+ buffer_size = req->buffer_size;
+ nchannels = req->nchannels;
+
+ snd_pcm_hw_params_alloca (&hw_params);
+
+ err = snd_pcm_open (
+ &handle,
+ pcm_name,
+ in ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
+ SND_PCM_NONBLOCK
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to open `%s':\n", pcm_name);
+ return -1;
+ }
+
+ err = snd_pcm_hw_params_any (handle, hw_params);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to initialize hardware parameters\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_access (
+ handle,
+ hw_params,
+ SND_PCM_ACCESS_RW_INTERLEAVED
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set access type\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set format %d\n", req->fmt);
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &freq, 0);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set frequency %d\n", req->freq);
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_set_channels_near (
+ handle,
+ hw_params,
+ &nchannels
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set number of channels %d\n",
+ req->nchannels);
+ goto err;
+ }
+
+ if (nchannels != 1 && nchannels != 2) {
+ alsa_logerr2 (err, typ,
+ "Can not handle obtained number of channels %d\n",
+ nchannels);
+ goto err;
+ }
+
+ if (!((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out))) {
+ if (!buffer_size) {
+ buffer_size = DEFAULT_BUFFER_SIZE;
+ period_size= DEFAULT_PERIOD_SIZE;
+ }
+ }
+
+ if (buffer_size) {
+ if ((in && conf.size_in_usec_in) || (!in && conf.size_in_usec_out)) {
+ if (period_size) {
+ err = snd_pcm_hw_params_set_period_time_near (
+ handle,
+ hw_params,
+ &period_size,
+ 0);
+ if (err < 0) {
+ alsa_logerr2 (err, typ,
+ "Failed to set period time %d\n",
+ req->period_size);
+ goto err;
+ }
+ }
+
+ err = snd_pcm_hw_params_set_buffer_time_near (
+ handle,
+ hw_params,
+ &buffer_size,
+ 0);
+
+ if (err < 0) {
+ alsa_logerr2 (err, typ,
+ "Failed to set buffer time %d\n",
+ req->buffer_size);
+ goto err;
+ }
+ }
+ else {
+ int dir;
+ snd_pcm_uframes_t minval;
+
+ if (period_size) {
+ minval = period_size;
+ dir = 0;
+
+ err = snd_pcm_hw_params_get_period_size_min (
+ hw_params,
+ &minval,
+ &dir
+ );
+ if (err < 0) {
+ alsa_logerr (
+ err,
+ "Can not get minmal period size for %s\n",
+ typ
+ );
+ }
+ else {
+ if (period_size < minval) {
+ if ((in && conf.period_size_in_overriden)
+ || (!in && conf.period_size_out_overriden)) {
+ dolog ("%s period size(%d) is less "
+ "than minmal period size(%ld)\n",
+ typ,
+ period_size,
+ minval);
+ }
+ period_size = minval;
+ }
+ }
+
+ err = snd_pcm_hw_params_set_period_size (
+ handle,
+ hw_params,
+ period_size,
+ 0
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set period size %d\n",
+ req->period_size);
+ goto err;
+ }
+ }
+
+ minval = buffer_size;
+ err = snd_pcm_hw_params_get_buffer_size_min (
+ hw_params,
+ &minval
+ );
+ if (err < 0) {
+ alsa_logerr (err, "Can not get minmal buffer size for %s\n",
+ typ);
+ }
+ else {
+ if (buffer_size < minval) {
+ if ((in && conf.buffer_size_in_overriden)
+ || (!in && conf.buffer_size_out_overriden)) {
+ dolog (
+ "%s buffer size(%d) is less "
+ "than minimal buffer size(%ld)\n",
+ typ,
+ buffer_size,
+ minval
+ );
+ }
+ buffer_size = minval;
+ }
+ }
+
+ err = snd_pcm_hw_params_set_buffer_size (
+ handle,
+ hw_params,
+ buffer_size
+ );
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to set buffer size %d\n",
+ req->buffer_size);
+ goto err;
+ }
+ }
+ }
+ else {
+ dolog ("warning: buffer size is not set\n");
+ }
+
+ err = snd_pcm_hw_params (handle, hw_params);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to apply audio parameters\n");
+ goto err;
+ }
+
+ err = snd_pcm_hw_params_get_buffer_size (hw_params, &obt_buffer_size);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Failed to get buffer size\n");
+ goto err;
+ }
+
+ err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr2 (err, typ, "Can not prepare handle %p\n", handle);
+ goto err;
+ }
+
+ obt->can_pause = snd_pcm_hw_params_can_pause (hw_params);
+ if (obt->can_pause < 0) {
+ alsa_logerr (err, "Can not get pause capability for %s\n", typ);
+ obt->can_pause = 0;
+ }
+
+ if (!in && conf.threshold) {
+ snd_pcm_uframes_t threshold;
+ int bytes_per_sec;
+
+ bytes_per_sec = freq
+ << (nchannels == 2)
+ << (req->fmt == AUD_FMT_S16 || req->fmt == AUD_FMT_U16);
+
+ threshold = (conf.threshold * bytes_per_sec) / 1000;
+ alsa_set_threshold (handle, threshold);
+ }
+
+ obt->fmt = req->fmt;
+ obt->nchannels = nchannels;
+ obt->freq = freq;
+ obt->buffer_size = snd_pcm_frames_to_bytes (handle, obt_buffer_size);
+ *handlep = handle;
+
+ if (obt->fmt != req->fmt ||
+ obt->nchannels != req->nchannels ||
+ obt->freq != req->freq) {
+#ifdef DEBUG_MISMATCHES
+ dolog ("Audio paramters mismatch for %s\n", typ);
+ alsa_dump_info (req, obt);
+#endif
+ }
+
+#ifdef DEBUG
+ alsa_dump_info (req, obt);
+#endif
+ return 0;
+
+ err:
+ alsa_anal_close (&handle);
+ return -1;
+}
+
+static int alsa_recover (snd_pcm_t *handle)
+{
+ int err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to prepare handle %p\n", handle);
+ return -1;
+ }
+ return 0;
+}
+
+static int alsa_run_out (HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ int rpos, live, decr;
+ int samples;
+ uint8_t *dst;
+ st_sample_t *src;
+ snd_pcm_sframes_t avail;
+
+ live = audio_pcm_hw_get_live_out (hw);
+ if (!live) {
+ return 0;
+ }
+
+ avail = snd_pcm_avail_update (alsa->handle);
+ if (avail < 0) {
+ if (avail == -EPIPE) {
+ if (!alsa_recover (alsa->handle)) {
+ avail = snd_pcm_avail_update (alsa->handle);
+ if (avail >= 0) {
+ goto ok;
+ }
+ }
+ }
+
+ alsa_logerr (avail, "Can not get amount free space\n");
+ return 0;
+ }
+
+ ok:
+ decr = audio_MIN (live, avail);
+ samples = decr;
+ rpos = hw->rpos;
+ while (samples) {
+ int left_till_end_samples = hw->samples - rpos;
+ int convert_samples = audio_MIN (samples, left_till_end_samples);
+ snd_pcm_sframes_t written;
+
+ src = hw->mix_buf + rpos;
+ dst = advance (alsa->pcm_buf, rpos << hw->info.shift);
+
+ hw->clip (dst, src, convert_samples);
+
+ again:
+ written = snd_pcm_writei (alsa->handle, dst, convert_samples);
+
+ if (written < 0) {
+ switch (written) {
+ case -EPIPE:
+ if (!alsa_recover (alsa->handle)) {
+ goto again;
+ }
+ dolog (
+ "Failed to write %d frames to %p, handle %p not prepared\n",
+ convert_samples,
+ dst,
+ alsa->handle
+ );
+ goto exit;
+
+ case -EAGAIN:
+ goto again;
+
+ default:
+ alsa_logerr (written, "Failed to write %d frames to %p\n",
+ convert_samples, dst);
+ goto exit;
+ }
+ }
+
+ mixeng_clear (src, written);
+ rpos = (rpos + written) % hw->samples;
+ samples -= written;
+ }
+
+ exit:
+ hw->rpos = rpos;
+ return decr;
+}
+
+static void alsa_fini_out (HWVoiceOut *hw)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+ ldebug ("alsa_fini\n");
+ alsa_anal_close (&alsa->handle);
+
+ if (alsa->pcm_buf) {
+ qemu_free (alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
+ }
+}
+
+static int alsa_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
+{
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+ struct alsa_params_req req;
+ struct alsa_params_obt obt;
+ audfmt_e effective_fmt;
+ int endianness;
+ int err;
+ snd_pcm_t *handle;
+
+ req.fmt = aud_to_alsafmt (fmt);
+ req.freq = freq;
+ req.nchannels = nchannels;
+ req.period_size = conf.period_size_out;
+ req.buffer_size = conf.buffer_size_out;
+
+ if (alsa_open (0, &req, &obt, &handle)) {
+ return -1;
+ }
+
+ err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+ if (err) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ audio_pcm_init_info (
+ &hw->info,
+ obt.freq,
+ obt.nchannels,
+ effective_fmt,
+ audio_need_to_swap_endian (endianness)
+ );
+ alsa->can_pause = obt.can_pause;
+ hw->bufsize = obt.buffer_size;
+
+ alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ if (!alsa->pcm_buf) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ alsa->handle = handle;
+ alsa->was_enabled = 0;
+ return 0;
+}
+
+static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+ int err;
+ ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ ldebug ("enabling voice\n");
+ audio_pcm_info_clear_buf (&hw->info, alsa->pcm_buf, hw->samples);
+ if (alsa->can_pause) {
+ /* Why this was_enabled madness is needed at all?? */
+ if (alsa->was_enabled) {
+ err = snd_pcm_pause (alsa->handle, 0);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to resume playing\n");
+ /* not fatal really */
+ }
+ }
+ else {
+ alsa->was_enabled = 1;
+ }
+ }
+ break;
+
+ case VOICE_DISABLE:
+ ldebug ("disabling voice\n");
+ if (alsa->can_pause) {
+ err = snd_pcm_pause (alsa->handle, 1);
+ if (err < 0) {
+ alsa_logerr (err, "Failed to stop playing\n");
+ /* not fatal really */
+ }
+ }
+ break;
+ }
+ return 0;
+}
+
+static int alsa_init_in (HWVoiceIn *hw,
+ int freq, int nchannels, audfmt_e fmt)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ struct alsa_params_req req;
+ struct alsa_params_obt obt;
+ int endianness;
+ int err;
+ audfmt_e effective_fmt;
+ snd_pcm_t *handle;
+
+ req.fmt = aud_to_alsafmt (fmt);
+ req.freq = freq;
+ req.nchannels = nchannels;
+ req.period_size = conf.period_size_in;
+ req.buffer_size = conf.buffer_size_in;
+
+ if (alsa_open (1, &req, &obt, &handle)) {
+ return -1;
+ }
+
+ err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+ if (err) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ audio_pcm_init_info (
+ &hw->info,
+ obt.freq,
+ obt.nchannels,
+ effective_fmt,
+ audio_need_to_swap_endian (endianness)
+ );
+ alsa->can_pause = obt.can_pause;
+ hw->bufsize = obt.buffer_size;
+ alsa->pcm_buf = qemu_mallocz (hw->bufsize);
+ if (!alsa->pcm_buf) {
+ alsa_anal_close (&handle);
+ return -1;
+ }
+
+ alsa->handle = handle;
+ return 0;
+}
+
+static void alsa_fini_in (HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+ alsa_anal_close (&alsa->handle);
+
+ if (alsa->pcm_buf) {
+ qemu_free (alsa->pcm_buf);
+ alsa->pcm_buf = NULL;
+ }
+}
+
+static int alsa_run_in (HWVoiceIn *hw)
+{
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+ int hwshift = hw->info.shift;
+ int i;
+ int live = audio_pcm_hw_get_live_in (hw);
+ int dead = hw->samples - live;
+ struct {
+ int add;
+ int len;
+ } bufs[2] = {
+ { hw->wpos, 0 },
+ { 0, 0 }
+ };
+
+ snd_pcm_uframes_t read_samples = 0;
+
+ if (!dead) {
+ return 0;
+ }
+
+ if (hw->wpos + dead > hw->samples) {
+ bufs[0].len = (hw->samples - hw->wpos);
+ bufs[1].len = (dead - (hw->samples - hw->wpos));
+ }
+ else {
+ bufs[0].len = dead;
+ }
+
+
+ for (i = 0; i < 2; ++i) {
+ void *src;
+ st_sample_t *dst;
+ snd_pcm_sframes_t nread;
+ snd_pcm_uframes_t len;
+
+ len = bufs[i].len;
+
+ src = advance (alsa->pcm_buf, bufs[i].add << hwshift);
+ dst = hw->conv_buf + bufs[i].add;
+
+ while (len) {
+ nread = snd_pcm_readi (alsa->handle, src, len);
+
+ if (nread < 0) {
+ switch (nread) {
+ case -EPIPE:
+ if (!alsa_recover (alsa->handle)) {
+ continue;
+ }
+ dolog (
+ "Failed to read %ld frames from %p, "
+ "handle %p not prepared\n",
+ len,
+ src,
+ alsa->handle
+ );
+ goto exit;
+
+ case -EAGAIN:
+ continue;
+
+ default:
+ alsa_logerr (
+ nread,
+ "Failed to read %ld frames from %p\n",
+ len,
+ src
+ );
+ goto exit;
+ }
+ }
+
+ hw->conv (dst, src, nread, &nominal_volume);
+
+ src = advance (src, nread << hwshift);
+ dst += nread;
+
+ read_samples += nread;
+ len -= nread;
+ }
+ }
+
+ exit:
+ hw->wpos = (hw->wpos + read_samples) % hw->samples;
+ return read_samples;
+}
+
+static int alsa_read (SWVoiceIn *sw, void *buf, int size)
+{
+ return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+ (void) hw;
+ (void) cmd;
+ return 0;
+}
+
+static void *alsa_audio_init (void)
+{
+ return &conf;
+}
+
+static void alsa_audio_fini (void *opaque)
+{
+ (void) opaque;
+}
+
+static struct audio_option alsa_options[] = {
+ {"DAC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_out,
+ "DAC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+ {"DAC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_out,
+ "DAC period size", &conf.period_size_out_overriden, 0},
+ {"DAC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_out,
+ "DAC buffer size", &conf.buffer_size_out_overriden, 0},
+
+ {"ADC_SIZE_IN_USEC", AUD_OPT_BOOL, &conf.size_in_usec_in,
+ "ADC period/buffer size in microseconds (otherwise in frames)", NULL, 0},
+ {"ADC_PERIOD_SIZE", AUD_OPT_INT, &conf.period_size_in,
+ "ADC period size", &conf.period_size_in_overriden, 0},
+ {"ADC_BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_size_in,
+ "ADC buffer size", &conf.buffer_size_in_overriden, 0},
+
+ {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+ "(undocumented)", NULL, 0},
+
+ {"DAC_DEV", AUD_OPT_STR, &conf.pcm_name_out,
+ "DAC device name (for instance dmix)", NULL, 0},
+
+ {"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
+ "ADC device name", NULL, 0},
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops alsa_pcm_ops = {
+ alsa_init_out,
+ alsa_fini_out,
+ alsa_run_out,
+ alsa_write,
+ alsa_ctl_out,
+
+ alsa_init_in,
+ alsa_fini_in,
+ alsa_run_in,
+ alsa_read,
+ alsa_ctl_in
+};
+
+struct audio_driver alsa_audio_driver = {
+ INIT_FIELD (name = ) "alsa",
+ INIT_FIELD (descr = ) "ALSA http://www.alsa-project.org",
+ INIT_FIELD (options = ) alsa_options,
+ INIT_FIELD (init = ) alsa_audio_init,
+ INIT_FIELD (fini = ) alsa_audio_fini,
+ INIT_FIELD (pcm_ops = ) &alsa_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) INT_MAX,
+ INIT_FIELD (max_voices_in = ) INT_MAX,
+ INIT_FIELD (voice_size_out = ) sizeof (ALSAVoiceOut),
+ INIT_FIELD (voice_size_in = ) sizeof (ALSAVoiceIn)
+};
diff --git a/audio/audio.c b/audio/audio.c
index 0c0c8dd86f..1a3925d4e9 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -1,8 +1,8 @@
/*
* QEMU Audio subsystem
- *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- *
+ *
+ * Copyright (c) 2003-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -21,34 +21,78 @@
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
-#include <assert.h>
#include "vl.h"
-#define USE_WAV_AUDIO
+#define AUDIO_CAP "audio"
+#include "audio_int.h"
-#include "audio/audio_int.h"
+static void audio_pcm_hw_fini_in (HWVoiceIn *hw);
+static void audio_pcm_hw_fini_out (HWVoiceOut *hw);
-#define dolog(...) AUD_log ("audio", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+static LIST_HEAD (hw_in_listhead, HWVoiceIn) hw_head_in;
+static LIST_HEAD (hw_out_listhead, HWVoiceOut) hw_head_out;
-#define QC_AUDIO_DRV "QEMU_AUDIO_DRV"
-#define QC_VOICES "QEMU_VOICES"
-#define QC_FIXED_FORMAT "QEMU_FIXED_FORMAT"
-#define QC_FIXED_FREQ "QEMU_FIXED_FREQ"
+/* #define DEBUG_PLIVE */
+/* #define DEBUG_LIVE */
+/* #define DEBUG_OUT */
-static HWVoice *hw_voices;
+static struct audio_driver *drvtab[] = {
+#ifdef CONFIG_OSS
+ &oss_audio_driver,
+#endif
+#ifdef CONFIG_ALSA
+ &alsa_audio_driver,
+#endif
+#ifdef CONFIG_COREAUDIO
+ &coreaudio_audio_driver,
+#endif
+#ifdef CONFIG_DSOUND
+ &dsound_audio_driver,
+#endif
+#ifdef CONFIG_FMOD
+ &fmod_audio_driver,
+#endif
+#ifdef CONFIG_SDL
+ &sdl_audio_driver,
+#endif
+ &no_audio_driver,
+ &wav_audio_driver
+};
AudioState audio_state = {
+ /* Out */
+ 1, /* use fixed settings */
+ 44100, /* fixed frequency */
+ 2, /* fixed channels */
+ AUD_FMT_S16, /* fixed format */
+ 1, /* number of hw voices */
+ 1, /* greedy */
+
+ /* In */
1, /* use fixed settings */
44100, /* fixed frequency */
2, /* fixed channels */
AUD_FMT_S16, /* fixed format */
1, /* number of hw voices */
- -1 /* voice size */
+ 1, /* greedy */
+
+ NULL, /* driver opaque */
+ NULL, /* driver */
+
+ NULL, /* timer handle */
+ { 0 }, /* period */
+ 0 /* plive */
+};
+
+volume_t nominal_volume = {
+ 0,
+#ifdef FLOAT_MIXENG
+ 1.0,
+ 1.0
+#else
+ UINT_MAX,
+ UINT_MAX
+#endif
};
/* http://www.df.lth.se/~john_e/gems/gem002d.html */
@@ -68,70 +112,334 @@ inline uint32_t lsbindex (uint32_t u)
return popcount ((u&-u)-1);
}
-int audio_get_conf_int (const char *key, int defval)
+#ifdef AUDIO_IS_FLAWLESS_AND_NO_CHECKS_ARE_REQURIED
+#error No its not
+#else
+int audio_bug (const char *funcname, int cond)
+{
+ if (cond) {
+ static int shown;
+
+ AUD_log (NULL, "Error a bug that was just triggered in %s\n", funcname);
+ if (!shown) {
+ shown = 1;
+ AUD_log (NULL, "Save all your work and restart without audio\n");
+ AUD_log (NULL, "Please send bug report to malc@pulsesoft.com\n");
+ AUD_log (NULL, "I am sorry\n");
+ }
+ AUD_log (NULL, "Context:\n");
+
+#if defined AUDIO_BREAKPOINT_ON_BUG
+# if defined HOST_I386
+# if defined __GNUC__
+ __asm__ ("int3");
+# elif defined _MSC_VER
+ _asm _emit 0xcc;
+# else
+ abort ();
+# endif
+# else
+ abort ();
+# endif
+#endif
+ }
+
+ return cond;
+}
+#endif
+
+static char *audio_alloc_prefix (const char *s)
+{
+ const char qemu_prefix[] = "QEMU_";
+ size_t len;
+ char *r;
+
+ if (!s) {
+ return NULL;
+ }
+
+ len = strlen (s);
+ r = qemu_malloc (len + sizeof (qemu_prefix));
+
+ if (r) {
+ size_t i;
+ char *u = r + sizeof (qemu_prefix) - 1;
+
+ strcpy (r, qemu_prefix);
+ strcat (r, s);
+
+ for (i = 0; i < len; ++i) {
+ u[i] = toupper (u[i]);
+ }
+ }
+ return r;
+}
+
+const char *audio_audfmt_to_string (audfmt_e fmt)
+{
+ switch (fmt) {
+ case AUD_FMT_U8:
+ return "U8";
+
+ case AUD_FMT_U16:
+ return "U16";
+
+ case AUD_FMT_S8:
+ return "S8";
+
+ case AUD_FMT_S16:
+ return "S16";
+ }
+
+ dolog ("Bogus audfmt %d returning S16\n", fmt);
+ return "S16";
+}
+
+audfmt_e audio_string_to_audfmt (const char *s, audfmt_e defval, int *defaultp)
{
- int val = defval;
+ if (!strcasecmp (s, "u8")) {
+ *defaultp = 0;
+ return AUD_FMT_U8;
+ }
+ else if (!strcasecmp (s, "u16")) {
+ *defaultp = 0;
+ return AUD_FMT_U16;
+ }
+ else if (!strcasecmp (s, "s8")) {
+ *defaultp = 0;
+ return AUD_FMT_S8;
+ }
+ else if (!strcasecmp (s, "s16")) {
+ *defaultp = 0;
+ return AUD_FMT_S16;
+ }
+ else {
+ dolog ("Bogus audio format `%s' using %s\n",
+ s, audio_audfmt_to_string (defval));
+ *defaultp = 1;
+ return defval;
+ }
+}
+
+static audfmt_e audio_get_conf_fmt (const char *envname,
+ audfmt_e defval,
+ int *defaultp)
+{
+ const char *var = getenv (envname);
+ if (!var) {
+ *defaultp = 1;
+ return defval;
+ }
+ return audio_string_to_audfmt (var, defval, defaultp);
+}
+
+static int audio_get_conf_int (const char *key, int defval, int *defaultp)
+{
+ int val;
char *strval;
strval = getenv (key);
if (strval) {
+ *defaultp = 0;
val = atoi (strval);
+ return val;
+ }
+ else {
+ *defaultp = 1;
+ return defval;
}
-
- return val;
}
-const char *audio_get_conf_str (const char *key, const char *defval)
+static const char *audio_get_conf_str (const char *key,
+ const char *defval,
+ int *defaultp)
{
const char *val = getenv (key);
- if (!val)
+ if (!val) {
+ *defaultp = 1;
return defval;
- else
+ }
+ else {
+ *defaultp = 0;
return val;
+ }
}
void AUD_log (const char *cap, const char *fmt, ...)
{
va_list ap;
- fprintf (stderr, "%s: ", cap);
+ if (cap) {
+ fprintf (stderr, "%s: ", cap);
+ }
va_start (ap, fmt);
vfprintf (stderr, fmt, ap);
va_end (ap);
}
-/*
- * Soft Voice
- */
-void pcm_sw_free_resources (SWVoice *sw)
+void AUD_vlog (const char *cap, const char *fmt, va_list ap)
{
- if (sw->buf) qemu_free (sw->buf);
- if (sw->rate) st_rate_stop (sw->rate);
- sw->buf = NULL;
- sw->rate = NULL;
+ if (cap) {
+ fprintf (stderr, "%s: ", cap);
+ }
+ vfprintf (stderr, fmt, ap);
}
-int pcm_sw_alloc_resources (SWVoice *sw)
+static void audio_print_options (const char *prefix,
+ struct audio_option *opt)
{
- sw->buf = qemu_mallocz (sw->hw->samples * sizeof (st_sample_t));
- if (!sw->buf)
- return -1;
+ char *uprefix;
- sw->rate = st_rate_start (sw->freq, sw->hw->freq);
- if (!sw->rate) {
- qemu_free (sw->buf);
- sw->buf = NULL;
- return -1;
+ if (!prefix) {
+ dolog ("No prefix specified\n");
+ return;
}
- return 0;
+
+ if (!opt) {
+ dolog ("No options\n");
+ return;
+ }
+
+ uprefix = audio_alloc_prefix (prefix);
+
+ for (; opt->name; opt++) {
+ const char *state = "default";
+ printf (" %s_%s: ", uprefix, opt->name);
+
+ if (opt->overridenp && *opt->overridenp) {
+ state = "current";
+ }
+
+ switch (opt->tag) {
+ case AUD_OPT_BOOL:
+ {
+ int *intp = opt->valp;
+ printf ("boolean, %s = %d\n", state, *intp ? 1 : 0);
+ }
+ break;
+
+ case AUD_OPT_INT:
+ {
+ int *intp = opt->valp;
+ printf ("integer, %s = %d\n", state, *intp);
+ }
+ break;
+
+ case AUD_OPT_FMT:
+ {
+ audfmt_e *fmtp = opt->valp;
+ printf (
+ "format, %s = %s, (one of: U8 S8 U16 S16)\n",
+ state,
+ audio_audfmt_to_string (*fmtp)
+ );
+ }
+ break;
+
+ case AUD_OPT_STR:
+ {
+ const char **strp = opt->valp;
+ printf ("string, %s = %s\n",
+ state,
+ *strp ? *strp : "(not set)");
+ }
+ break;
+
+ default:
+ printf ("???\n");
+ dolog ("Bad value tag for option %s_%s %d\n",
+ uprefix, opt->name, opt->tag);
+ break;
+ }
+ printf (" %s\n", opt->descr);
+ }
+
+ qemu_free (uprefix);
}
-void pcm_sw_fini (SWVoice *sw)
+static void audio_process_options (const char *prefix,
+ struct audio_option *opt)
{
- pcm_sw_free_resources (sw);
+ char *optname;
+ const char qemu_prefix[] = "QEMU_";
+ size_t preflen;
+
+ if (audio_bug (AUDIO_FUNC, !prefix)) {
+ dolog ("prefix = NULL\n");
+ return;
+ }
+
+ if (audio_bug (AUDIO_FUNC, !opt)) {
+ dolog ("opt = NULL\n");
+ return;
+ }
+
+ preflen = strlen (prefix);
+
+ for (; opt->name; opt++) {
+ size_t len, i;
+ int def;
+
+ if (!opt->valp) {
+ dolog ("Option value pointer for `%s' is not set\n",
+ opt->name);
+ continue;
+ }
+
+ len = strlen (opt->name);
+ optname = qemu_malloc (len + preflen + sizeof (qemu_prefix) + 1);
+ if (!optname) {
+ dolog ("Can not allocate memory for option name `%s'\n",
+ opt->name);
+ continue;
+ }
+
+ strcpy (optname, qemu_prefix);
+ for (i = 0; i <= preflen; ++i) {
+ optname[i + sizeof (qemu_prefix) - 1] = toupper (prefix[i]);
+ }
+ strcat (optname, "_");
+ strcat (optname, opt->name);
+
+ def = 1;
+ switch (opt->tag) {
+ case AUD_OPT_BOOL:
+ case AUD_OPT_INT:
+ {
+ int *intp = opt->valp;
+ *intp = audio_get_conf_int (optname, *intp, &def);
+ }
+ break;
+
+ case AUD_OPT_FMT:
+ {
+ audfmt_e *fmtp = opt->valp;
+ *fmtp = audio_get_conf_fmt (optname, *fmtp, &def);
+ }
+ break;
+
+ case AUD_OPT_STR:
+ {
+ const char **strp = opt->valp;
+ *strp = audio_get_conf_str (optname, *strp, &def);
+ }
+ break;
+
+ default:
+ dolog ("Bad value tag for option `%s' - %d\n",
+ optname, opt->tag);
+ break;
+ }
+
+ if (!opt->overridenp) {
+ opt->overridenp = &opt->overriden;
+ }
+ *opt->overridenp = !def;
+ qemu_free (optname);
+ }
}
-int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
- int nchannels, audfmt_e fmt)
+static int audio_pcm_info_eq (struct audio_pcm_info *info, int freq,
+ int nchannels, audfmt_e fmt)
{
int bits = 8, sign = 0;
@@ -147,746 +455,1167 @@ int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
bits = 16;
break;
}
+ return info->freq == freq
+ && info->nchannels == nchannels
+ && info->sign == sign
+ && info->bits == bits;
+}
- sw->hw = hw;
- sw->freq = freq;
- sw->fmt = fmt;
- sw->nchannels = nchannels;
- sw->shift = (nchannels == 2) + (bits == 16);
- sw->align = (1 << sw->shift) - 1;
- sw->left = 0;
- sw->pos = 0;
- sw->wpos = 0;
- sw->live = 0;
- sw->ratio = (sw->hw->freq * ((int64_t) INT_MAX)) / sw->freq;
- sw->bytes_per_second = sw->freq << sw->shift;
- sw->conv = mixeng_conv[nchannels == 2][sign][bits == 16];
-
- pcm_sw_free_resources (sw);
- return pcm_sw_alloc_resources (sw);
-}
-
-/* Hard voice */
-void pcm_hw_free_resources (HWVoice *hw)
-{
- if (hw->mix_buf)
- qemu_free (hw->mix_buf);
- hw->mix_buf = NULL;
+void audio_pcm_init_info (struct audio_pcm_info *info, int freq,
+ int nchannels, audfmt_e fmt, int swap_endian)
+{
+ int bits = 8, sign = 0;
+
+ switch (fmt) {
+ case AUD_FMT_S8:
+ sign = 1;
+ case AUD_FMT_U8:
+ break;
+
+ case AUD_FMT_S16:
+ sign = 1;
+ case AUD_FMT_U16:
+ bits = 16;
+ break;
+ }
+
+ info->freq = freq;
+ info->bits = bits;
+ info->sign = sign;
+ info->nchannels = nchannels;
+ info->shift = (nchannels == 2) + (bits == 16);
+ info->align = (1 << info->shift) - 1;
+ info->bytes_per_second = info->freq << info->shift;
+ info->swap_endian = swap_endian;
}
-int pcm_hw_alloc_resources (HWVoice *hw)
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len)
{
- hw->mix_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
- if (!hw->mix_buf)
- return -1;
- return 0;
+ if (!len) {
+ return;
+ }
+
+ if (info->sign) {
+ memset (buf, len << info->shift, 0x00);
+ }
+ else {
+ if (info->bits == 8) {
+ memset (buf, len << info->shift, 0x80);
+ }
+ else {
+ int i;
+ uint16_t *p = buf;
+ int shift = info->nchannels - 1;
+ short s = INT16_MAX;
+
+ if (info->swap_endian) {
+ s = bswap16 (s);
+ }
+
+ for (i = 0; i < len << shift; i++) {
+ p[i] = s;
+ }
+ }
+ }
}
-void pcm_hw_fini (HWVoice *hw)
+/*
+ * Hard voice (capture)
+ */
+static void audio_pcm_hw_free_resources_in (HWVoiceIn *hw)
{
- if (hw->active) {
- ldebug ("pcm_hw_fini: %d %d %d\n", hw->freq, hw->nchannels, hw->fmt);
- pcm_hw_free_resources (hw);
- hw->pcm_ops->fini (hw);
- memset (hw, 0, audio_state.drv->voice_size);
+ if (hw->conv_buf) {
+ qemu_free (hw->conv_buf);
}
+ hw->conv_buf = NULL;
}
-void pcm_hw_gc (HWVoice *hw)
+static int audio_pcm_hw_alloc_resources_in (HWVoiceIn *hw)
{
- if (hw->nb_voices)
- return;
+ hw->conv_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
+ if (!hw->conv_buf) {
+ return -1;
+ }
+ return 0;
+}
+
+static int audio_pcm_hw_init_in (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt)
+{
+ audio_pcm_hw_fini_in (hw);
- pcm_hw_fini (hw);
+ if (hw->pcm_ops->init_in (hw, freq, nchannels, fmt)) {
+ memset (hw, 0, audio_state.drv->voice_size_in);
+ return -1;
+ }
+ LIST_INIT (&hw->sw_head);
+ hw->active = 1;
+ hw->samples = hw->bufsize >> hw->info.shift;
+ hw->conv =
+ mixeng_conv
+ [nchannels == 2]
+ [hw->info.sign]
+ [hw->info.swap_endian]
+ [hw->info.bits == 16];
+ if (audio_pcm_hw_alloc_resources_in (hw)) {
+ audio_pcm_hw_free_resources_in (hw);
+ return -1;
+ }
+ return 0;
}
-int pcm_hw_get_live (HWVoice *hw)
+static uint64_t audio_pcm_hw_find_min_in (HWVoiceIn *hw)
{
- int i, alive = 0, live = hw->samples;
+ SWVoiceIn *sw;
+ int m = hw->total_samples_captured;
- for (i = 0; i < hw->nb_voices; i++) {
- if (hw->pvoice[i]->live) {
- live = audio_MIN (hw->pvoice[i]->live, live);
- alive += 1;
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (sw->active) {
+ m = audio_MIN (m, sw->total_hw_samples_acquired);
}
}
+ return m;
+}
- if (alive)
- return live;
- else
- return -1;
+int audio_pcm_hw_get_live_in (HWVoiceIn *hw)
+{
+ int live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw);
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+ return live;
}
-int pcm_hw_get_live2 (HWVoice *hw, int *nb_active)
+/*
+ * Soft voice (capture)
+ */
+static void audio_pcm_sw_free_resources_in (SWVoiceIn *sw)
{
- int i, alive = 0, live = hw->samples;
+ if (sw->conv_buf) {
+ qemu_free (sw->conv_buf);
+ }
- *nb_active = 0;
- for (i = 0; i < hw->nb_voices; i++) {
- if (hw->pvoice[i]->live) {
- if (hw->pvoice[i]->live < live) {
- *nb_active = hw->pvoice[i]->active != 0;
- live = hw->pvoice[i]->live;
- }
- alive += 1;
- }
+ if (sw->rate) {
+ st_rate_stop (sw->rate);
}
- if (alive)
- return live;
- else
- return -1;
+ sw->conv_buf = NULL;
+ sw->rate = NULL;
}
-void pcm_hw_dec_live (HWVoice *hw, int decr)
+static int audio_pcm_sw_alloc_resources_in (SWVoiceIn *sw)
{
- int i;
+ int samples = ((int64_t) sw->hw->samples << 32) / sw->ratio;
+ sw->conv_buf = qemu_mallocz (samples * sizeof (st_sample_t));
+ if (!sw->conv_buf) {
+ return -1;
+ }
- for (i = 0; i < hw->nb_voices; i++) {
- if (hw->pvoice[i]->live) {
- hw->pvoice[i]->live -= decr;
- }
+ sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq);
+ if (!sw->rate) {
+ qemu_free (sw->conv_buf);
+ sw->conv_buf = NULL;
+ return -1;
}
+ return 0;
}
-void pcm_hw_clear (HWVoice *hw, void *buf, int len)
+static int audio_pcm_sw_init_in (SWVoiceIn *sw, HWVoiceIn *hw, const char *name,
+ int freq, int nchannels, audfmt_e fmt)
{
- if (!len)
- return;
+ audio_pcm_init_info (&sw->info, freq, nchannels, fmt,
+ /* None of the cards emulated by QEMU are big-endian
+ hence following shortcut */
+ audio_need_to_swap_endian (0));
+ sw->hw = hw;
+ sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;
- switch (hw->fmt) {
- case AUD_FMT_S16:
- case AUD_FMT_S8:
- memset (buf, len << hw->shift, 0x00);
- break;
+ sw->clip =
+ mixeng_clip
+ [nchannels == 2]
+ [sw->info.sign]
+ [sw->info.swap_endian]
+ [sw->info.bits == 16];
- case AUD_FMT_U8:
- memset (buf, len << hw->shift, 0x80);
- break;
+ sw->name = qemu_strdup (name);
+ audio_pcm_sw_free_resources_in (sw);
+ return audio_pcm_sw_alloc_resources_in (sw);
+}
- case AUD_FMT_U16:
- {
- unsigned int i;
- uint16_t *p = buf;
- int shift = hw->nchannels - 1;
+static int audio_pcm_sw_get_rpos_in (SWVoiceIn *sw)
+{
+ HWVoiceIn *hw = sw->hw;
+ int live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ int rpos;
- for (i = 0; i < len << shift; i++) {
- p[i] = INT16_MAX;
- }
- }
- break;
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+
+ rpos = hw->wpos - live;
+ if (rpos >= 0) {
+ return rpos;
+ }
+ else {
+ return hw->samples + rpos;
}
}
-int pcm_hw_write (SWVoice *sw, void *buf, int size)
+int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int size)
{
- int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
- int ret = 0, pos = 0;
- if (!sw)
- return size;
+ HWVoiceIn *hw = sw->hw;
+ int samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0;
+ st_sample_t *src, *dst = sw->conv_buf;
- hwsamples = sw->hw->samples;
- samples = size >> sw->shift;
+ rpos = audio_pcm_sw_get_rpos_in (sw) % hw->samples;
- if (!sw->live) {
- sw->wpos = sw->hw->rpos;
+ live = hw->total_samples_captured - sw->total_hw_samples_acquired;
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live_in=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+
+ samples = size >> sw->info.shift;
+ if (!live) {
+ return 0;
}
- wpos = sw->wpos;
- live = sw->live;
- dead = hwsamples - live;
- swlim = (dead * ((int64_t) INT_MAX)) / sw->ratio;
- swlim = audio_MIN (swlim, samples);
- ldebug ("size=%d live=%d dead=%d swlim=%d wpos=%d\n",
- size, live, dead, swlim, wpos);
- if (swlim)
- sw->conv (sw->buf, buf, swlim);
+ swlim = (live * sw->ratio) >> 32;
+ swlim = audio_MIN (swlim, samples);
while (swlim) {
- dead = hwsamples - live;
- left = hwsamples - wpos;
- blck = audio_MIN (dead, left);
- if (!blck) {
- /* dolog ("swlim=%d\n", swlim); */
+ src = hw->conv_buf + rpos;
+ isamp = hw->wpos - rpos;
+ /* XXX: <= ? */
+ if (isamp <= 0) {
+ isamp = hw->samples - rpos;
+ }
+
+ if (!isamp) {
break;
}
- isamp = swlim;
- osamp = blck;
- st_rate_flow (sw->rate, sw->buf + pos, sw->hw->mix_buf + wpos, &isamp, &osamp);
- ret += isamp;
- swlim -= isamp;
- pos += isamp;
- live += osamp;
- wpos = (wpos + osamp) % hwsamples;
+ osamp = swlim;
+
+ if (audio_bug (AUDIO_FUNC, osamp < 0)) {
+ dolog ("osamp=%d\n", osamp);
+ }
+
+ st_rate_flow (sw->rate, src, dst, &isamp, &osamp);
+ swlim -= osamp;
+ rpos = (rpos + isamp) % hw->samples;
+ dst += osamp;
+ ret += osamp;
+ total += isamp;
}
- sw->wpos = wpos;
- sw->live = live;
- return ret << sw->shift;
+ sw->clip (buf, sw->conv_buf, ret);
+ sw->total_hw_samples_acquired += total;
+ return ret << sw->info.shift;
}
-int pcm_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+/*
+ * Hard voice (playback)
+ */
+static int audio_pcm_hw_find_min_out (HWVoiceOut *hw, int *nb_livep)
{
- int sign = 0, bits = 8;
+ SWVoiceOut *sw;
+ int m = INT_MAX;
+ int nb_live = 0;
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (sw->active || !sw->empty) {
+ m = audio_MIN (m, sw->total_hw_samples_mixed);
+ nb_live += 1;
+ }
+ }
+
+ *nb_livep = nb_live;
+ return m;
+}
- pcm_hw_fini (hw);
- ldebug ("pcm_hw_init: %d %d %d\n", freq, nchannels, fmt);
- if (hw->pcm_ops->init (hw, freq, nchannels, fmt)) {
- memset (hw, 0, audio_state.drv->voice_size);
+static void audio_pcm_hw_free_resources_out (HWVoiceOut *hw)
+{
+ if (hw->mix_buf) {
+ qemu_free (hw->mix_buf);
+ }
+
+ hw->mix_buf = NULL;
+}
+
+static int audio_pcm_hw_alloc_resources_out (HWVoiceOut *hw)
+{
+ hw->mix_buf = qemu_mallocz (hw->samples * sizeof (st_sample_t));
+ if (!hw->mix_buf) {
return -1;
}
- switch (hw->fmt) {
- case AUD_FMT_S8:
- sign = 1;
- case AUD_FMT_U8:
- break;
+ return 0;
+}
- case AUD_FMT_S16:
- sign = 1;
- case AUD_FMT_U16:
- bits = 16;
- break;
+static int audio_pcm_hw_init_out (HWVoiceOut *hw, int freq,
+ int nchannels, audfmt_e fmt)
+{
+ audio_pcm_hw_fini_out (hw);
+ if (hw->pcm_ops->init_out (hw, freq, nchannels, fmt)) {
+ memset (hw, 0, audio_state.drv->voice_size_out);
+ return -1;
}
- hw->nb_voices = 0;
+ LIST_INIT (&hw->sw_head);
hw->active = 1;
- hw->shift = (hw->nchannels == 2) + (bits == 16);
- hw->bytes_per_second = hw->freq << hw->shift;
- hw->align = (1 << hw->shift) - 1;
- hw->samples = hw->bufsize >> hw->shift;
- hw->clip = mixeng_clip[hw->nchannels == 2][sign][bits == 16];
- if (pcm_hw_alloc_resources (hw)) {
- pcm_hw_fini (hw);
+ hw->samples = hw->bufsize >> hw->info.shift;
+ hw->clip =
+ mixeng_clip
+ [nchannels == 2]
+ [hw->info.sign]
+ [hw->info.swap_endian]
+ [hw->info.bits == 16];
+ if (audio_pcm_hw_alloc_resources_out (hw)) {
+ audio_pcm_hw_fini_out (hw);
return -1;
}
return 0;
}
-static int dist (void *hw)
+int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live)
{
- if (hw) {
- return (((uint8_t *) hw - (uint8_t *) hw_voices)
- / audio_state.drv->voice_size) + 1;
+ int smin;
+
+ smin = audio_pcm_hw_find_min_out (hw, nb_live);
+
+ if (!*nb_live) {
+ return 0;
}
else {
- return 0;
+ int live = smin;
+
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
+ }
+ return live;
}
}
-#define ADVANCE(hw) \
- ((hw) ? advance (hw, audio_state.drv->voice_size) : hw_voices)
-
-HWVoice *pcm_hw_find_any (HWVoice *hw)
+int audio_pcm_hw_get_live_out (HWVoiceOut *hw)
{
- int i = dist (hw);
- for (; i < audio_state.nb_hw_voices; i++) {
- hw = ADVANCE (hw);
- return hw;
+ int nb_live;
+ int live;
+
+ live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ return 0;
}
- return NULL;
+ return live;
}
-HWVoice *pcm_hw_find_any_active (HWVoice *hw)
+/*
+ * Soft voice (playback)
+ */
+static void audio_pcm_sw_free_resources_out (SWVoiceOut *sw)
{
- int i = dist (hw);
- for (; i < audio_state.nb_hw_voices; i++) {
- hw = ADVANCE (hw);
- if (hw->active)
- return hw;
+ if (sw->buf) {
+ qemu_free (sw->buf);
}
- return NULL;
-}
-HWVoice *pcm_hw_find_any_active_enabled (HWVoice *hw)
-{
- int i = dist (hw);
- for (; i < audio_state.nb_hw_voices; i++) {
- hw = ADVANCE (hw);
- if (hw->active && hw->enabled)
- return hw;
+ if (sw->rate) {
+ st_rate_stop (sw->rate);
}
- return NULL;
+
+ sw->buf = NULL;
+ sw->rate = NULL;
}
-HWVoice *pcm_hw_find_any_passive (HWVoice *hw)
+static int audio_pcm_sw_alloc_resources_out (SWVoiceOut *sw)
{
- int i = dist (hw);
- for (; i < audio_state.nb_hw_voices; i++) {
- hw = ADVANCE (hw);
- if (!hw->active)
- return hw;
+ sw->buf = qemu_mallocz (sw->hw->samples * sizeof (st_sample_t));
+ if (!sw->buf) {
+ return -1;
+ }
+
+ sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq);
+ if (!sw->rate) {
+ qemu_free (sw->buf);
+ sw->buf = NULL;
+ return -1;
}
- return NULL;
+ return 0;
}
-HWVoice *pcm_hw_find_specific (HWVoice *hw, int freq,
- int nchannels, audfmt_e fmt)
+static int audio_pcm_sw_init_out (SWVoiceOut *sw, HWVoiceOut *hw,
+ const char *name, int freq,
+ int nchannels, audfmt_e fmt)
{
- while ((hw = pcm_hw_find_any_active (hw))) {
- if (hw->freq == freq &&
- hw->nchannels == nchannels &&
- hw->fmt == fmt)
- return hw;
- }
- return NULL;
+ audio_pcm_init_info (&sw->info, freq, nchannels, fmt,
+ /* None of the cards emulated by QEMU are big-endian
+ hence following shortcut */
+ audio_need_to_swap_endian (0));
+ sw->hw = hw;
+ sw->empty = 1;
+ sw->active = 0;
+ sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq;
+ sw->total_hw_samples_mixed = 0;
+
+ sw->conv =
+ mixeng_conv
+ [nchannels == 2]
+ [sw->info.sign]
+ [sw->info.swap_endian]
+ [sw->info.bits == 16];
+ sw->name = qemu_strdup (name);
+
+ audio_pcm_sw_free_resources_out (sw);
+ return audio_pcm_sw_alloc_resources_out (sw);
}
-HWVoice *pcm_hw_add (int freq, int nchannels, audfmt_e fmt)
+int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int size)
{
- HWVoice *hw;
+ int hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck;
+ int ret = 0, pos = 0, total = 0;
+
+ if (!sw) {
+ return size;
+ }
- if (audio_state.fixed_format) {
- freq = audio_state.fixed_freq;
- nchannels = audio_state.fixed_channels;
- fmt = audio_state.fixed_fmt;
+ hwsamples = sw->hw->samples;
+
+ live = sw->total_hw_samples_mixed;
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hwsamples)){
+ dolog ("live=%d hw->samples=%d\n", live, hwsamples);
+ return 0;
}
- hw = pcm_hw_find_specific (NULL, freq, nchannels, fmt);
+ if (live == hwsamples) {
+ return 0;
+ }
- if (hw)
- return hw;
+ wpos = (sw->hw->rpos + live) % hwsamples;
+ samples = size >> sw->info.shift;
- hw = pcm_hw_find_any_passive (NULL);
- if (hw) {
- hw->pcm_ops = audio_state.drv->pcm_ops;
- if (!hw->pcm_ops)
- return NULL;
+ dead = hwsamples - live;
+ swlim = ((int64_t) dead << 32) / sw->ratio;
+ swlim = audio_MIN (swlim, samples);
+ if (swlim) {
+ sw->conv (sw->buf, buf, swlim, &sw->vol);
+ }
- if (pcm_hw_init (hw, freq, nchannels, fmt)) {
- pcm_hw_gc (hw);
- return NULL;
+ while (swlim) {
+ dead = hwsamples - live;
+ left = hwsamples - wpos;
+ blck = audio_MIN (dead, left);
+ if (!blck) {
+ break;
}
- else
- return hw;
+ isamp = swlim;
+ osamp = blck;
+ st_rate_flow_mix (
+ sw->rate,
+ sw->buf + pos,
+ sw->hw->mix_buf + wpos,
+ &isamp,
+ &osamp
+ );
+ ret += isamp;
+ swlim -= isamp;
+ pos += isamp;
+ live += osamp;
+ wpos = (wpos + osamp) % hwsamples;
+ total += osamp;
}
- return pcm_hw_find_any (NULL);
+ sw->total_hw_samples_mixed += total;
+ sw->empty = sw->total_hw_samples_mixed == 0;
+
+#ifdef DEBUG_OUT
+ dolog (
+ "%s: write size %d ret %d total sw %d, hw %d\n",
+ sw->name,
+ size >> sw->info.shift,
+ ret,
+ sw->total_hw_samples_mixed,
+ sw->hw->total_samples_played
+ );
+#endif
+
+ return ret << sw->info.shift;
}
-int pcm_hw_add_sw (HWVoice *hw, SWVoice *sw)
+#ifdef DEBUG_AUDIO
+static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)
{
- SWVoice **pvoice = qemu_mallocz ((hw->nb_voices + 1) * sizeof (sw));
- if (!pvoice)
- return -1;
-
- memcpy (pvoice, hw->pvoice, hw->nb_voices * sizeof (sw));
- qemu_free (hw->pvoice);
- hw->pvoice = pvoice;
- hw->pvoice[hw->nb_voices++] = sw;
- return 0;
+ dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",
+ cap, info->bits, info->sign, info->freq, info->nchannels);
}
+#endif
-int pcm_hw_del_sw (HWVoice *hw, SWVoice *sw)
-{
- int i, j;
- if (hw->nb_voices > 1) {
- SWVoice **pvoice = qemu_mallocz ((hw->nb_voices - 1) * sizeof (sw));
+#define DAC
+#include "audio_template.h"
+#undef DAC
+#include "audio_template.h"
- if (!pvoice) {
- dolog ("Can not maintain consistent state (not enough memory)\n");
- return -1;
- }
+int AUD_write (SWVoiceOut *sw, void *buf, int size)
+{
+ int bytes;
- for (i = 0, j = 0; i < hw->nb_voices; i++) {
- if (j >= hw->nb_voices - 1) {
- dolog ("Can not maintain consistent state "
- "(invariant violated)\n");
- return -1;
- }
- if (hw->pvoice[i] != sw)
- pvoice[j++] = hw->pvoice[i];
- }
- qemu_free (hw->pvoice);
- hw->pvoice = pvoice;
- hw->nb_voices -= 1;
+ if (!sw) {
+ /* XXX: Consider options */
+ return size;
}
- else {
- qemu_free (hw->pvoice);
- hw->pvoice = NULL;
- hw->nb_voices = 0;
+
+ if (!sw->hw->enabled) {
+ dolog ("Writing to disabled voice %s\n", sw->name);
+ return 0;
}
- return 0;
+
+ bytes = sw->hw->pcm_ops->write (sw, buf, size);
+ return bytes;
}
-SWVoice *pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt)
+int AUD_read (SWVoiceIn *sw, void *buf, int size)
{
- SWVoice *sw;
- HWVoice *hw;
-
- sw = qemu_mallocz (sizeof (*sw));
- if (!sw)
- goto err1;
-
- hw = pcm_hw_add (freq, nchannels, fmt);
- if (!hw)
- goto err2;
+ int bytes;
- if (pcm_hw_add_sw (hw, sw))
- goto err3;
+ if (!sw) {
+ /* XXX: Consider options */
+ return size;
+ }
- if (pcm_sw_init (sw, hw, freq, nchannels, fmt))
- goto err4;
+ if (!sw->hw->enabled) {
+ dolog ("Reading from disabled voice %s\n", sw->name);
+ return 0;
+ }
- return sw;
+ bytes = sw->hw->pcm_ops->read (sw, buf, size);
+ return bytes;
+}
-err4:
- pcm_hw_del_sw (hw, sw);
-err3:
- pcm_hw_gc (hw);
-err2:
- qemu_free (sw);
-err1:
- return NULL;
+int AUD_get_buffer_size_out (SWVoiceOut *sw)
+{
+ return sw->hw->bufsize;
}
-SWVoice *AUD_open (SWVoice *sw, const char *name,
- int freq, int nchannels, audfmt_e fmt)
+void AUD_set_active_out (SWVoiceOut *sw, int on)
{
- if (!audio_state.drv) {
- return NULL;
- }
+ HWVoiceOut *hw;
- if (sw && freq == sw->freq && sw->nchannels == nchannels && sw->fmt == fmt) {
- return sw;
+ if (!sw) {
+ return;
}
- if (sw) {
- ldebug ("Different format %s %d %d %d\n",
- name,
- sw->freq == freq,
- sw->nchannels == nchannels,
- sw->fmt == fmt);
- }
+ hw = sw->hw;
+ if (sw->active != on) {
+ SWVoiceOut *temp_sw;
- if (nchannels != 1 && nchannels != 2) {
- dolog ("Bogus channel count %d for voice %s\n", nchannels, name);
- return NULL;
- }
+ if (on) {
+ int total;
- if (!audio_state.fixed_format && sw) {
- pcm_sw_fini (sw);
- pcm_hw_del_sw (sw->hw, sw);
- pcm_hw_gc (sw->hw);
- if (sw->name) {
- qemu_free (sw->name);
- sw->name = NULL;
- }
- qemu_free (sw);
- sw = NULL;
- }
+ hw->pending_disable = 0;
+ if (!hw->enabled) {
+ hw->enabled = 1;
+ hw->pcm_ops->ctl_out (hw, VOICE_ENABLE);
+ }
- if (sw) {
- HWVoice *hw = sw->hw;
- if (!hw) {
- dolog ("Internal logic error voice %s has no hardware store\n",
- name);
- return sw;
+ if (sw->empty) {
+ total = 0;
+ }
}
+ else {
+ if (hw->enabled) {
+ int nb_active = 0;
- if (pcm_sw_init (sw, hw, freq, nchannels, fmt)) {
- pcm_sw_fini (sw);
- pcm_hw_del_sw (hw, sw);
- pcm_hw_gc (hw);
- if (sw->name) {
- qemu_free (sw->name);
- sw->name = NULL;
+ for (temp_sw = hw->sw_head.lh_first; temp_sw;
+ temp_sw = temp_sw->entries.le_next) {
+ nb_active += temp_sw->active != 0;
+ }
+
+ hw->pending_disable = nb_active == 1;
}
- qemu_free (sw);
- return NULL;
}
+ sw->active = on;
}
- else {
- sw = pcm_create_voice_pair (freq, nchannels, fmt);
- if (!sw) {
- dolog ("Failed to create voice %s\n", name);
- return NULL;
- }
+}
+
+void AUD_set_active_in (SWVoiceIn *sw, int on)
+{
+ HWVoiceIn *hw;
+
+ if (!sw) {
+ return;
}
- if (sw->name) {
- qemu_free (sw->name);
- sw->name = NULL;
+ hw = sw->hw;
+ if (sw->active != on) {
+ SWVoiceIn *temp_sw;
+
+ if (on) {
+ if (!hw->enabled) {
+ hw->enabled = 1;
+ hw->pcm_ops->ctl_in (hw, VOICE_ENABLE);
+ }
+ sw->total_hw_samples_acquired = hw->total_samples_captured;
+ }
+ else {
+ if (hw->enabled) {
+ int nb_active = 0;
+
+ for (temp_sw = hw->sw_head.lh_first; temp_sw;
+ temp_sw = temp_sw->entries.le_next) {
+ nb_active += temp_sw->active != 0;
+ }
+
+ if (nb_active == 1) {
+ hw->enabled = 0;
+ hw->pcm_ops->ctl_in (hw, VOICE_DISABLE);
+ }
+ }
+ }
+ sw->active = on;
}
- sw->name = qemu_strdup (name);
- return sw;
}
-void AUD_close (SWVoice *sw)
+static int audio_get_avail (SWVoiceIn *sw)
{
- if (!sw)
- return;
+ int live;
+
+ if (!sw) {
+ return 0;
+ }
- pcm_sw_fini (sw);
- pcm_hw_del_sw (sw->hw, sw);
- pcm_hw_gc (sw->hw);
- if (sw->name) {
- qemu_free (sw->name);
- sw->name = NULL;
+ live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
+ dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ return 0;
}
- qemu_free (sw);
+
+ ldebug (
+ "%s: get_avail live %d ret %lld\n",
+ sw->name,
+ live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift
+ );
+
+ return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;
}
-int AUD_write (SWVoice *sw, void *buf, int size)
+static int audio_get_free (SWVoiceOut *sw)
{
- int bytes;
+ int live, dead;
- if (!sw->hw->enabled)
- dolog ("Writing to disabled voice %s\n", sw->name);
- bytes = sw->hw->pcm_ops->write (sw, buf, size);
- return bytes;
+ if (!sw) {
+ return 0;
+ }
+
+ live = sw->total_hw_samples_mixed;
+
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > sw->hw->samples)) {
+ dolog ("live=%d sw->hw->samples=%d\n", live, sw->hw->samples);
+ }
+
+ dead = sw->hw->samples - live;
+
+#ifdef DEBUG_OUT
+ dolog ("%s: get_free live %d dead %d ret %lld\n",
+ sw->name,
+ live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);
+#endif
+
+ return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;
}
-void AUD_run (void)
+static void audio_run_out (void)
{
- HWVoice *hw = NULL;
+ HWVoiceOut *hw = NULL;
+ SWVoiceOut *sw;
- while ((hw = pcm_hw_find_any_active_enabled (hw))) {
- int i;
- if (hw->pending_disable && pcm_hw_get_live (hw) <= 0) {
+ while ((hw = audio_pcm_hw_find_any_active_enabled_out (hw))) {
+ int played;
+ int live, free, nb_live;
+
+ live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
+ if (!nb_live) {
+ live = 0;
+ }
+ if (audio_bug (AUDIO_FUNC, live < 0 || live > hw->samples)) {
+ dolog ("live=%d hw->samples=%d\n", live, hw->samples);
+ }
+
+ if (hw->pending_disable && !nb_live) {
+#ifdef DEBUG_OUT
+ dolog ("Disabling voice\n");
+#endif
hw->enabled = 0;
- hw->pcm_ops->ctl (hw, VOICE_DISABLE);
- for (i = 0; i < hw->nb_voices; i++) {
- hw->pvoice[i]->live = 0;
- /* hw->pvoice[i]->old_ticks = 0; */
+ hw->pending_disable = 0;
+ hw->pcm_ops->ctl_out (hw, VOICE_DISABLE);
+ continue;
+ }
+
+ if (!live) {
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ if (sw->active) {
+ free = audio_get_free (sw);
+ if (free > 0) {
+ sw->callback.fn (sw->callback.opaque, free);
+ }
+ }
}
continue;
}
- hw->pcm_ops->run (hw);
- assert (hw->rpos < hw->samples);
- for (i = 0; i < hw->nb_voices; i++) {
- SWVoice *sw = hw->pvoice[i];
- if (!sw->active && !sw->live && sw->old_ticks) {
- int64_t delta = qemu_get_clock (vm_clock) - sw->old_ticks;
- if (delta > audio_state.ticks_threshold) {
- ldebug ("resetting old_ticks for %s\n", sw->name);
- sw->old_ticks = 0;
+ played = hw->pcm_ops->run_out (hw);
+ if (audio_bug (AUDIO_FUNC, hw->rpos >= hw->samples)) {
+ dolog ("hw->rpos=%d hw->samples=%d played=%d\n",
+ hw->rpos, hw->samples, played);
+ hw->rpos = 0;
+ }
+
+#ifdef DEBUG_OUT
+ dolog ("played = %d total %d\n", played, hw->total_samples_played);
+#endif
+
+ if (played) {
+ hw->ts_helper += played;
+ }
+
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ again:
+ if (!sw->active && sw->empty) {
+ continue;
+ }
+
+ if (audio_bug (AUDIO_FUNC, played > sw->total_hw_samples_mixed)) {
+ dolog ("played=%d sw->total_hw_samples_mixed=%d\n",
+ played, sw->total_hw_samples_mixed);
+ played = sw->total_hw_samples_mixed;
+ }
+
+ sw->total_hw_samples_mixed -= played;
+
+ if (!sw->total_hw_samples_mixed) {
+ sw->empty = 1;
+
+ if (!sw->active && !sw->callback.fn) {
+ SWVoiceOut *temp = sw->entries.le_next;
+
+#ifdef DEBUG_PLIVE
+ dolog ("Finishing with old voice\n");
+#endif
+ AUD_close_out (sw);
+ sw = temp;
+ if (sw) {
+ goto again;
+ }
+ else {
+ break;
+ }
+ }
+ }
+
+ if (sw->active) {
+ free = audio_get_free (sw);
+ if (free > 0) {
+ sw->callback.fn (sw->callback.opaque, free);
}
}
}
}
}
-int AUD_get_free (SWVoice *sw)
+static void audio_run_in (void)
{
- int free;
+ HWVoiceIn *hw = NULL;
- if (!sw)
- return 4096;
+ while ((hw = audio_pcm_hw_find_any_active_enabled_in (hw))) {
+ SWVoiceIn *sw;
+ int captured, min;
- free = ((sw->hw->samples - sw->live) << sw->hw->shift) * sw->ratio
- / INT_MAX;
+ captured = hw->pcm_ops->run_in (hw);
- free &= ~sw->hw->align;
- if (!free) return 0;
+ min = audio_pcm_hw_find_min_in (hw);
+ hw->total_samples_captured += captured - min;
+ hw->ts_helper += captured;
- return free;
-}
+ for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) {
+ sw->total_hw_samples_acquired -= min;
-int AUD_get_buffer_size (SWVoice *sw)
-{
- return sw->hw->bufsize;
-}
+ if (sw->active) {
+ int avail;
-void AUD_adjust (SWVoice *sw, int bytes)
-{
- if (!sw)
- return;
- sw->old_ticks += (ticks_per_sec * (int64_t) bytes) / sw->bytes_per_second;
+ avail = audio_get_avail (sw);
+ if (avail > 0) {
+ sw->callback.fn (sw->callback.opaque, avail);
+ }
+ }
+ }
+ }
}
-void AUD_reset (SWVoice *sw)
-{
- sw->active = 0;
- sw->old_ticks = 0;
-}
+static struct audio_option audio_options[] = {
+ /* DAC */
+ {"DAC_FIXED_SETTINGS", AUD_OPT_BOOL, &audio_state.fixed_settings_out,
+ "Use fixed settings for host DAC", NULL, 0},
-int AUD_calc_elapsed (SWVoice *sw)
-{
- int64_t now, delta, bytes;
- int dead, swlim;
+ {"DAC_FIXED_FREQ", AUD_OPT_INT, &audio_state.fixed_freq_out,
+ "Frequency for fixed host DAC", NULL, 0},
- if (!sw)
- return 0;
+ {"DAC_FIXED_FMT", AUD_OPT_FMT, &audio_state.fixed_fmt_out,
+ "Format for fixed host DAC", NULL, 0},
- now = qemu_get_clock (vm_clock);
- delta = now - sw->old_ticks;
- bytes = (delta * sw->bytes_per_second) / ticks_per_sec;
- if (delta < 0) {
- dolog ("whoops delta(<0)=%lld\n", delta);
- return 0;
- }
+ {"DAC_FIXED_CHANNELS", AUD_OPT_INT, &audio_state.fixed_channels_out,
+ "Number of channels for fixed DAC (1 - mono, 2 - stereo)", NULL, 0},
- dead = sw->hw->samples - sw->live;
- swlim = ((dead * (int64_t) INT_MAX) / sw->ratio);
+ {"DAC_VOICES", AUD_OPT_INT, &audio_state.nb_hw_voices_out,
+ "Number of voices for DAC", NULL, 0},
- if (bytes > swlim) {
- return swlim;
- }
- else {
- return bytes;
- }
-}
+ /* ADC */
+ {"ADC_FIXED_SETTINGS", AUD_OPT_BOOL, &audio_state.fixed_settings_out,
+ "Use fixed settings for host ADC", NULL, 0},
-void AUD_enable (SWVoice *sw, int on)
-{
- int i;
- HWVoice *hw;
+ {"ADC_FIXED_FREQ", AUD_OPT_INT, &audio_state.fixed_freq_out,
+ "Frequency for fixed ADC", NULL, 0},
- if (!sw)
- return;
+ {"ADC_FIXED_FMT", AUD_OPT_FMT, &audio_state.fixed_fmt_out,
+ "Format for fixed ADC", NULL, 0},
- hw = sw->hw;
- if (on) {
- if (!sw->live)
- sw->wpos = sw->hw->rpos;
- if (!sw->old_ticks) {
- sw->old_ticks = qemu_get_clock (vm_clock);
+ {"ADC_FIXED_CHANNELS", AUD_OPT_INT, &audio_state.fixed_channels_in,
+ "Number of channels for fixed ADC (1 - mono, 2 - stereo)", NULL, 0},
+
+ {"ADC_VOICES", AUD_OPT_INT, &audio_state.nb_hw_voices_out,
+ "Number of voices for ADC", NULL, 0},
+
+ /* Misc */
+ {"TIMER_PERIOD", AUD_OPT_INT, &audio_state.period.usec,
+ "Timer period in microseconds (0 - try lowest possible)", NULL, 0},
+
+ {"PLIVE", AUD_OPT_BOOL, &audio_state.plive,
+ "(undocumented)", NULL, 0},
+
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+void AUD_help (void)
+{
+ size_t i;
+
+ audio_process_options ("AUDIO", audio_options);
+ for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+ struct audio_driver *d = drvtab[i];
+ if (d->options) {
+ audio_process_options (d->name, d->options);
}
}
- if (sw->active != on) {
- if (on) {
- hw->pending_disable = 0;
- if (!hw->enabled) {
- hw->enabled = 1;
- for (i = 0; i < hw->nb_voices; i++) {
- ldebug ("resetting voice\n");
- sw = hw->pvoice[i];
- sw->old_ticks = qemu_get_clock (vm_clock);
- }
- hw->pcm_ops->ctl (hw, VOICE_ENABLE);
- }
+ printf ("Audio options:\n");
+ audio_print_options ("AUDIO", audio_options);
+ printf ("\n");
+
+ printf ("Available drivers:\n");
+
+ for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+ struct audio_driver *d = drvtab[i];
+
+ printf ("Name: %s\n", d->name);
+ printf ("Description: %s\n", d->descr);
+
+ switch (d->max_voices_out) {
+ case 0:
+ printf ("Does not support DAC\n");
+ break;
+ case 1:
+ printf ("One DAC voice\n");
+ break;
+ case INT_MAX:
+ printf ("Theoretically supports many DAC voices\n");
+ break;
+ default:
+ printf ("Theoretically supports upto %d DAC voices\n",
+ d->max_voices_out);
+ break;
}
- else {
- if (hw->enabled && !hw->pending_disable) {
- int nb_active = 0;
- for (i = 0; i < hw->nb_voices; i++) {
- nb_active += hw->pvoice[i]->active != 0;
- }
- if (nb_active == 1) {
- hw->pending_disable = 1;
- }
- }
+ switch (d->max_voices_in) {
+ case 0:
+ printf ("Does not support ADC\n");
+ break;
+ case 1:
+ printf ("One ADC voice\n");
+ break;
+ case INT_MAX:
+ printf ("Theoretically supports many ADC voices\n");
+ break;
+ default:
+ printf ("Theoretically supports upto %d ADC voices\n",
+ d->max_voices_in);
+ break;
}
- sw->active = on;
+
+ if (d->options) {
+ printf ("Options:\n");
+ audio_print_options (d->name, d->options);
+ }
+ else {
+ printf ("No options\n");
+ }
+ printf ("\n");
}
-}
-static struct audio_output_driver *drvtab[] = {
-#ifdef CONFIG_OSS
- &oss_output_driver,
-#endif
-#ifdef CONFIG_FMOD
- &fmod_output_driver,
-#endif
-#ifdef CONFIG_SDL
- &sdl_output_driver,
-#endif
- &no_output_driver,
-#ifdef USE_WAV_AUDIO
- &wav_output_driver,
+ printf (
+ "Options are settable through environment variables.\n"
+ "Example:\n"
+#ifdef _WIN32
+ " set QEMU_AUDIO_DRV=wav\n"
+ " set QEMU_WAV_PATH=c:/tune.wav\n"
+#else
+ " export QEMU_AUDIO_DRV=wav\n"
+ " export QEMU_WAV_PATH=$HOME/tune.wav\n"
+ "(for csh replace export with setenv in the above)\n"
#endif
-};
+ " qemu ...\n\n"
+ );
+}
+
+void audio_timer (void *opaque)
+{
+ AudioState *s = opaque;
-static int voice_init (struct audio_output_driver *drv)
+ audio_run_out ();
+ audio_run_in ();
+
+ qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + s->period.ticks);
+}
+
+static int audio_driver_init (struct audio_driver *drv)
{
+ if (drv->options) {
+ audio_process_options (drv->name, drv->options);
+ }
audio_state.opaque = drv->init ();
+
if (audio_state.opaque) {
- if (audio_state.nb_hw_voices > drv->max_voices) {
- dolog ("`%s' does not support %d multiple hardware channels\n"
- "Resetting to %d\n",
- drv->name, audio_state.nb_hw_voices, drv->max_voices);
- audio_state.nb_hw_voices = drv->max_voices;
+ int i;
+ HWVoiceOut *hwo;
+ HWVoiceIn *hwi;
+
+ if (audio_state.nb_hw_voices_out > drv->max_voices_out) {
+ if (!drv->max_voices_out) {
+ dolog ("`%s' does not support DAC\n", drv->name);
+ }
+ else {
+ dolog (
+ "`%s' does not support %d multiple DAC voicess\n"
+ "Resetting to %d\n",
+ drv->name,
+ audio_state.nb_hw_voices_out,
+ drv->max_voices_out
+ );
+ }
+ audio_state.nb_hw_voices_out = drv->max_voices_out;
+ }
+
+ LIST_INIT (&hw_head_out);
+ hwo = qemu_mallocz (audio_state.nb_hw_voices_out * drv->voice_size_out);
+ if (!hwo) {
+ dolog (
+ "Not enough memory for %d `%s' DAC voices (each %d bytes)\n",
+ audio_state.nb_hw_voices_out,
+ drv->name,
+ drv->voice_size_out
+ );
+ drv->fini (audio_state.opaque);
+ return -1;
}
- hw_voices = qemu_mallocz (audio_state.nb_hw_voices * drv->voice_size);
- if (hw_voices) {
- audio_state.drv = drv;
- return 1;
+
+ for (i = 0; i < audio_state.nb_hw_voices_out; ++i) {
+ LIST_INSERT_HEAD (&hw_head_out, hwo, entries);
+ hwo = advance (hwo, drv->voice_size_out);
}
- else {
- dolog ("Not enough memory for %d `%s' voices (each %d bytes)\n",
- audio_state.nb_hw_voices, drv->name, drv->voice_size);
+
+ if (!drv->voice_size_in && drv->max_voices_in) {
+ ldebug ("warning: No ADC voice size defined for `%s'\n",
+ drv->name);
+ drv->max_voices_in = 0;
+ }
+
+ if (!drv->voice_size_out && drv->max_voices_out) {
+ ldebug ("warning: No DAC voice size defined for `%s'\n",
+ drv->name);
+ }
+
+ if (drv->voice_size_in && !drv->max_voices_in) {
+ ldebug ("warning: ADC voice size is %d for ADC less driver `%s'\n",
+ drv->voice_size_out, drv->name);
+ }
+
+ if (drv->voice_size_out && !drv->max_voices_out) {
+ ldebug ("warning: DAC voice size is %d for DAC less driver `%s'\n",
+ drv->voice_size_in, drv->name);
+ }
+
+ if (audio_state.nb_hw_voices_in > drv->max_voices_in) {
+ if (!drv->max_voices_in) {
+ ldebug ("`%s' does not support ADC\n", drv->name);
+ }
+ else {
+ dolog (
+ "`%s' does not support %d multiple ADC voices\n"
+ "Resetting to %d\n",
+ drv->name,
+ audio_state.nb_hw_voices_in,
+ drv->max_voices_in
+ );
+ }
+ audio_state.nb_hw_voices_in = drv->max_voices_in;
+ }
+
+ LIST_INIT (&hw_head_in);
+ hwi = qemu_mallocz (audio_state.nb_hw_voices_in * drv->voice_size_in);
+ if (!hwi) {
+ dolog (
+ "Not enough memory for %d `%s' ADC voices (each %d bytes)\n",
+ audio_state.nb_hw_voices_in,
+ drv->name,
+ drv->voice_size_in
+ );
+ qemu_free (hwo);
drv->fini (audio_state.opaque);
- return 0;
+ return -1;
}
+
+ for (i = 0; i < audio_state.nb_hw_voices_in; ++i) {
+ LIST_INSERT_HEAD (&hw_head_in, hwi, entries);
+ hwi = advance (hwi, drv->voice_size_in);
+ }
+
+ audio_state.drv = drv;
+ return 0;
}
else {
- dolog ("Could not init `%s' audio\n", drv->name);
- return 0;
+ dolog ("Could not init `%s' audio driver\n", drv->name);
+ return -1;
}
}
static void audio_vm_stop_handler (void *opaque, int reason)
{
- HWVoice *hw = NULL;
+ HWVoiceOut *hwo = NULL;
+ HWVoiceIn *hwi = NULL;
+ int op = reason ? VOICE_ENABLE : VOICE_DISABLE;
+
+ (void) opaque;
+ while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+ if (!hwo->pcm_ops) {
+ continue;
+ }
+
+ if (hwo->enabled != reason) {
+ hwo->pcm_ops->ctl_out (hwo, op);
+ }
+ }
- while ((hw = pcm_hw_find_any (hw))) {
- if (!hw->pcm_ops)
+ while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+ if (!hwi->pcm_ops) {
continue;
+ }
- hw->pcm_ops->ctl (hw, reason ? VOICE_ENABLE : VOICE_DISABLE);
+ if (hwi->enabled != reason) {
+ hwi->pcm_ops->ctl_in (hwi, op);
+ }
}
}
static void audio_atexit (void)
{
- HWVoice *hw = NULL;
+ HWVoiceOut *hwo = NULL;
+ HWVoiceIn *hwi = NULL;
+
+ while ((hwo = audio_pcm_hw_find_any_out (hwo))) {
+ if (!hwo->pcm_ops) {
+ continue;
+ }
+
+ if (hwo->enabled) {
+ hwo->pcm_ops->ctl_out (hwo, VOICE_DISABLE);
+ }
+ hwo->pcm_ops->fini_out (hwo);
+ }
- while ((hw = pcm_hw_find_any (hw))) {
- if (!hw->pcm_ops)
+ while ((hwi = audio_pcm_hw_find_any_in (hwi))) {
+ if (!hwi->pcm_ops) {
continue;
+ }
- hw->pcm_ops->ctl (hw, VOICE_DISABLE);
- hw->pcm_ops->fini (hw);
+ if (hwi->enabled) {
+ hwi->pcm_ops->ctl_in (hwi, VOICE_DISABLE);
+ }
+ hwi->pcm_ops->fini_in (hwi);
}
audio_state.drv->fini (audio_state.opaque);
}
static void audio_save (QEMUFile *f, void *opaque)
{
+ (void) f;
+ (void) opaque;
}
static int audio_load (QEMUFile *f, void *opaque, int version_id)
{
- if (version_id != 1)
+ (void) f;
+ (void) opaque;
+
+ if (version_id != 1) {
return -EINVAL;
+ }
return 0;
}
void AUD_init (void)
{
- int i;
+ size_t i;
int done = 0;
const char *drvname;
+ AudioState *s = &audio_state;
- audio_state.fixed_format =
- !!audio_get_conf_int (QC_FIXED_FORMAT, audio_state.fixed_format);
- audio_state.fixed_freq =
- audio_get_conf_int (QC_FIXED_FREQ, audio_state.fixed_freq);
- audio_state.nb_hw_voices =
- audio_get_conf_int (QC_VOICES, audio_state.nb_hw_voices);
+ audio_process_options ("AUDIO", audio_options);
- if (audio_state.nb_hw_voices <= 0) {
- dolog ("Bogus number of voices %d, resetting to 1\n",
- audio_state.nb_hw_voices);
+ if (s->nb_hw_voices_out <= 0) {
+ dolog ("Bogus number of DAC voices %d\n",
+ s->nb_hw_voices_out);
+ s->nb_hw_voices_out = 1;
+ }
+
+ if (s->nb_hw_voices_in <= 0) {
+ dolog ("Bogus number of ADC voices %d\n",
+ s->nb_hw_voices_in);
+ s->nb_hw_voices_in = 1;
+ }
+
+ {
+ int def;
+ drvname = audio_get_conf_str ("QEMU_AUDIO_DRV", NULL, &def);
+ }
+
+ s->ts = qemu_new_timer (vm_clock, audio_timer, s);
+ if (!s->ts) {
+ dolog ("Can not create audio timer\n");
+ return;
}
- drvname = audio_get_conf_str (QC_AUDIO_DRV, NULL);
if (drvname) {
int found = 0;
+
for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
if (!strcmp (drvname, drvtab[i]->name)) {
- done = voice_init (drvtab[i]);
+ done = !audio_driver_init (drvtab[i]);
found = 1;
break;
}
}
+
if (!found) {
dolog ("Unknown audio driver `%s'\n", drvname);
+ dolog ("Run with -audio-help to list available drivers\n");
}
}
@@ -895,17 +1624,32 @@ void AUD_init (void)
if (!done) {
for (i = 0; !done && i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
- if (drvtab[i]->can_be_default)
- done = voice_init (drvtab[i]);
+ if (drvtab[i]->can_be_default) {
+ done = !audio_driver_init (drvtab[i]);
+ }
}
}
- audio_state.ticks_threshold = ticks_per_sec / 50;
- audio_state.freq_threshold = 100;
-
register_savevm ("audio", 0, 1, audio_save, audio_load, NULL);
if (!done) {
- dolog ("Can not initialize audio subsystem\n");
- voice_init (&no_output_driver);
+ if (audio_driver_init (&no_audio_driver)) {
+ dolog ("Can not initialize audio subsystem\n");
+ }
+ else {
+ dolog ("warning: using timer based audio emulation\n");
+ }
}
+
+ if (s->period.usec <= 0) {
+ if (s->period.usec < 0) {
+ dolog ("warning: timer period is negative - %d treating as zero\n",
+ s->period.usec);
+ }
+ s->period.ticks = 1;
+ }
+ else {
+ s->period.ticks = (ticks_per_sec * s->period.usec) / 1000000;
+ }
+
+ qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + s->period.ticks);
}
diff --git a/audio/audio.h b/audio/audio.h
index 7520383a47..6dd2fd22e3 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -1,8 +1,8 @@
/*
* QEMU Audio subsystem header
- *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- *
+ *
+ * Copyright (c) 2003-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -24,7 +24,7 @@
#ifndef QEMU_AUDIO_H
#define QEMU_AUDIO_H
-#include "mixeng.h"
+typedef void (*audio_callback_fn_t) (void *opaque, int avail);
typedef enum {
AUD_FMT_U8,
@@ -33,22 +33,60 @@ typedef enum {
AUD_FMT_S16
} audfmt_e;
-typedef struct SWVoice SWVoice;
+typedef struct SWVoiceOut SWVoiceOut;
+typedef struct SWVoiceIn SWVoiceIn;
+
+typedef struct QEMUAudioTimeStamp {
+ uint64_t old_ts;
+} QEMUAudioTimeStamp;
+
+void AUD_vlog (const char *cap, const char *fmt, va_list ap);
+void AUD_log (const char *cap, const char *fmt, ...)
+#ifdef __GNUC__
+ __attribute__ ((__format__ (__printf__, 2, 3)))
+#endif
+ ;
-SWVoice * AUD_open (SWVoice *sw, const char *name, int freq,
- int nchannels, audfmt_e fmt);
-void AUD_init (void);
-void AUD_log (const char *cap, const char *fmt, ...)
- __attribute__ ((__format__ (__printf__, 2, 3)));;
-void AUD_close (SWVoice *sw);
-int AUD_write (SWVoice *sw, void *pcm_buf, int size);
-void AUD_adjust (SWVoice *sw, int leftover);
-void AUD_reset (SWVoice *sw);
-int AUD_get_free (SWVoice *sw);
-int AUD_get_buffer_size (SWVoice *sw);
-void AUD_run (void);
-void AUD_enable (SWVoice *sw, int on);
-int AUD_calc_elapsed (SWVoice *sw);
+void AUD_init (void);
+void AUD_help (void);
+
+SWVoiceOut *AUD_open_out (
+ SWVoiceOut *sw,
+ const char *name,
+ void *callback_opaque,
+ audio_callback_fn_t callback_fn,
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ );
+void AUD_close_out (SWVoiceOut *sw);
+int AUD_write (SWVoiceOut *sw, void *pcm_buf, int size);
+int AUD_get_buffer_size_out (SWVoiceOut *sw);
+void AUD_set_active_out (SWVoiceOut *sw, int on);
+int AUD_is_active_out (SWVoiceOut *sw);
+void AUD_init_time_stamp_out (SWVoiceOut *sw,
+ QEMUAudioTimeStamp *ts);
+uint64_t AUD_time_stamp_get_elapsed_usec_out (SWVoiceOut *sw,
+ QEMUAudioTimeStamp *ts);
+
+SWVoiceIn *AUD_open_in (
+ SWVoiceIn *sw,
+ const char *name,
+ void *callback_opaque,
+ audio_callback_fn_t callback_fn,
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ );
+void AUD_close_in (SWVoiceIn *sw);
+int AUD_read (SWVoiceIn *sw, void *pcm_buf, int size);
+void AUD_adjust_in (SWVoiceIn *sw, int leftover);
+void AUD_set_active_in (SWVoiceIn *sw, int on);
+int AUD_is_active_in (SWVoiceIn *sw);
+void AUD_init_time_stamp_in (SWVoiceIn *sw,
+ QEMUAudioTimeStamp *ts);
+uint64_t AUD_time_stamp_get_elapsed_usec_in (SWVoiceIn *sw,
+ QEMUAudioTimeStamp *ts);
static inline void *advance (void *p, int incr)
{
@@ -59,7 +97,21 @@ static inline void *advance (void *p, int incr)
uint32_t popcount (uint32_t u);
inline uint32_t lsbindex (uint32_t u);
+#ifdef __GNUC__
+#define audio_MIN(a, b) ( __extension__ ({ \
+ __typeof (a) ta = a; \
+ __typeof (b) tb = b; \
+ ((ta)>(tb)?(tb):(ta)); \
+}))
+
+#define audio_MAX(a, b) ( __extension__ ({ \
+ __typeof (a) ta = a; \
+ __typeof (b) tb = b; \
+ ((ta)<(tb)?(tb):(ta)); \
+}))
+#else
#define audio_MIN(a, b) ((a)>(b)?(b):(a))
#define audio_MAX(a, b) ((a)<(b)?(b):(a))
+#endif
#endif /* audio.h */
diff --git a/audio/audio_int.h b/audio/audio_int.h
index 0be2a61662..9d288292a7 100644
--- a/audio/audio_int.h
+++ b/audio/audio_int.h
@@ -1,8 +1,8 @@
/*
* QEMU Audio subsystem header
- *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- *
+ *
+ * Copyright (c) 2003-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -24,140 +24,266 @@
#ifndef QEMU_AUDIO_INT_H
#define QEMU_AUDIO_INT_H
-#include "vl.h"
+#include "sys-queue.h"
+
+#ifdef CONFIG_COREAUDIO
+#define FLOAT_MIXENG
+/* #define RECIPROCAL */
+#endif
+#include "mixeng.h"
+
+int audio_bug (const char *funcname, int cond);
+
+struct audio_pcm_ops;
+
+typedef enum {
+ AUD_OPT_INT,
+ AUD_OPT_FMT,
+ AUD_OPT_STR,
+ AUD_OPT_BOOL
+} audio_option_tag_e;
+
+struct audio_option {
+ const char *name;
+ audio_option_tag_e tag;
+ void *valp;
+ const char *descr;
+ int *overridenp;
+ int overriden;
+};
+
+struct audio_callback {
+ void *opaque;
+ audio_callback_fn_t fn;
+};
-struct pcm_ops;
+struct audio_pcm_info {
+ int bits;
+ int sign;
+ int freq;
+ int nchannels;
+ int align;
+ int shift;
+ int bytes_per_second;
+ int swap_endian;
+};
-typedef struct HWVoice {
+typedef struct HWVoiceOut {
int active;
int enabled;
int pending_disable;
int valid;
- int freq;
+ struct audio_pcm_info info;
f_sample *clip;
- audfmt_e fmt;
- int nchannels;
-
- int align;
- int shift;
int rpos;
int bufsize;
+ uint64_t ts_helper;
- int bytes_per_second;
st_sample_t *mix_buf;
int samples;
- int64_t old_ticks;
- int nb_voices;
- struct SWVoice **pvoice;
- struct pcm_ops *pcm_ops;
-} HWVoice;
+ LIST_HEAD (sw_out_listhead, SWVoiceOut) sw_head;
+ struct audio_pcm_ops *pcm_ops;
+ LIST_ENTRY (HWVoiceOut) entries;
+} HWVoiceOut;
-extern struct pcm_ops no_pcm_ops;
-extern struct audio_output_driver no_output_driver;
+typedef struct HWVoiceIn {
+ int enabled;
+ int active;
+ struct audio_pcm_info info;
+
+ t_sample *conv;
-extern struct pcm_ops oss_pcm_ops;
-extern struct audio_output_driver oss_output_driver;
+ int wpos;
+ int bufsize;
+ int total_samples_captured;
+ uint64_t ts_helper;
-extern struct pcm_ops sdl_pcm_ops;
-extern struct audio_output_driver sdl_output_driver;
+ st_sample_t *conv_buf;
-extern struct pcm_ops wav_pcm_ops;
-extern struct audio_output_driver wav_output_driver;
+ int samples;
+ LIST_HEAD (sw_in_listhead, SWVoiceIn) sw_head;
+ struct audio_pcm_ops *pcm_ops;
+ LIST_ENTRY (HWVoiceIn) entries;
+} HWVoiceIn;
-extern struct pcm_ops fmod_pcm_ops;
-extern struct audio_output_driver fmod_output_driver;
+extern struct audio_driver no_audio_driver;
+extern struct audio_driver oss_audio_driver;
+extern struct audio_driver sdl_audio_driver;
+extern struct audio_driver wav_audio_driver;
+extern struct audio_driver fmod_audio_driver;
+extern struct audio_driver alsa_audio_driver;
+extern struct audio_driver coreaudio_audio_driver;
+extern struct audio_driver dsound_audio_driver;
+extern volume_t nominal_volume;
-struct audio_output_driver {
+struct audio_driver {
const char *name;
+ const char *descr;
+ struct audio_option *options;
void *(*init) (void);
void (*fini) (void *);
- struct pcm_ops *pcm_ops;
+ struct audio_pcm_ops *pcm_ops;
int can_be_default;
- int max_voices;
- int voice_size;
+ int max_voices_out;
+ int max_voices_in;
+ int voice_size_out;
+ int voice_size_in;
};
typedef struct AudioState {
- int fixed_format;
- int fixed_freq;
- int fixed_channels;
- int fixed_fmt;
- int nb_hw_voices;
- int64_t ticks_threshold;
- int freq_threshold;
+ int fixed_settings_out;
+ int fixed_freq_out;
+ int fixed_channels_out;
+ int fixed_fmt_out;
+ int nb_hw_voices_out;
+ int greedy_out;
+
+ int fixed_settings_in;
+ int fixed_freq_in;
+ int fixed_channels_in;
+ int fixed_fmt_in;
+ int nb_hw_voices_in;
+ int greedy_in;
+
void *opaque;
- struct audio_output_driver *drv;
-} AudioState;
-extern AudioState audio_state;
+ struct audio_driver *drv;
-struct SWVoice {
- int freq;
- audfmt_e fmt;
- int nchannels;
+ QEMUTimer *ts;
+ union {
+ int usec;
+ int64_t ticks;
+ } period;
- int shift;
- int align;
+ int plive;
+} AudioState;
+extern AudioState audio_state;
+struct SWVoiceOut {
+ struct audio_pcm_info info;
t_sample *conv;
-
- int left;
- int pos;
- int bytes_per_second;
int64_t ratio;
st_sample_t *buf;
void *rate;
+ int total_hw_samples_mixed;
+ int active;
+ int empty;
+ HWVoiceOut *hw;
+ char *name;
+ volume_t vol;
+ struct audio_callback callback;
+ LIST_ENTRY (SWVoiceOut) entries;
+};
- int wpos;
- int live;
+struct SWVoiceIn {
int active;
- int64_t old_ticks;
- HWVoice *hw;
+ struct audio_pcm_info info;
+ int64_t ratio;
+ void *rate;
+ int total_hw_samples_acquired;
+ st_sample_t *conv_buf;
+ f_sample *clip;
+ HWVoiceIn *hw;
char *name;
+ volume_t vol;
+ struct audio_callback callback;
+ LIST_ENTRY (SWVoiceIn) entries;
};
-struct pcm_ops {
- int (*init) (HWVoice *hw, int freq, int nchannels, audfmt_e fmt);
- void (*fini) (HWVoice *hw);
- void (*run) (HWVoice *hw);
- int (*write) (SWVoice *sw, void *buf, int size);
- int (*ctl) (HWVoice *hw, int cmd, ...);
+struct audio_pcm_ops {
+ int (*init_out)(HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt);
+ void (*fini_out)(HWVoiceOut *hw);
+ int (*run_out) (HWVoiceOut *hw);
+ int (*write) (SWVoiceOut *sw, void *buf, int size);
+ int (*ctl_out) (HWVoiceOut *hw, int cmd, ...);
+
+ int (*init_in) (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt);
+ void (*fini_in) (HWVoiceIn *hw);
+ int (*run_in) (HWVoiceIn *hw);
+ int (*read) (SWVoiceIn *sw, void *buf, int size);
+ int (*ctl_in) (HWVoiceIn *hw, int cmd, ...);
};
-void pcm_sw_free_resources (SWVoice *sw);
-int pcm_sw_alloc_resources (SWVoice *sw);
-void pcm_sw_fini (SWVoice *sw);
-int pcm_sw_init (SWVoice *sw, HWVoice *hw, int freq,
- int nchannels, audfmt_e fmt);
-
-void pcm_hw_clear (HWVoice *hw, void *buf, int len);
-HWVoice * pcm_hw_find_any (HWVoice *hw);
-HWVoice * pcm_hw_find_any_active (HWVoice *hw);
-HWVoice * pcm_hw_find_any_passive (HWVoice *hw);
-HWVoice * pcm_hw_find_specific (HWVoice *hw, int freq,
- int nchannels, audfmt_e fmt);
-HWVoice * pcm_hw_add (int freq, int nchannels, audfmt_e fmt);
-int pcm_hw_add_sw (HWVoice *hw, SWVoice *sw);
-int pcm_hw_del_sw (HWVoice *hw, SWVoice *sw);
-SWVoice * pcm_create_voice_pair (int freq, int nchannels, audfmt_e fmt);
-
-void pcm_hw_free_resources (HWVoice *hw);
-int pcm_hw_alloc_resources (HWVoice *hw);
-void pcm_hw_fini (HWVoice *hw);
-void pcm_hw_gc (HWVoice *hw);
-int pcm_hw_get_live (HWVoice *hw);
-int pcm_hw_get_live2 (HWVoice *hw, int *nb_active);
-void pcm_hw_dec_live (HWVoice *hw, int decr);
-int pcm_hw_write (SWVoice *sw, void *buf, int len);
-
-int audio_get_conf_int (const char *key, int defval);
-const char *audio_get_conf_str (const char *key, const char *defval);
-
-struct audio_output_driver;
+void audio_pcm_init_info (struct audio_pcm_info *info, int freq,
+ int nchannels, audfmt_e fmt, int swap_endian);
+void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len);
+
+int audio_pcm_sw_write (SWVoiceOut *sw, void *buf, int len);
+int audio_pcm_hw_get_live_in (HWVoiceIn *hw);
+
+int audio_pcm_sw_read (SWVoiceIn *sw, void *buf, int len);
+int audio_pcm_hw_get_live_out (HWVoiceOut *hw);
+int audio_pcm_hw_get_live_out2 (HWVoiceOut *hw, int *nb_live);
#define VOICE_ENABLE 1
#define VOICE_DISABLE 2
+static inline int audio_ring_dist (int dst, int src, int len)
+{
+ return (dst >= src) ? (dst - src) : (len - src + dst);
+}
+
+static inline int audio_need_to_swap_endian (int endianness)
+{
+#ifdef WORDS_BIGENDIAN
+ return endianness != 1;
+#else
+ return endianness != 0;
+#endif
+}
+
+#if defined __GNUC__
+#define GCC_ATTR __attribute__ ((__unused__, __format__ (__printf__, 1, 2)))
+#define INIT_FIELD(f) . f
+#define GCC_FMT_ATTR(n, m) __attribute__ ((__format__ (printf, n, m)))
+#else
+#define GCC_ATTR /**/
+#define INIT_FIELD(f) /**/
+#define GCC_FMT_ATTR(n, m)
+#endif
+
+static void GCC_ATTR dolog (const char *fmt, ...)
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+}
+
+#ifdef DEBUG
+static void GCC_ATTR ldebug (const char *fmt, ...)
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+}
+#else
+#if defined NDEBUG && defined __GNUC__
+#define ldebug(...)
+#elif defined NDEBUG && defined _MSC_VER
+#define ldebug __noop
+#else
+static void GCC_ATTR ldebug (const char *fmt, ...)
+{
+ (void) fmt;
+}
+#endif
+#endif
+
+#undef GCC_ATTR
+
+#define AUDIO_STRINGIFY_(n) #n
+#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n)
+
+#if defined _MSC_VER || defined __GNUC__
+#define AUDIO_FUNC __FUNCTION__
+#else
+#define AUDIO_FUNC __FILE__ ":" AUDIO_STRINGIFY (__LINE__)
+#endif
+
#endif /* audio_int.h */
diff --git a/audio/audio_template.h b/audio/audio_template.h
new file mode 100644
index 0000000000..25ea72fd41
--- /dev/null
+++ b/audio/audio_template.h
@@ -0,0 +1,401 @@
+/*
+ * QEMU Audio subsystem header
+ *
+ * Copyright (c) 2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#ifdef DAC
+#define TYPE out
+#define HW glue (HWVoice, Out)
+#define SW glue (SWVoice, Out)
+#else
+#define TYPE in
+#define HW glue (HWVoice, In)
+#define SW glue (SWVoice, In)
+#endif
+
+static void glue (audio_pcm_sw_fini_, TYPE) (SW *sw)
+{
+ glue (audio_pcm_sw_free_resources_, TYPE) (sw);
+ if (sw->name) {
+ qemu_free (sw->name);
+ sw->name = NULL;
+ }
+}
+
+static void glue (audio_pcm_hw_add_sw_, TYPE) (HW *hw, SW *sw)
+{
+ LIST_INSERT_HEAD (&hw->sw_head, sw, entries);
+}
+
+static void glue (audio_pcm_hw_del_sw_, TYPE) (SW *sw)
+{
+ LIST_REMOVE (sw, entries);
+}
+
+static void glue (audio_pcm_hw_fini_, TYPE) (HW *hw)
+{
+ if (hw->active) {
+ glue (audio_pcm_hw_free_resources_ ,TYPE) (hw);
+ glue (hw->pcm_ops->fini_, TYPE) (hw);
+ memset (hw, 0, glue (audio_state.drv->voice_size_, TYPE));
+ }
+}
+
+static void glue (audio_pcm_hw_gc_, TYPE) (HW *hw)
+{
+ if (!hw->sw_head.lh_first) {
+ glue (audio_pcm_hw_fini_, TYPE) (hw);
+ }
+}
+
+static HW *glue (audio_pcm_hw_find_any_, TYPE) (HW *hw)
+{
+ return hw ? hw->entries.le_next : glue (hw_head_, TYPE).lh_first;
+}
+
+static HW *glue (audio_pcm_hw_find_any_active_, TYPE) (HW *hw)
+{
+ while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+ if (hw->active) {
+ return hw;
+ }
+ }
+ return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_any_active_enabled_, TYPE) (HW *hw)
+{
+ while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+ if (hw->active && hw->enabled) {
+ return hw;
+ }
+ }
+ return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_any_passive_, TYPE) (HW *hw)
+{
+ while ((hw = glue (audio_pcm_hw_find_any_, TYPE) (hw))) {
+ if (!hw->active) {
+ return hw;
+ }
+ }
+ return NULL;
+}
+
+static HW *glue (audio_pcm_hw_find_specific_, TYPE) (
+ HW *hw,
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ )
+{
+ while ((hw = glue (audio_pcm_hw_find_any_active_, TYPE) (hw))) {
+ if (audio_pcm_info_eq (&hw->info, freq, nchannels, fmt)) {
+ return hw;
+ }
+ }
+ return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_new_, TYPE) (
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ )
+{
+ HW *hw;
+
+ hw = glue (audio_pcm_hw_find_any_passive_, TYPE) (NULL);
+ if (hw) {
+ hw->pcm_ops = audio_state.drv->pcm_ops;
+ if (!hw->pcm_ops) {
+ return NULL;
+ }
+
+ if (glue (audio_pcm_hw_init_, TYPE) (hw, freq, nchannels, fmt)) {
+ glue (audio_pcm_hw_gc_, TYPE) (hw);
+ return NULL;
+ }
+ else {
+ return hw;
+ }
+ }
+
+ return NULL;
+}
+
+static HW *glue (audio_pcm_hw_add_, TYPE) (
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ )
+{
+ HW *hw;
+
+ if (glue (audio_state.greedy_, TYPE)) {
+ hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+ if (hw) {
+ return hw;
+ }
+ }
+
+ hw = glue (audio_pcm_hw_find_specific_, TYPE) (NULL, freq, nchannels, fmt);
+ if (hw) {
+ return hw;
+ }
+
+ hw = glue (audio_pcm_hw_add_new_, TYPE) (freq, nchannels, fmt);
+ if (hw) {
+ return hw;
+ }
+
+ return glue (audio_pcm_hw_find_any_active_, TYPE) (NULL);
+}
+
+static SW *glue (audio_pcm_create_voice_pair_, TYPE) (
+ const char *name,
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ )
+{
+ SW *sw;
+ HW *hw;
+ int hw_freq = freq;
+ int hw_nchannels = nchannels;
+ int hw_fmt = fmt;
+
+ if (glue (audio_state.fixed_settings_, TYPE)) {
+ hw_freq = glue (audio_state.fixed_freq_, TYPE);
+ hw_nchannels = glue (audio_state.fixed_channels_, TYPE);
+ hw_fmt = glue (audio_state.fixed_fmt_, TYPE);
+ }
+
+ sw = qemu_mallocz (sizeof (*sw));
+ if (!sw) {
+ goto err1;
+ }
+
+ hw = glue (audio_pcm_hw_add_, TYPE) (hw_freq, hw_nchannels, hw_fmt);
+ if (!hw) {
+ goto err2;
+ }
+
+ glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw);
+
+ if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, name, freq, nchannels, fmt)) {
+ goto err3;
+ }
+
+ return sw;
+
+err3:
+ glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+ glue (audio_pcm_hw_gc_, TYPE) (hw);
+err2:
+ qemu_free (sw);
+err1:
+ return NULL;
+}
+
+void glue (AUD_close_, TYPE) (SW *sw)
+{
+ if (sw) {
+ glue (audio_pcm_sw_fini_, TYPE) (sw);
+ glue (audio_pcm_hw_del_sw_, TYPE) (sw);
+ glue (audio_pcm_hw_gc_, TYPE) (sw->hw);
+ qemu_free (sw);
+ }
+}
+
+SW *glue (AUD_open_, TYPE) (
+ SW *sw,
+ const char *name,
+ void *callback_opaque ,
+ audio_callback_fn_t callback_fn,
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ )
+{
+#ifdef DAC
+ int live = 0;
+ SW *old_sw = NULL;
+#endif
+
+ if (!callback_fn) {
+ dolog ("No callback specifed for voice `%s'\n", name);
+ goto fail;
+ }
+
+ if (nchannels != 1 && nchannels != 2) {
+ dolog ("Bogus channel count %d for voice `%s'\n", nchannels, name);
+ goto fail;
+ }
+
+ if (!audio_state.drv) {
+ dolog ("No audio driver defined\n");
+ goto fail;
+ }
+
+ if (sw && audio_pcm_info_eq (&sw->info, freq, nchannels, fmt)) {
+ return sw;
+ }
+
+#ifdef DAC
+ if (audio_state.plive && sw && (!sw->active && !sw->empty)) {
+ live = sw->total_hw_samples_mixed;
+
+#ifdef DEBUG_PLIVE
+ dolog ("Replacing voice %s with %d live samples\n", sw->name, live);
+ dolog ("Old %s freq %d, bits %d, channels %d\n",
+ sw->name, sw->info.freq, sw->info.bits, sw->info.nchannels);
+ dolog ("New %s freq %d, bits %d, channels %d\n",
+ name, freq, (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) ? 16 : 8,
+ nchannels);
+#endif
+
+ if (live) {
+ old_sw = sw;
+ old_sw->callback.fn = NULL;
+ sw = NULL;
+ }
+ }
+#endif
+
+ if (!glue (audio_state.fixed_settings_, TYPE) && sw) {
+ glue (AUD_close_, TYPE) (sw);
+ sw = NULL;
+ }
+
+ if (sw) {
+ HW *hw = sw->hw;
+
+ if (!hw) {
+ dolog ("Internal logic error voice %s has no hardware store\n",
+ name);
+ goto fail;
+ }
+
+ if (glue (audio_pcm_sw_init_, TYPE) (
+ sw,
+ hw,
+ name,
+ freq,
+ nchannels,
+ fmt
+ )) {
+ goto fail;
+ }
+ }
+ else {
+ sw = glue (audio_pcm_create_voice_pair_, TYPE) (
+ name,
+ freq,
+ nchannels,
+ fmt);
+ if (!sw) {
+ dolog ("Failed to create voice %s\n", name);
+ goto fail;
+ }
+ }
+
+ if (sw) {
+ sw->vol = nominal_volume;
+ sw->callback.fn = callback_fn;
+ sw->callback.opaque = callback_opaque;
+
+#ifdef DAC
+ if (live) {
+ int mixed =
+ (live << old_sw->info.shift)
+ * old_sw->info.bytes_per_second
+ / sw->info.bytes_per_second;
+
+#ifdef DEBUG_PLIVE
+ dolog ("Silence will be mixed %d\n", mixed);
+#endif
+ sw->total_hw_samples_mixed += mixed;
+ }
+#endif
+
+#ifdef DEBUG_AUDIO
+ dolog ("%s\n", name);
+ audio_pcm_print_info ("hw", &sw->hw->info);
+ audio_pcm_print_info ("sw", &sw->info);
+#endif
+ }
+
+ return sw;
+
+ fail:
+ glue (AUD_close_, TYPE) (sw);
+ return NULL;
+}
+
+int glue (AUD_is_active_, TYPE) (SW *sw)
+{
+ return sw ? sw->active : 0;
+}
+
+void glue (AUD_init_time_stamp_, TYPE) (SW *sw, QEMUAudioTimeStamp *ts)
+{
+ if (!sw) {
+ return;
+ }
+
+ ts->old_ts = sw->hw->ts_helper;
+}
+
+uint64_t glue (AUD_time_stamp_get_elapsed_usec_, TYPE) (
+ SW *sw,
+ QEMUAudioTimeStamp *ts
+ )
+{
+ uint64_t delta, cur_ts, old_ts;
+
+ if (!sw) {
+ return 0;
+ }
+
+ cur_ts = sw->hw->ts_helper;
+ old_ts = ts->old_ts;
+ /* dolog ("cur %lld old %lld\n", cur_ts, old_ts); */
+
+ if (cur_ts >= old_ts) {
+ delta = cur_ts - old_ts;
+ }
+ else {
+ delta = UINT64_MAX - old_ts + cur_ts;
+ }
+
+ if (!delta) {
+ return 0;
+ }
+
+ return (delta * sw->hw->info.freq) / 1000000;
+}
+
+#undef TYPE
+#undef HW
+#undef SW
diff --git a/audio/coreaudio.c b/audio/coreaudio.c
new file mode 100644
index 0000000000..eee12386ce
--- /dev/null
+++ b/audio/coreaudio.c
@@ -0,0 +1,513 @@
+/*
+ * QEMU OS X CoreAudio audio driver
+ *
+ * Copyright (c) 2005 Mike Kronenberg
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <CoreAudio/CoreAudio.h>
+#include <string.h> /* strerror */
+#include <pthread.h> /* pthread_X */
+
+#include "vl.h"
+
+#define AUDIO_CAP "coreaudio"
+#include "audio_int.h"
+
+#define DEVICE_BUFFER_FRAMES (512)
+
+struct {
+ int buffer_frames;
+} conf = {
+ .buffer_frames = 512
+};
+
+typedef struct coreaudioVoiceOut {
+ HWVoiceOut hw;
+ pthread_mutex_t mutex;
+ AudioDeviceID outputDeviceID;
+ UInt32 audioDevicePropertyBufferSize;
+ AudioStreamBasicDescription outputStreamBasicDescription;
+ int isPlaying;
+ int live;
+ int decr;
+ int rpos;
+} coreaudioVoiceOut;
+
+static void coreaudio_logstatus (OSStatus status)
+{
+ char *str = "BUG";
+
+ switch(status) {
+ case kAudioHardwareNoError:
+ str = "kAudioHardwareNoError";
+ break;
+
+ case kAudioHardwareNotRunningError:
+ str = "kAudioHardwareNotRunningError";
+ break;
+
+ case kAudioHardwareUnspecifiedError:
+ str = "kAudioHardwareUnspecifiedError";
+ break;
+
+ case kAudioHardwareUnknownPropertyError:
+ str = "kAudioHardwareUnknownPropertyError";
+ break;
+
+ case kAudioHardwareBadPropertySizeError:
+ str = "kAudioHardwareBadPropertySizeError";
+ break;
+
+ case kAudioHardwareIllegalOperationError:
+ str = "kAudioHardwareIllegalOperationError";
+ break;
+
+ case kAudioHardwareBadDeviceError:
+ str = "kAudioHardwareBadDeviceError";
+ break;
+
+ case kAudioHardwareBadStreamError:
+ str = "kAudioHardwareBadStreamError";
+ break;
+
+ case kAudioHardwareUnsupportedOperationError:
+ str = "kAudioHardwareUnsupportedOperationError";
+ break;
+
+ case kAudioDeviceUnsupportedFormatError:
+ str = "kAudioDeviceUnsupportedFormatError";
+ break;
+
+ case kAudioDevicePermissionsError:
+ str = "kAudioDevicePermissionsError";
+ break;
+
+ default:
+ AUD_log (AUDIO_CAP, "Reason: status code %ld\n", status);
+ return;
+ }
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", str);
+}
+
+static void GCC_FMT_ATTR (2, 3) coreaudio_logerr (
+ OSStatus status,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_log (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ coreaudio_logstatus (status);
+}
+
+static void GCC_FMT_ATTR (3, 4) coreaudio_logerr2 (
+ OSStatus status,
+ const char *typ,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ coreaudio_logstatus (status);
+}
+
+static int coreaudio_lock (coreaudioVoiceOut *core, const char *fn_name)
+{
+ int err;
+
+ err = pthread_mutex_lock (&core->mutex);
+ if (err) {
+ dolog ("Can not lock voice for %s\nReason: %s\n",
+ fn_name, strerror (err));
+ return -1;
+ }
+ return 0;
+}
+
+static int coreaudio_unlock (coreaudioVoiceOut *core, const char *fn_name)
+{
+ int err;
+
+ err = pthread_mutex_unlock (&core->mutex);
+ if (err) {
+ dolog ("Can not unlock voice for %s\nReason: %s\n",
+ fn_name, strerror (err));
+ return -1;
+ }
+ return 0;
+}
+
+static int coreaudio_run_out (HWVoiceOut *hw)
+{
+ int live, decr;
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+ if (coreaudio_lock (core, "coreaudio_run_out")) {
+ return 0;
+ }
+
+ live = audio_pcm_hw_get_live_out (hw);
+
+ if (core->decr > live) {
+ ldebug ("core->decr %d live %d core->live %d\n",
+ core->decr,
+ live,
+ core->live);
+ }
+
+ decr = audio_MIN (core->decr, live);
+ core->decr -= decr;
+
+ core->live = live - decr;
+ hw->rpos = core->rpos;
+
+ coreaudio_unlock (core, "coreaudio_run_out");
+ return decr;
+}
+
+/* callback to feed audiooutput buffer */
+static OSStatus audioDeviceIOProc(
+ AudioDeviceID inDevice,
+ const AudioTimeStamp* inNow,
+ const AudioBufferList* inInputData,
+ const AudioTimeStamp* inInputTime,
+ AudioBufferList* outOutputData,
+ const AudioTimeStamp* inOutputTime,
+ void* hwptr)
+{
+ unsigned int frame, frameCount;
+ float *out = outOutputData->mBuffers[0].mData;
+ HWVoiceOut *hw = hwptr;
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hwptr;
+ int rpos, live;
+ st_sample_t *src;
+#ifndef FLOAT_MIXENG
+#ifdef RECIPROCAL
+ const float scale = 1.f / UINT_MAX;
+#else
+ const float scale = UINT_MAX;
+#endif
+#endif
+
+ if (coreaudio_lock (core, "audioDeviceIOProc")) {
+ inInputTime = 0;
+ return 0;
+ }
+
+ frameCount = conf.buffer_frames;
+ live = core->live;
+
+ /* if there are not enough samples, set signal and return */
+ if (live < frameCount) {
+ inInputTime = 0;
+ coreaudio_unlock (core, "audioDeviceIOProc(empty)");
+ return 0;
+ }
+
+ rpos = core->rpos;
+ src = hw->mix_buf + rpos;
+
+ /* fill buffer */
+ for (frame = 0; frame < frameCount; frame++) {
+#ifdef FLOAT_MIXENG
+ *out++ = src[frame].l; /* left channel */
+ *out++ = src[frame].r; /* right channel */
+#else
+#ifdef RECIPROCAL
+ *out++ = src[frame].l * scale; /* left channel */
+ *out++ = src[frame].r * scale; /* right channel */
+#else
+ *out++ = src[frame].l / scale; /* left channel */
+ *out++ = src[frame].r / scale; /* right channel */
+#endif
+#endif
+ }
+
+ /* cleanup */
+ mixeng_clear (src, frameCount);
+ rpos = (rpos + frameCount) % hw->samples;
+ core->decr = frameCount;
+ core->rpos = rpos;
+
+ coreaudio_unlock (core, "audioDeviceIOProc");
+ return 0;
+}
+
+static int coreaudio_write (SWVoiceOut *sw, void *buf, int len)
+{
+ return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int coreaudio_init_out (HWVoiceOut *hw, int freq,
+ int nchannels, audfmt_e fmt)
+{
+ OSStatus status;
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+ UInt32 propertySize;
+ int err;
+ int bits = 8;
+ int endianess = 0;
+ const char *typ = "DAC";
+
+ /* create mutex */
+ err = pthread_mutex_init(&core->mutex, NULL);
+ if (err) {
+ dolog("Can not create mutex\nReason: %s\n", strerror (err));
+ return -1;
+ }
+
+ if (fmt == AUD_FMT_S16 || fmt == AUD_FMT_U16) {
+ bits = 16;
+ endianess = 1;
+ }
+
+ audio_pcm_init_info (
+ &hw->info,
+ freq,
+ nchannels,
+ fmt,
+ /* Following is irrelevant actually since we do not use
+ mixengs clipping routines */
+ audio_need_to_swap_endian (endianess)
+ );
+ hw->bufsize = 4 * conf.buffer_frames * nchannels * bits;
+
+ /* open default output device */
+ propertySize = sizeof(core->outputDeviceID);
+ status = AudioHardwareGetProperty(
+ kAudioHardwarePropertyDefaultOutputDevice,
+ &propertySize,
+ &core->outputDeviceID);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr2 (status, typ,
+ "Can not get default output Device\n");
+ return -1;
+ }
+ if (core->outputDeviceID == kAudioDeviceUnknown) {
+ dolog ("Can not initialize %s - Unknown Audiodevice\n", typ);
+ return -1;
+ }
+
+ /* set Buffersize to conf.buffer_frames frames */
+ propertySize = sizeof(core->audioDevicePropertyBufferSize);
+ core->audioDevicePropertyBufferSize =
+ conf.buffer_frames * sizeof(float) * 2;
+ status = AudioDeviceSetProperty(
+ core->outputDeviceID,
+ NULL,
+ 0,
+ false,
+ kAudioDevicePropertyBufferSize,
+ propertySize,
+ &core->audioDevicePropertyBufferSize);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr2 (status, typ,
+ "Can not set device buffer size %d\n",
+ kAudioDevicePropertyBufferSize);
+ return -1;
+ }
+
+ /* get Buffersize */
+ propertySize = sizeof(core->audioDevicePropertyBufferSize);
+ status = AudioDeviceGetProperty(
+ core->outputDeviceID,
+ 0,
+ false,
+ kAudioDevicePropertyBufferSize,
+ &propertySize,
+ &core->audioDevicePropertyBufferSize);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr2 (status, typ, "Can not get device buffer size\n");
+ return -1;
+ }
+
+ /* get StreamFormat */
+ propertySize = sizeof(core->outputStreamBasicDescription);
+ status = AudioDeviceGetProperty(
+ core->outputDeviceID,
+ 0,
+ false,
+ kAudioDevicePropertyStreamFormat,
+ &propertySize,
+ &core->outputStreamBasicDescription);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr2 (status, typ,
+ "Can not get Device Stream properties\n");
+ core->outputDeviceID = kAudioDeviceUnknown;
+ return -1;
+ }
+
+ /* set Samplerate */
+ core->outputStreamBasicDescription.mSampleRate = (Float64)freq;
+ propertySize = sizeof(core->outputStreamBasicDescription);
+ status = AudioDeviceSetProperty(
+ core->outputDeviceID,
+ 0,
+ 0,
+ 0,
+ kAudioDevicePropertyStreamFormat,
+ propertySize,
+ &core->outputStreamBasicDescription);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr2 (status, typ, "Can not set samplerate %d\n", freq);
+ core->outputDeviceID = kAudioDeviceUnknown;
+ return -1;
+ }
+
+ /* set Callback */
+ status = AudioDeviceAddIOProc(core->outputDeviceID, audioDeviceIOProc, hw);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr2 (status, typ, "Can not set IOProc\n");
+ core->outputDeviceID = kAudioDeviceUnknown;
+ return -1;
+ }
+
+ /* start Playback */
+ if (!core->isPlaying) {
+ status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr2 (status, typ, "Can not start playback\n");
+ AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
+ core->outputDeviceID = kAudioDeviceUnknown;
+ return -1;
+ }
+ core->isPlaying = 1;
+ }
+
+ return 0;
+}
+
+static void coreaudio_fini_out (HWVoiceOut *hw)
+{
+ OSStatus status;
+ int err;
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+ /* stop playback */
+ if (core->isPlaying) {
+ status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr (status, "Can not stop playback\n");
+ }
+ core->isPlaying = 0;
+ }
+
+ /* remove callback */
+ status = AudioDeviceRemoveIOProc(core->outputDeviceID, audioDeviceIOProc);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr (status, "Can not remove IOProc\n");
+ }
+ core->outputDeviceID = kAudioDeviceUnknown;
+
+ /* destroy mutex */
+ err = pthread_mutex_destroy(&core->mutex);
+ if (err) {
+ dolog("Can not destroy mutex\nReason: %s\n", strerror (err));
+ }
+}
+
+static int coreaudio_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+ OSStatus status;
+ coreaudioVoiceOut *core = (coreaudioVoiceOut *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ /* start playback */
+ if (!core->isPlaying) {
+ status = AudioDeviceStart(core->outputDeviceID, audioDeviceIOProc);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr (status, "Can not unpause playback\n");
+ }
+ core->isPlaying = 1;
+ }
+ break;
+
+ case VOICE_DISABLE:
+ /* stop playback */
+ if (core->isPlaying) {
+ status = AudioDeviceStop(core->outputDeviceID, audioDeviceIOProc);
+ if (status != kAudioHardwareNoError) {
+ coreaudio_logerr (status, "Can not pause playback\n");
+ }
+ core->isPlaying = 0;
+ }
+ break;
+ }
+ return 0;
+}
+
+static void *coreaudio_audio_init (void)
+{
+ return &coreaudio_audio_init;
+}
+
+static void coreaudio_audio_fini (void *opaque)
+{
+ (void) opaque;
+}
+
+static struct audio_option coreaudio_options[] = {
+ {"BUFFER_SIZE", AUD_OPT_INT, &conf.buffer_frames,
+ "Size of the buffer in frames", NULL, 0},
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops coreaudio_pcm_ops = {
+ coreaudio_init_out,
+ coreaudio_fini_out,
+ coreaudio_run_out,
+ coreaudio_write,
+ coreaudio_ctl_out,
+
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ NULL
+};
+
+struct audio_driver coreaudio_audio_driver = {
+ INIT_FIELD (name = ) "coreaudio",
+ INIT_FIELD (descr = )
+ "CoreAudio http://developer.apple.com/audio/coreaudio.html",
+ INIT_FIELD (options = ) coreaudio_options,
+ INIT_FIELD (init = ) coreaudio_audio_init,
+ INIT_FIELD (fini = ) coreaudio_audio_fini,
+ INIT_FIELD (pcm_ops = ) &coreaudio_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) 1,
+ INIT_FIELD (max_voices_in = ) 0,
+ INIT_FIELD (voice_size_out = ) sizeof (coreaudioVoiceOut),
+ INIT_FIELD (voice_size_in = ) 0
+};
diff --git a/audio/dsound_template.h b/audio/dsound_template.h
new file mode 100644
index 0000000000..a04806eae0
--- /dev/null
+++ b/audio/dsound_template.h
@@ -0,0 +1,298 @@
+/*
+ * QEMU DirectSound audio driver header
+ *
+ * Copyright (c) 2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+#ifdef DSBTYPE_IN
+#define NAME "capture buffer"
+#define TYPE in
+#define IFACE IDirectSoundCaptureBuffer
+#define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER
+#define FIELD dsound_capture_buffer
+#else
+#define NAME "playback buffer"
+#define TYPE out
+#define IFACE IDirectSoundBuffer
+#define BUFPTR LPDIRECTSOUNDBUFFER
+#define FIELD dsound_buffer
+#endif
+
+static int glue (dsound_unlock_, TYPE) (
+ BUFPTR buf,
+ LPVOID p1,
+ LPVOID p2,
+ DWORD blen1,
+ DWORD blen2
+ )
+{
+ HRESULT hr;
+
+ hr = glue (IFACE, _Unlock) (buf, p1, blen1, p2, blen2);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not unlock " NAME "\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+static int glue (dsound_lock_, TYPE) (
+ BUFPTR buf,
+ struct audio_pcm_info *info,
+ DWORD pos,
+ DWORD len,
+ LPVOID *p1p,
+ LPVOID *p2p,
+ DWORD *blen1p,
+ DWORD *blen2p,
+ int entire
+ )
+{
+ HRESULT hr;
+ int i;
+ LPVOID p1 = NULL, p2 = NULL;
+ DWORD blen1 = 0, blen2 = 0;
+
+ for (i = 0; i < conf.lock_retries; ++i) {
+ hr = glue (IFACE, _Lock) (
+ buf,
+ pos,
+ len,
+ &p1,
+ &blen1,
+ &p2,
+ &blen2,
+ (entire
+#ifdef DSBTYPE_IN
+ ? DSCBLOCK_ENTIREBUFFER
+#else
+ ? DSBLOCK_ENTIREBUFFER
+#endif
+ : 0)
+ );
+
+ if (FAILED (hr)) {
+#ifndef DSBTYPE_IN
+ if (hr == DSERR_BUFFERLOST) {
+ if (glue (dsound_restore_, TYPE) (buf)) {
+ dsound_logerr (hr, "Can not lock " NAME "\n");
+ goto fail;
+ }
+ continue;
+ }
+#endif
+ dsound_logerr (hr, "Can not lock " NAME "\n");
+ goto fail;
+ }
+
+ break;
+ }
+
+ if (i == conf.lock_retries) {
+ dolog ("%d attempts to lock " NAME " failed\n", i);
+ goto fail;
+ }
+
+ if ((p1 && (blen1 & info->align)) || (p2 && (blen2 & info->align))) {
+ dolog ("DirectSound returned misaligned buffer %ld %ld\n",
+ blen1, blen2);
+ glue (dsound_unlock_, TYPE) (buf, p1, p2, blen1, blen2);
+ goto fail;
+ }
+
+ if (!p1 && blen1) {
+ dolog ("warning: !p1 && blen1=%ld\n", blen1);
+ blen1 = 0;
+ }
+
+ if (!p2 && blen2) {
+ dolog ("warning: !p2 && blen2=%ld\n", blen2);
+ blen2 = 0;
+ }
+
+ *p1p = p1;
+ *p2p = p2;
+ *blen1p = blen1;
+ *blen2p = blen2;
+ return 0;
+
+ fail:
+ *p1p = NULL - 1;
+ *p2p = NULL - 1;
+ *blen1p = -1;
+ *blen2p = -1;
+ return -1;
+}
+
+#ifdef DSBTYPE_IN
+static void dsound_fini_in (HWVoiceIn *hw)
+#else
+static void dsound_fini_out (HWVoiceOut *hw)
+#endif
+{
+ HRESULT hr;
+#ifdef DSBTYPE_IN
+ DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+#else
+ DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+#endif
+
+ if (ds->FIELD) {
+ hr = glue (IFACE, _Stop) (ds->FIELD);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not stop " NAME "\n");
+ }
+
+ hr = glue (IFACE, _Release) (ds->FIELD);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not release " NAME "\n");
+ }
+ ds->FIELD = NULL;
+ }
+}
+
+#ifdef DSBTYPE_IN
+static int dsound_init_in (
+ HWVoiceIn *hw,
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ )
+#else
+static int dsound_init_out (
+ HWVoiceOut *hw,
+ int freq,
+ int nchannels,
+ audfmt_e fmt
+ )
+#endif
+{
+ int err;
+ HRESULT hr;
+ dsound *s = &glob_dsound;
+ WAVEFORMATEX wfx;
+ struct full_fmt full_fmt;
+#ifdef DSBTYPE_IN
+ const char *typ = "ADC";
+ DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+ DSCBUFFERDESC bd;
+ DSCBCAPS bc;
+#else
+ const char *typ = "DAC";
+ DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+ DSBUFFERDESC bd;
+ DSBCAPS bc;
+#endif
+
+ full_fmt.freq = freq;
+ full_fmt.nchannels = nchannels;
+ full_fmt.fmt = fmt;
+ err = waveformat_from_full_fmt (&wfx, &full_fmt);
+ if (err) {
+ return -1;
+ }
+
+ memset (&bd, 0, sizeof (bd));
+ bd.dwSize = sizeof (bd);
+ bd.lpwfxFormat = &wfx;
+#ifdef DSBTYPE_IN
+ bd.dwBufferBytes = conf.bufsize_in;
+ hr = IDirectSoundCapture_CreateCaptureBuffer (
+ s->dsound_capture,
+ &bd,
+ &ds->dsound_capture_buffer,
+ NULL
+ );
+#else
+ bd.dwFlags = DSBCAPS_STICKYFOCUS | DSBCAPS_GETCURRENTPOSITION2;
+ bd.dwBufferBytes = conf.bufsize_out;
+ hr = IDirectSound_CreateSoundBuffer (
+ s->dsound,
+ &bd,
+ &ds->dsound_buffer,
+ NULL
+ );
+#endif
+
+ if (FAILED (hr)) {
+ dsound_logerr2 (hr, typ, "Can not create " NAME "\n");
+ return -1;
+ }
+
+ hr = glue (IFACE, _GetFormat) (
+ ds->FIELD,
+ &wfx,
+ sizeof (wfx),
+ NULL
+ );
+ if (FAILED (hr)) {
+ dsound_logerr2 (hr, typ, "Can not get " NAME " format\n");
+ goto fail0;
+ }
+
+#ifdef DEBUG_DSOUND
+ dolog (NAME "\n");
+ print_wave_format (&wfx);
+#endif
+
+ memset (&bc, 0, sizeof (bc));
+ bc.dwSize = sizeof (bc);
+
+ hr = glue (IFACE, _GetCaps) (ds->FIELD, &bc);
+ if (FAILED (hr)) {
+ dsound_logerr2 (hr, typ, "Can not get " NAME " format\n");
+ goto fail0;
+ }
+
+ err = waveformat_to_full_fmt (&wfx, &full_fmt);
+ if (err) {
+ goto fail0;
+ }
+
+ ds->first_time = 1;
+ hw->bufsize = bc.dwBufferBytes;
+ audio_pcm_init_info (
+ &hw->info,
+ full_fmt.freq,
+ full_fmt.nchannels,
+ full_fmt.fmt,
+ audio_need_to_swap_endian (0)
+ );
+
+#ifdef DEBUG_DSOUND
+ dolog ("caps %ld, desc %ld\n",
+ bc.dwBufferBytes, bd.dwBufferBytes);
+
+ dolog ("bufsize %d, freq %d, chan %d, fmt %d\n",
+ hw->bufsize, full_fmt.freq, full_fmt.nchannels, full_fmt.fmt);
+#endif
+ return 0;
+
+ fail0:
+ glue (dsound_fini_, TYPE) (hw);
+ return -1;
+}
+
+#undef NAME
+#undef TYPE
+#undef IFACE
+#undef BUFPTR
+#undef FIELD
diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c
new file mode 100644
index 0000000000..64b84174d1
--- /dev/null
+++ b/audio/dsoundaudio.c
@@ -0,0 +1,1082 @@
+/*
+ * QEMU DirectSound audio driver
+ *
+ * Copyright (c) 2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/*
+ * SEAL 1.07 by Carlos 'pel' Hasan was used as documentation
+ */
+
+#include "vl.h"
+
+#define AUDIO_CAP "dsound"
+#include "audio_int.h"
+
+#include <windows.h>
+#include <objbase.h>
+#include <dsound.h>
+
+/* #define DEBUG_DSOUND */
+
+struct full_fmt {
+ int freq;
+ int nchannels;
+ audfmt_e fmt;
+};
+
+static struct {
+ int lock_retries;
+ int restore_retries;
+ int getstatus_retries;
+ int set_primary;
+ int bufsize_in;
+ int bufsize_out;
+ struct full_fmt full_fmt;
+ int latency_millis;
+} conf = {
+ 1,
+ 1,
+ 1,
+ 0,
+ 16384,
+ 16384,
+ {
+ 44100,
+ 2,
+ AUD_FMT_S16
+ },
+ 10
+};
+
+typedef struct {
+ LPDIRECTSOUND dsound;
+ LPDIRECTSOUNDCAPTURE dsound_capture;
+ LPDIRECTSOUNDBUFFER dsound_primary_buffer;
+ struct full_fmt fmt;
+} dsound;
+
+static dsound glob_dsound;
+
+typedef struct {
+ HWVoiceOut hw;
+ LPDIRECTSOUNDBUFFER dsound_buffer;
+ DWORD old_pos;
+ int first_time;
+#ifdef DEBUG_DSOUND
+ DWORD old_ppos;
+ DWORD played;
+ DWORD mixed;
+#endif
+} DSoundVoiceOut;
+
+typedef struct {
+ HWVoiceIn hw;
+ int first_time;
+ LPDIRECTSOUNDCAPTUREBUFFER dsound_capture_buffer;
+} DSoundVoiceIn;
+
+static void dsound_log_hresult (HRESULT hr)
+{
+ const char *str = "BUG";
+
+ switch (hr) {
+ case DS_OK:
+ str = "The method succeeded";
+ break;
+#ifdef DS_NO_VIRTUALIZATION
+ case DS_NO_VIRTUALIZATION:
+ str = "The buffer was created, but another 3D algorithm was substituted";
+ break;
+#endif
+#ifdef DS_INCOMPLETE
+ case DS_INCOMPLETE:
+ str = "The method succeeded, but not all the optional effects were obtained";
+ break;
+#endif
+#ifdef DSERR_ACCESSDENIED
+ case DSERR_ACCESSDENIED:
+ str = "The request failed because access was denied";
+ break;
+#endif
+#ifdef DSERR_ALLOCATED
+ case DSERR_ALLOCATED:
+ str = "The request failed because resources, such as a priority level, were already in use by another caller";
+ break;
+#endif
+#ifdef DSERR_ALREADYINITIALIZED
+ case DSERR_ALREADYINITIALIZED:
+ str = "The object is already initialized";
+ break;
+#endif
+#ifdef DSERR_BADFORMAT
+ case DSERR_BADFORMAT:
+ str = "The specified wave format is not supported";
+ break;
+#endif
+#ifdef DSERR_BADSENDBUFFERGUID
+ case DSERR_BADSENDBUFFERGUID:
+ str = "The GUID specified in an audiopath file does not match a valid mix-in buffer";
+ break;
+#endif
+#ifdef DSERR_BUFFERLOST
+ case DSERR_BUFFERLOST:
+ str = "The buffer memory has been lost and must be restored";
+ break;
+#endif
+#ifdef DSERR_BUFFERTOOSMALL
+ case DSERR_BUFFERTOOSMALL:
+ str = "The buffer size is not great enough to enable effects processing";
+ break;
+#endif
+#ifdef DSERR_CONTROLUNAVAIL
+ case DSERR_CONTROLUNAVAIL:
+ str = "The buffer control (volume, pan, and so on) requested by the caller is not available. Controls must be specified when the buffer is created, using the dwFlags member of DSBUFFERDESC";
+ break;
+#endif
+#ifdef DSERR_DS8_REQUIRED
+ case DSERR_DS8_REQUIRED:
+ str = "A DirectSound object of class CLSID_DirectSound8 or later is required for the requested functionality. For more information, see IDirectSound8 Interface";
+ break;
+#endif
+#ifdef DSERR_FXUNAVAILABLE
+ case DSERR_FXUNAVAILABLE:
+ str = "The effects requested could not be found on the system, or they are in the wrong order or in the wrong location; for example, an effect expected in hardware was found in software";
+ break;
+#endif
+#ifdef DSERR_GENERIC
+ case DSERR_GENERIC :
+ str = "An undetermined error occurred inside the DirectSound subsystem";
+ break;
+#endif
+#ifdef DSERR_INVALIDCALL
+ case DSERR_INVALIDCALL:
+ str = "This function is not valid for the current state of this object";
+ break;
+#endif
+#ifdef DSERR_INVALIDPARAM
+ case DSERR_INVALIDPARAM:
+ str = "An invalid parameter was passed to the returning function";
+ break;
+#endif
+#ifdef DSERR_NOAGGREGATION
+ case DSERR_NOAGGREGATION:
+ str = "The object does not support aggregation";
+ break;
+#endif
+#ifdef DSERR_NODRIVER
+ case DSERR_NODRIVER:
+ str = "No sound driver is available for use, or the given GUID is not a valid DirectSound device ID";
+ break;
+#endif
+#ifdef DSERR_NOINTERFACE
+ case DSERR_NOINTERFACE:
+ str = "The requested COM interface is not available";
+ break;
+#endif
+#ifdef DSERR_OBJECTNOTFOUND
+ case DSERR_OBJECTNOTFOUND:
+ str = "The requested object was not found";
+ break;
+#endif
+#ifdef DSERR_OTHERAPPHASPRIO
+ case DSERR_OTHERAPPHASPRIO:
+ str = "Another application has a higher priority level, preventing this call from succeeding";
+ break;
+#endif
+#ifdef DSERR_OUTOFMEMORY
+ case DSERR_OUTOFMEMORY:
+ str = "The DirectSound subsystem could not allocate sufficient memory to complete the caller's request";
+ break;
+#endif
+#ifdef DSERR_PRIOLEVELNEEDED
+ case DSERR_PRIOLEVELNEEDED:
+ str = "A cooperative level of DSSCL_PRIORITY or higher is required";
+ break;
+#endif
+#ifdef DSERR_SENDLOOP
+ case DSERR_SENDLOOP:
+ str = "A circular loop of send effects was detected";
+ break;
+#endif
+#ifdef DSERR_UNINITIALIZED
+ case DSERR_UNINITIALIZED:
+ str = "The Initialize method has not been called or has not been called successfully before other methods were called";
+ break;
+#endif
+#ifdef DSERR_UNSUPPORTED
+ case DSERR_UNSUPPORTED:
+ str = "The function called is not supported at this time";
+ break;
+#endif
+ default:
+ AUD_log (AUDIO_CAP, "Reason: Unknown (HRESULT %#lx)\n", hr);
+ return;
+ }
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", str);
+}
+
+static void GCC_FMT_ATTR (2, 3) dsound_logerr (
+ HRESULT hr,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ dsound_log_hresult (hr);
+}
+
+static void GCC_FMT_ATTR (3, 4) dsound_logerr2 (
+ HRESULT hr,
+ const char *typ,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ dsound_log_hresult (hr);
+}
+
+static DWORD millis_to_bytes (struct audio_pcm_info *info, DWORD millis)
+{
+ return (millis * info->bytes_per_second) / 1000;
+}
+
+#ifdef DEBUG_DSOUND
+static void print_wave_format (WAVEFORMATEX *wfx)
+{
+ dolog ("tag = %d\n", wfx->wFormatTag);
+ dolog ("nChannels = %d\n", wfx->nChannels);
+ dolog ("nSamplesPerSec = %ld\n", wfx->nSamplesPerSec);
+ dolog ("nAvgBytesPerSec = %ld\n", wfx->nAvgBytesPerSec);
+ dolog ("nBlockAlign = %d\n", wfx->nBlockAlign);
+ dolog ("wBitsPerSample = %d\n", wfx->wBitsPerSample);
+ dolog ("cbSize = %d\n", wfx->cbSize);
+}
+#endif
+
+static int dsound_restore_out (LPDIRECTSOUNDBUFFER dsb)
+{
+ HRESULT hr;
+ int i;
+
+ for (i = 0; i < conf.restore_retries; ++i) {
+ hr = IDirectSoundBuffer_Restore (dsb);
+
+ switch (hr) {
+ case DS_OK:
+ return 0;
+
+ case DSERR_BUFFERLOST:
+ continue;
+
+ default:
+ dsound_logerr (hr, "Can not restore playback buffer\n");
+ return -1;
+ }
+ }
+
+ dolog ("%d attempts to restore playback buffer failed\n", i);
+ return -1;
+}
+
+static int waveformat_from_full_fmt (WAVEFORMATEX *wfx,
+ struct full_fmt *full_fmt)
+{
+ memset (wfx, 0, sizeof (*wfx));
+
+ wfx->wFormatTag = WAVE_FORMAT_PCM;
+ wfx->nChannels = full_fmt->nchannels;
+ wfx->nSamplesPerSec = full_fmt->freq;
+ wfx->nAvgBytesPerSec = full_fmt->freq << (full_fmt->nchannels == 2);
+ wfx->nBlockAlign = 1 << (full_fmt->nchannels == 2);
+ wfx->cbSize = 0;
+
+ switch (full_fmt->fmt) {
+ case AUD_FMT_S8:
+ wfx->wBitsPerSample = 8;
+ break;
+
+ case AUD_FMT_U8:
+ wfx->wBitsPerSample = 8;
+ break;
+
+ case AUD_FMT_S16:
+ wfx->wBitsPerSample = 16;
+ wfx->nAvgBytesPerSec <<= 1;
+ wfx->nBlockAlign <<= 1;
+ break;
+
+ case AUD_FMT_U16:
+ wfx->wBitsPerSample = 16;
+ wfx->nAvgBytesPerSec <<= 1;
+ wfx->nBlockAlign <<= 1;
+ break;
+
+ default:
+ dolog ("Internal logic error: Bad audio format %d\n",
+ full_fmt->freq);
+ return -1;
+ }
+
+ return 0;
+}
+
+static int waveformat_to_full_fmt (WAVEFORMATEX *wfx,
+ struct full_fmt *full_fmt)
+{
+ if (wfx->wFormatTag != WAVE_FORMAT_PCM) {
+ dolog ("Invalid wave format, tag is not PCM, but %d\n",
+ wfx->wFormatTag);
+ return -1;
+ }
+
+ if (!wfx->nSamplesPerSec) {
+ dolog ("Invalid wave format, frequency is zero\n");
+ return -1;
+ }
+ full_fmt->freq = wfx->nSamplesPerSec;
+
+ switch (wfx->nChannels) {
+ case 1:
+ full_fmt->nchannels = 1;
+ break;
+
+ case 2:
+ full_fmt->nchannels = 2;
+ break;
+
+ default:
+ dolog (
+ "Invalid wave format, number of channels is not 1 or 2, but %d\n",
+ wfx->nChannels
+ );
+ return -1;
+ }
+
+ switch (wfx->wBitsPerSample) {
+ case 8:
+ full_fmt->fmt = AUD_FMT_U8;
+ break;
+
+ case 16:
+ full_fmt->fmt = AUD_FMT_S16;
+ break;
+
+ default:
+ dolog ("Invalid wave format, bits per sample is not 8 or 16, but %d\n",
+ wfx->wBitsPerSample);
+ return -1;
+ }
+
+ return 0;
+}
+
+#include "dsound_template.h"
+#define DSBTYPE_IN
+#include "dsound_template.h"
+#undef DSBTYPE_IN
+
+static int dsound_get_status_out (LPDIRECTSOUNDBUFFER dsb, DWORD *statusp)
+{
+ HRESULT hr;
+ int i;
+
+ for (i = 0; i < conf.getstatus_retries; ++i) {
+ hr = IDirectSoundBuffer_GetStatus (dsb, statusp);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not get playback buffer status\n");
+ return -1;
+ }
+
+ if (*statusp & DSERR_BUFFERLOST) {
+ if (dsound_restore_out (dsb)) {
+ return -1;
+ }
+ continue;
+ }
+ break;
+ }
+
+ return 0;
+}
+
+static int dsound_get_status_in (LPDIRECTSOUNDCAPTUREBUFFER dscb,
+ DWORD *statusp)
+{
+ HRESULT hr;
+
+ hr = IDirectSoundCaptureBuffer_GetStatus (dscb, statusp);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not get capture buffer status\n");
+ return -1;
+ }
+
+ return 0;
+}
+
+static void dsound_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
+{
+ int src_len1 = dst_len;
+ int src_len2 = 0;
+ int pos = hw->rpos + dst_len;
+ st_sample_t *src1 = hw->mix_buf + hw->rpos;
+ st_sample_t *src2 = NULL;
+
+ if (pos > hw->samples) {
+ src_len1 = hw->samples - hw->rpos;
+ src2 = hw->mix_buf;
+ src_len2 = dst_len - src_len1;
+ pos = src_len2;
+ }
+
+ if (src_len1) {
+ hw->clip (dst, src1, src_len1);
+ mixeng_clear (src1, src_len1);
+ }
+
+ if (src_len2) {
+ dst = advance (dst, src_len1 << hw->info.shift);
+ hw->clip (dst, src2, src_len2);
+ mixeng_clear (src2, src_len2);
+ }
+
+ hw->rpos = pos % hw->samples;
+}
+
+static void dsound_clear_sample (HWVoiceOut *hw, LPDIRECTSOUNDBUFFER dsb)
+{
+ int err;
+ LPVOID p1, p2;
+ DWORD blen1, blen2, len1, len2;
+
+ err = dsound_lock_out (
+ dsb,
+ &hw->info,
+ 0,
+ hw->samples << hw->info.shift,
+ &p1, &p2,
+ &blen1, &blen2,
+ 1
+ );
+ if (err) {
+ return;
+ }
+
+ len1 = blen1 >> hw->info.shift;
+ len2 = blen2 >> hw->info.shift;
+
+#ifdef DEBUG_DSOUND
+ dolog ("clear %p,%ld,%ld %p,%ld,%ld\n",
+ p1, blen1, len1,
+ p2, blen2, len2);
+#endif
+
+ if (p1 && len1) {
+ audio_pcm_info_clear_buf (&hw->info, p1, len1);
+ }
+
+ if (p2 && len2) {
+ audio_pcm_info_clear_buf (&hw->info, p2, len2);
+ }
+
+ dsound_unlock_out (dsb, p1, p2, blen1, blen2);
+}
+
+static void dsound_close (dsound *s)
+{
+ HRESULT hr;
+
+ if (s->dsound_primary_buffer) {
+ hr = IDirectSoundBuffer_Release (s->dsound_primary_buffer);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not release primary buffer\n");
+ }
+ s->dsound_primary_buffer = NULL;
+ }
+}
+
+static int dsound_open (dsound *s)
+{
+ int err;
+ HRESULT hr;
+ WAVEFORMATEX wfx;
+ DSBUFFERDESC dsbd;
+ HWND hwnd;
+
+ hwnd = GetForegroundWindow ();
+ hr = IDirectSound_SetCooperativeLevel (
+ s->dsound,
+ hwnd,
+ DSSCL_PRIORITY
+ );
+
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not set cooperative level for window %p\n",
+ hwnd);
+ return -1;
+ }
+
+ if (!conf.set_primary) {
+ return 0;
+ }
+
+ err = waveformat_from_full_fmt (&wfx, &conf.full_fmt);
+ if (err) {
+ return -1;
+ }
+
+ memset (&dsbd, 0, sizeof (dsbd));
+ dsbd.dwSize = sizeof (dsbd);
+ dsbd.dwFlags = DSBCAPS_PRIMARYBUFFER;
+ dsbd.dwBufferBytes = 0;
+ dsbd.lpwfxFormat = NULL;
+
+ hr = IDirectSound_CreateSoundBuffer (
+ s->dsound,
+ &dsbd,
+ &s->dsound_primary_buffer,
+ NULL
+ );
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not create primary playback buffer\n");
+ return -1;
+ }
+
+ hr = IDirectSoundBuffer_SetFormat (s->dsound_primary_buffer, &wfx);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not set primary playback buffer format\n");
+ }
+
+ hr = IDirectSoundBuffer_GetFormat (
+ s->dsound_primary_buffer,
+ &wfx,
+ sizeof (wfx),
+ NULL
+ );
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not get primary playback buffer format\n");
+ goto fail0;
+ }
+
+#ifdef DEBUG_DSOUND
+ dolog ("Primary\n");
+ print_wave_format (&wfx);
+#endif
+
+ err = waveformat_to_full_fmt (&wfx, &s->fmt);
+ if (err) {
+ goto fail0;
+ }
+
+ return 0;
+
+ fail0:
+ dsound_close (s);
+ return -1;
+}
+
+static int dsound_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
+ HRESULT hr;
+ DWORD status;
+ DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+ LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+
+ if (!dsb) {
+ dolog ("Attempt to control voice without a buffer\n");
+ return 0;
+ }
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ if (dsound_get_status_out (dsb, &status)) {
+ return -1;
+ }
+
+ if (status & DSBSTATUS_PLAYING) {
+ dolog ("warning: voice is already playing\n");
+ return 0;
+ }
+
+ dsound_clear_sample (hw, dsb);
+
+ hr = IDirectSoundBuffer_Play (dsb, 0, 0, DSBPLAY_LOOPING);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not start playing buffer\n");
+ return -1;
+ }
+ break;
+
+ case VOICE_DISABLE:
+ if (dsound_get_status_out (dsb, &status)) {
+ return -1;
+ }
+
+ if (status & DSBSTATUS_PLAYING) {
+ hr = IDirectSoundBuffer_Stop (dsb);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not stop playing buffer\n");
+ return -1;
+ }
+ }
+ else {
+ dolog ("warning: voice is not playing\n");
+ }
+ break;
+ }
+ return 0;
+}
+
+static int dsound_write (SWVoiceOut *sw, void *buf, int len)
+{
+ return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int dsound_run_out (HWVoiceOut *hw)
+{
+ int err;
+ HRESULT hr;
+ DSoundVoiceOut *ds = (DSoundVoiceOut *) hw;
+ LPDIRECTSOUNDBUFFER dsb = ds->dsound_buffer;
+ int live, len, hwshift;
+ DWORD blen1, blen2;
+ DWORD len1, len2;
+ DWORD decr;
+ DWORD wpos, ppos, old_pos;
+ LPVOID p1, p2;
+
+ if (!dsb) {
+ dolog ("Attempt to run empty with playback buffer\n");
+ return 0;
+ }
+
+ hwshift = hw->info.shift;
+
+ live = audio_pcm_hw_get_live_out (hw);
+
+ hr = IDirectSoundBuffer_GetCurrentPosition (
+ dsb,
+ &ppos,
+ ds->first_time ? &wpos : NULL
+ );
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not get playback buffer position\n");
+ return 0;
+ }
+
+ len = live << hwshift;
+
+ if (ds->first_time) {
+ if (conf.latency_millis) {
+ DWORD cur_blat = audio_ring_dist (wpos, ppos, hw->bufsize);
+
+ ds->first_time = 0;
+ old_pos = wpos;
+ old_pos +=
+ millis_to_bytes (&hw->info, conf.latency_millis) - cur_blat;
+ old_pos %= hw->bufsize;
+ old_pos &= ~hw->info.align;
+ }
+ else {
+ old_pos = wpos;
+ }
+#ifdef DEBUG_DSOUND
+ ds->played = 0;
+ ds->mixed = 0;
+#endif
+ }
+ else {
+ if (ds->old_pos == ppos) {
+#ifdef DEBUG_DSOUND
+ dolog ("old_pos == ppos\n");
+#endif
+ return 0;
+ }
+
+#ifdef DEBUG_DSOUND
+ ds->played += audio_ring_dist (ds->old_pos, ppos, hw->bufsize);
+#endif
+ old_pos = ds->old_pos;
+ }
+
+ if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
+ len = ppos - old_pos;
+ }
+ else {
+ if ((old_pos > ppos) && ((old_pos + len) > (ppos + hw->bufsize))) {
+ len = hw->bufsize - old_pos + ppos;
+ }
+ }
+
+ if (audio_bug (AUDIO_FUNC, len < 0 || len > hw->bufsize)) {
+ dolog ("len=%d hw->bufsize=%d old_pos=%ld ppos=%ld\n",
+ len, hw->bufsize, old_pos, ppos);
+ return 0;
+ }
+
+ len &= ~hw->info.align;
+ if (!len) {
+ return 0;
+ }
+
+#ifdef DEBUG_DSOUND
+ ds->old_ppos = ppos;
+#endif
+ err = dsound_lock_out (
+ dsb,
+ &hw->info,
+ old_pos,
+ len,
+ &p1, &p2,
+ &blen1, &blen2,
+ 0
+ );
+ if (err) {
+ return 0;
+ }
+
+ len1 = blen1 >> hwshift;
+ len2 = blen2 >> hwshift;
+ decr = len1 + len2;
+
+ if (p1 && len1) {
+ dsound_write_sample (hw, p1, len1);
+ }
+
+ if (p2 && len2) {
+ dsound_write_sample (hw, p2, len2);
+ }
+
+ dsound_unlock_out (dsb, p1, p2, blen1, blen2);
+ ds->old_pos = (old_pos + (decr << hwshift)) % hw->bufsize;
+
+#ifdef DEBUG_DSOUND
+ ds->mixed += decr << hwshift;
+
+ dolog ("played %lu mixed %lu diff %ld sec %f\n",
+ ds->played,
+ ds->mixed,
+ ds->mixed - ds->played,
+ abs (ds->mixed - ds->played) / (double) hw->info.bytes_per_second);
+#endif
+ return decr;
+}
+
+static int dsound_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+ HRESULT hr;
+ DWORD status;
+ DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+ LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
+
+ if (!dscb) {
+ dolog ("Attempt to control capture voice without a buffer\n");
+ return -1;
+ }
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ if (dsound_get_status_in (dscb, &status)) {
+ return -1;
+ }
+
+ if (status & DSCBSTATUS_CAPTURING) {
+ dolog ("warning: voice is already capturing\n");
+ return 0;
+ }
+
+ /* clear ?? */
+
+ hr = IDirectSoundCaptureBuffer_Start (dscb, DSCBSTART_LOOPING);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not start capturing\n");
+ return -1;
+ }
+ break;
+
+ case VOICE_DISABLE:
+ if (dsound_get_status_in (dscb, &status)) {
+ return -1;
+ }
+
+ if (status & DSCBSTATUS_CAPTURING) {
+ hr = IDirectSoundCaptureBuffer_Stop (dscb);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not stop capturing\n");
+ return -1;
+ }
+ }
+ else {
+ dolog ("warning: voice is not capturing\n");
+ }
+ break;
+ }
+ return 0;
+}
+
+static int dsound_read (SWVoiceIn *sw, void *buf, int len)
+{
+ return audio_pcm_sw_read (sw, buf, len);
+}
+
+static int dsound_run_in (HWVoiceIn *hw)
+{
+ int err;
+ HRESULT hr;
+ DSoundVoiceIn *ds = (DSoundVoiceIn *) hw;
+ LPDIRECTSOUNDCAPTUREBUFFER dscb = ds->dsound_capture_buffer;
+ int live, len, dead;
+ DWORD blen1, blen2;
+ DWORD len1, len2;
+ DWORD decr;
+ DWORD cpos, rpos;
+ LPVOID p1, p2;
+ int hwshift;
+
+ if (!dscb) {
+ dolog ("Attempt to run without capture buffer\n");
+ return 0;
+ }
+
+ hwshift = hw->info.shift;
+
+ live = audio_pcm_hw_get_live_in (hw);
+ dead = hw->samples - live;
+ if (!dead) {
+ return 0;
+ }
+
+ hr = IDirectSoundCaptureBuffer_GetCurrentPosition (
+ dscb,
+ &cpos,
+ ds->first_time ? &rpos : NULL
+ );
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not get capture buffer position\n");
+ return 0;
+ }
+
+ if (ds->first_time) {
+ ds->first_time = 0;
+ if (rpos & hw->info.align) {
+ ldebug ("warning: misaligned capture read position %ld(%d)\n",
+ rpos, hw->info.align);
+ }
+ hw->wpos = rpos >> hwshift;
+ }
+
+ if (cpos & hw->info.align) {
+ ldebug ("warning: misaligned capture position %ld(%d)\n",
+ cpos, hw->info.align);
+ }
+ cpos >>= hwshift;
+
+ len = audio_ring_dist (cpos, hw->wpos, hw->samples);
+ if (!len) {
+ return 0;
+ }
+ len = audio_MIN (len, dead);
+
+ err = dsound_lock_in (
+ dscb,
+ &hw->info,
+ hw->wpos << hwshift,
+ len << hwshift,
+ &p1,
+ &p2,
+ &blen1,
+ &blen2,
+ 0
+ );
+ if (err) {
+ return 0;
+ }
+
+ len1 = blen1 >> hwshift;
+ len2 = blen2 >> hwshift;
+ decr = len1 + len2;
+
+ if (p1 && len1) {
+ hw->conv (hw->conv_buf + hw->wpos, p1, len1, &nominal_volume);
+ }
+
+ if (p2 && len2) {
+ hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
+ }
+
+ dsound_unlock_in (dscb, p1, p2, blen1, blen2);
+ hw->wpos = (hw->wpos + decr) % hw->samples;
+ return decr;
+}
+
+static void dsound_audio_fini (void *opaque)
+{
+ HRESULT hr;
+ dsound *s = opaque;
+
+ if (!s->dsound) {
+ return;
+ }
+
+ hr = IDirectSound_Release (s->dsound);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not release DirectSound\n");
+ }
+ s->dsound = NULL;
+
+ if (!s->dsound_capture) {
+ return;
+ }
+
+ hr = IDirectSoundCapture_Release (s->dsound_capture);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not release DirectSoundCapture\n");
+ }
+ s->dsound_capture = NULL;
+}
+
+static void *dsound_audio_init (void)
+{
+ int err;
+ HRESULT hr;
+ dsound *s = &glob_dsound;
+
+ hr = CoInitialize (NULL);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not initialize COM\n");
+ return NULL;
+ }
+
+ hr = CoCreateInstance (
+ &CLSID_DirectSound,
+ NULL,
+ CLSCTX_ALL,
+ &IID_IDirectSound,
+ (void **) &s->dsound
+ );
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not create DirectSound instance\n");
+ return NULL;
+ }
+
+ hr = IDirectSound_Initialize (s->dsound, NULL);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not initialize DirectSound\n");
+ return NULL;
+ }
+
+ hr = CoCreateInstance (
+ &CLSID_DirectSoundCapture,
+ NULL,
+ CLSCTX_ALL,
+ &IID_IDirectSoundCapture,
+ (void **) &s->dsound_capture
+ );
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not create DirectSoundCapture instance\n");
+ }
+ else {
+ hr = IDirectSoundCapture_Initialize (s->dsound_capture, NULL);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not initialize DirectSoundCapture\n");
+
+ hr = IDirectSoundCapture_Release (s->dsound_capture);
+ if (FAILED (hr)) {
+ dsound_logerr (hr, "Can not release DirectSoundCapture\n");
+ }
+ s->dsound_capture = NULL;
+ }
+ }
+
+ err = dsound_open (s);
+ if (err) {
+ dsound_audio_fini (s);
+ return NULL;
+ }
+
+ return s;
+}
+
+static struct audio_option dsound_options[] = {
+ {"LOCK_RETRIES", AUD_OPT_INT, &conf.lock_retries,
+ "Number of times to attempt locking the buffer", NULL, 0},
+ {"RESTOURE_RETRIES", AUD_OPT_INT, &conf.restore_retries,
+ "Number of times to attempt restoring the buffer", NULL, 0},
+ {"GETSTATUS_RETRIES", AUD_OPT_INT, &conf.getstatus_retries,
+ "Number of times to attempt getting status of the buffer", NULL, 0},
+ {"SET_PRIMARY", AUD_OPT_BOOL, &conf.set_primary,
+ "Set the parameters of primary buffer", NULL, 0},
+ {"LATENCY_MILLIS", AUD_OPT_INT, &conf.latency_millis,
+ "(undocumented)", NULL, 0},
+ {"PRIMARY_FREQ", AUD_OPT_INT, &conf.full_fmt.freq,
+ "Primary buffer frequency", NULL, 0},
+ {"PRIMARY_CHANNELS", AUD_OPT_INT, &conf.full_fmt.nchannels,
+ "Primary buffer number of channels (1 - mono, 2 - stereo)", NULL, 0},
+ {"PRIMARY_FMT", AUD_OPT_FMT, &conf.full_fmt.fmt,
+ "Primary buffer format", NULL, 0},
+ {"BUFSIZE_OUT", AUD_OPT_INT, &conf.bufsize_out,
+ "(undocumented)", NULL, 0},
+ {"BUFSIZE_IN", AUD_OPT_INT, &conf.bufsize_in,
+ "(undocumented)", NULL, 0},
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops dsound_pcm_ops = {
+ dsound_init_out,
+ dsound_fini_out,
+ dsound_run_out,
+ dsound_write,
+ dsound_ctl_out,
+
+ dsound_init_in,
+ dsound_fini_in,
+ dsound_run_in,
+ dsound_read,
+ dsound_ctl_in
+};
+
+struct audio_driver dsound_audio_driver = {
+ INIT_FIELD (name = ) "dsound",
+ INIT_FIELD (descr = )
+ "DirectSound http://wikipedia.org/wiki/DirectSound",
+ INIT_FIELD (options = ) dsound_options,
+ INIT_FIELD (init = ) dsound_audio_init,
+ INIT_FIELD (fini = ) dsound_audio_fini,
+ INIT_FIELD (pcm_ops = ) &dsound_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) INT_MAX,
+ INIT_FIELD (max_voices_in = ) 1,
+ INIT_FIELD (voice_size_out = ) sizeof (DSoundVoiceOut),
+ INIT_FIELD (voice_size_in = ) sizeof (DSoundVoiceIn)
+};
diff --git a/audio/fmodaudio.c b/audio/fmodaudio.c
index 7b79026a82..36b8d47c43 100644
--- a/audio/fmodaudio.c
+++ b/audio/fmodaudio.c
@@ -1,8 +1,8 @@
/*
- * QEMU FMOD audio output driver
- *
- * Copyright (c) 2004 Vassili Karpov (malc)
- *
+ * QEMU FMOD audio driver
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -25,53 +25,77 @@
#include <fmod_errors.h>
#include "vl.h"
-#include "audio/audio_int.h"
+#define AUDIO_CAP "fmod"
+#include "audio_int.h"
-typedef struct FMODVoice {
- HWVoice hw;
+typedef struct FMODVoiceOut {
+ HWVoiceOut hw;
unsigned int old_pos;
FSOUND_SAMPLE *fmod_sample;
int channel;
-} FMODVoice;
-
-#define dolog(...) AUD_log ("fmod", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+} FMODVoiceOut;
-#define QC_FMOD_DRV "QEMU_FMOD_DRV"
-#define QC_FMOD_FREQ "QEMU_FMOD_FREQ"
-#define QC_FMOD_SAMPLES "QEMU_FMOD_SAMPLES"
-#define QC_FMOD_CHANNELS "QEMU_FMOD_CHANNELS"
-#define QC_FMOD_BUFSIZE "QEMU_FMOD_BUFSIZE"
-#define QC_FMOD_THRESHOLD "QEMU_FMOD_THRESHOLD"
+typedef struct FMODVoiceIn {
+ HWVoiceIn hw;
+ FSOUND_SAMPLE *fmod_sample;
+} FMODVoiceIn;
static struct {
+ const char *drvname;
int nb_samples;
int freq;
int nb_channels;
int bufsize;
int threshold;
+ int broken_adc;
} conf = {
- 2048,
+ NULL,
+ 2048 * 2,
44100,
- 1,
+ 2,
+ 0,
0,
- 128
+ 0
};
-#define errstr() FMOD_ErrorString (FSOUND_GetError ())
+static void GCC_FMT_ATTR (1, 2) fmod_logerr (const char *fmt, ...)
+{
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n",
+ FMOD_ErrorString (FSOUND_GetError ()));
+}
+
+static void GCC_FMT_ATTR (2, 3) fmod_logerr2 (
+ const char *typ,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n",
+ FMOD_ErrorString (FSOUND_GetError ()));
+}
-static int fmod_hw_write (SWVoice *sw, void *buf, int len)
+static int fmod_write (SWVoiceOut *sw, void *buf, int len)
{
- return pcm_hw_write (sw, buf, len);
+ return audio_pcm_sw_write (sw, buf, len);
}
-static void fmod_clear_sample (FMODVoice *fmd)
+static void fmod_clear_sample (FMODVoiceOut *fmd)
{
- HWVoice *hw = &fmd->hw;
+ HWVoiceOut *hw = &fmd->hw;
int status;
void *p1 = 0, *p2 = 0;
unsigned int len1 = 0, len2 = 0;
@@ -79,7 +103,7 @@ static void fmod_clear_sample (FMODVoice *fmd)
status = FSOUND_Sample_Lock (
fmd->fmod_sample,
0,
- hw->samples << hw->shift,
+ hw->samples << hw->info.shift,
&p1,
&p2,
&len1,
@@ -87,78 +111,88 @@ static void fmod_clear_sample (FMODVoice *fmd)
);
if (!status) {
- dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+ fmod_logerr ("Failed to lock sample\n");
return;
}
- if ((len1 & hw->align) || (len2 & hw->align)) {
- dolog ("Locking sample returned unaligned length %d, %d\n",
- len1, len2);
+ if ((len1 & hw->info.align) || (len2 & hw->info.align)) {
+ dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
+ len1, len2, hw->info.align + 1);
goto fail;
}
- if (len1 + len2 != hw->samples << hw->shift) {
- dolog ("Locking sample returned incomplete length %d, %d\n",
- len1 + len2, hw->samples << hw->shift);
+ if ((len1 + len2) - (hw->samples << hw->info.shift)) {
+ dolog ("Lock returned incomplete length %d, %d\n",
+ len1 + len2, hw->samples << hw->info.shift);
goto fail;
}
- pcm_hw_clear (hw, p1, hw->samples);
+
+ audio_pcm_info_clear_buf (&hw->info, p1, hw->samples);
fail:
status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, len1, len2);
if (!status) {
- dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+ fmod_logerr ("Failed to unlock sample\n");
}
}
-static int fmod_write_sample (HWVoice *hw, uint8_t *dst, st_sample_t *src,
- int src_size, int src_pos, int dst_len)
+static void fmod_write_sample (HWVoiceOut *hw, uint8_t *dst, int dst_len)
{
- int src_len1 = dst_len, src_len2 = 0, pos = src_pos + dst_len;
- st_sample_t *src1 = src + src_pos, *src2 = 0;
-
- if (src_pos + dst_len > src_size) {
- src_len1 = src_size - src_pos;
- src2 = src;
+ int src_len1 = dst_len;
+ int src_len2 = 0;
+ int pos = hw->rpos + dst_len;
+ st_sample_t *src1 = hw->mix_buf + hw->rpos;
+ st_sample_t *src2 = NULL;
+
+ if (pos > hw->samples) {
+ src_len1 = hw->samples - hw->rpos;
+ src2 = hw->mix_buf;
src_len2 = dst_len - src_len1;
pos = src_len2;
}
if (src_len1) {
hw->clip (dst, src1, src_len1);
- memset (src1, 0, src_len1 * sizeof (st_sample_t));
- advance (dst, src_len1);
+ mixeng_clear (src1, src_len1);
}
if (src_len2) {
+ dst = advance (dst, src_len1 << hw->info.shift);
hw->clip (dst, src2, src_len2);
- memset (src2, 0, src_len2 * sizeof (st_sample_t));
+ mixeng_clear (src2, src_len2);
}
- return pos;
+
+ hw->rpos = pos % hw->samples;
}
-static int fmod_unlock_sample (FMODVoice *fmd, void *p1, void *p2,
+static int fmod_unlock_sample (FSOUND_SAMPLE *sample, void *p1, void *p2,
unsigned int blen1, unsigned int blen2)
{
- int status = FSOUND_Sample_Unlock (fmd->fmod_sample, p1, p2, blen1, blen2);
+ int status = FSOUND_Sample_Unlock (sample, p1, p2, blen1, blen2);
if (!status) {
- dolog ("Failed to unlock sample\nReason: %s\n", errstr ());
+ fmod_logerr ("Failed to unlock sample\n");
return -1;
}
return 0;
}
-static int fmod_lock_sample (FMODVoice *fmd, int pos, int len,
- void **p1, void **p2,
- unsigned int *blen1, unsigned int *blen2)
+static int fmod_lock_sample (
+ FSOUND_SAMPLE *sample,
+ struct audio_pcm_info *info,
+ int pos,
+ int len,
+ void **p1,
+ void **p2,
+ unsigned int *blen1,
+ unsigned int *blen2
+ )
{
- HWVoice *hw = &fmd->hw;
int status;
status = FSOUND_Sample_Lock (
- fmd->fmod_sample,
- pos << hw->shift,
- len << hw->shift,
+ sample,
+ pos << info->shift,
+ len << info->shift,
p1,
p2,
blen1,
@@ -166,89 +200,117 @@ static int fmod_lock_sample (FMODVoice *fmd, int pos, int len,
);
if (!status) {
- dolog ("Failed to lock sample\nReason: %s\n", errstr ());
+ fmod_logerr ("Failed to lock sample\n");
return -1;
}
- if ((*blen1 & hw->align) || (*blen2 & hw->align)) {
- dolog ("Locking sample returned unaligned length %d, %d\n",
- *blen1, *blen2);
- fmod_unlock_sample (fmd, *p1, *p2, *blen1, *blen2);
+ if ((*blen1 & info->align) || (*blen2 & info->align)) {
+ dolog ("Lock returned misaligned length %d, %d, alignment %d\n",
+ *blen1, *blen2, info->align + 1);
+
+ fmod_unlock_sample (sample, *p1, *p2, *blen1, *blen2);
+
+ *p1 = NULL - 1;
+ *p2 = NULL - 1;
+ *blen1 = ~0U;
+ *blen2 = ~0U;
return -1;
}
+
+ if (!*p1 && *blen1) {
+ dolog ("warning: !p1 && blen1=%d\n", *blen1);
+ *blen1 = 0;
+ }
+
+ if (!p2 && *blen2) {
+ dolog ("warning: !p2 && blen2=%d\n", *blen2);
+ *blen2 = 0;
+ }
+
return 0;
}
-static void fmod_hw_run (HWVoice *hw)
+static int fmod_run_out (HWVoiceOut *hw)
{
- FMODVoice *fmd = (FMODVoice *) hw;
- int rpos, live, decr;
+ FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
+ int live, decr;
void *p1 = 0, *p2 = 0;
unsigned int blen1 = 0, blen2 = 0;
unsigned int len1 = 0, len2 = 0;
- int nb_active;
+ int nb_live;
- live = pcm_hw_get_live2 (hw, &nb_active);
- if (live <= 0) {
- return;
+ live = audio_pcm_hw_get_live_out2 (hw, &nb_live);
+ if (!live) {
+ return 0;
}
if (!hw->pending_disable
- && nb_active
- && conf.threshold
- && live <= conf.threshold) {
- ldebug ("live=%d nb_active=%d\n", live, nb_active);
- return;
+ && nb_live
+ && (conf.threshold && live <= conf.threshold)) {
+ ldebug ("live=%d nb_live=%d\n", live, nb_live);
+ return 0;
}
decr = live;
-#if 1
if (fmd->channel >= 0) {
- int pos2 = (fmd->old_pos + decr) % hw->samples;
- int pos = FSOUND_GetCurrentPosition (fmd->channel);
+ int len = decr;
+ int old_pos = fmd->old_pos;
+ int ppos = FSOUND_GetCurrentPosition (fmd->channel);
- if (fmd->old_pos < pos && pos2 >= pos) {
- decr = pos - fmd->old_pos - (pos2 == pos) - 1;
+ if (ppos == old_pos || !ppos) {
+ return 0;
}
- else if (fmd->old_pos > pos && pos2 >= pos && pos2 < fmd->old_pos) {
- decr = (hw->samples - fmd->old_pos) + pos - (pos2 == pos) - 1;
+
+ if ((old_pos < ppos) && ((old_pos + len) > ppos)) {
+ len = ppos - old_pos;
+ }
+ else {
+ if ((old_pos > ppos) && ((old_pos + len) > (ppos + hw->samples))) {
+ len = hw->samples - old_pos + ppos;
+ }
+ }
+ decr = len;
+
+ if (audio_bug (AUDIO_FUNC, decr < 0)) {
+ dolog ("decr=%d live=%d ppos=%d old_pos=%d len=%d\n",
+ decr, live, ppos, old_pos, len);
+ return 0;
}
-/* ldebug ("pos=%d pos2=%d old=%d live=%d decr=%d\n", */
-/* pos, pos2, fmd->old_pos, live, decr); */
}
-#endif
- if (decr <= 0) {
- return;
+
+ if (!decr) {
+ return 0;
}
- if (fmod_lock_sample (fmd, fmd->old_pos, decr, &p1, &p2, &blen1, &blen2)) {
- return;
+ if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
+ fmd->old_pos, decr,
+ &p1, &p2,
+ &blen1, &blen2)) {
+ return 0;
}
- len1 = blen1 >> hw->shift;
- len2 = blen2 >> hw->shift;
+ len1 = blen1 >> hw->info.shift;
+ len2 = blen2 >> hw->info.shift;
ldebug ("%p %p %d %d %d %d\n", p1, p2, len1, len2, blen1, blen2);
decr = len1 + len2;
- rpos = hw->rpos;
- if (len1) {
- rpos = fmod_write_sample (hw, p1, hw->mix_buf, hw->samples, rpos, len1);
+ if (p1 && len1) {
+ fmod_write_sample (hw, p1, len1);
}
- if (len2) {
- rpos = fmod_write_sample (hw, p2, hw->mix_buf, hw->samples, rpos, len2);
+ if (p2 && len2) {
+ fmod_write_sample (hw, p2, len2);
}
- fmod_unlock_sample (fmd, p1, p2, blen1, blen2);
+ fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
- pcm_hw_dec_live (hw, decr);
- hw->rpos = rpos % hw->samples;
fmd->old_pos = (fmd->old_pos + decr) % hw->samples;
+ return decr;
}
-static int AUD_to_fmodfmt (audfmt_e fmt, int stereo)
+static int aud_to_fmodfmt (audfmt_e fmt, int stereo)
{
int mode = FSOUND_LOOP_NORMAL;
@@ -270,16 +332,19 @@ static int AUD_to_fmodfmt (audfmt_e fmt, int stereo)
break;
default:
- dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
- exit (EXIT_FAILURE);
+ dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_FMOD
+ abort ();
+#endif
+ mode |= FSOUND_8BITS;
}
mode |= stereo ? FSOUND_STEREO : FSOUND_MONO;
return mode;
}
-static void fmod_hw_fini (HWVoice *hw)
+static void fmod_fini_out (HWVoiceOut *hw)
{
- FMODVoice *fmd = (FMODVoice *) hw;
+ FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
if (fmd->fmod_sample) {
FSOUND_Sample_Free (fmd->fmod_sample);
@@ -291,12 +356,12 @@ static void fmod_hw_fini (HWVoice *hw)
}
}
-static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int fmod_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
{
int bits16, mode, channel;
- FMODVoice *fmd = (FMODVoice *) hw;
+ FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
- mode = AUD_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+ mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
fmd->fmod_sample = FSOUND_Sample_Alloc (
FSOUND_FREE, /* index */
conf.nb_samples, /* length */
@@ -308,52 +373,145 @@ static int fmod_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
);
if (!fmd->fmod_sample) {
- dolog ("Failed to allocate FMOD sample\nReason: %s\n", errstr ());
+ fmod_logerr2 ("DAC", "Failed to allocate FMOD sample\n");
return -1;
}
channel = FSOUND_PlaySoundEx (FSOUND_FREE, fmd->fmod_sample, 0, 1);
if (channel < 0) {
- dolog ("Failed to start playing sound\nReason: %s\n", errstr ());
+ fmod_logerr2 ("DAC", "Failed to start playing sound\n");
FSOUND_Sample_Free (fmd->fmod_sample);
return -1;
}
fmd->channel = channel;
- hw->freq = freq;
- hw->fmt = fmt;
- hw->nchannels = nchannels;
- bits16 = fmt == AUD_FMT_U16 || fmt == AUD_FMT_S16;
+ /* FMOD always operates on little endian frames? */
+ audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
+ audio_need_to_swap_endian (0));
+ bits16 = (mode & FSOUND_16BITS) != 0;
hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
return 0;
}
-static int fmod_hw_ctl (HWVoice *hw, int cmd, ...)
+static int fmod_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int status;
- FMODVoice *fmd = (FMODVoice *) hw;
+ FMODVoiceOut *fmd = (FMODVoiceOut *) hw;
switch (cmd) {
case VOICE_ENABLE:
fmod_clear_sample (fmd);
status = FSOUND_SetPaused (fmd->channel, 0);
if (!status) {
- dolog ("Failed to resume channel %d\nReason: %s\n",
- fmd->channel, errstr ());
+ fmod_logerr ("Failed to resume channel %d\n", fmd->channel);
}
break;
case VOICE_DISABLE:
status = FSOUND_SetPaused (fmd->channel, 1);
if (!status) {
- dolog ("Failed to pause channel %d\nReason: %s\n",
- fmd->channel, errstr ());
+ fmod_logerr ("Failed to pause channel %d\n", fmd->channel);
}
break;
}
return 0;
}
+static int fmod_init_in (HWVoiceIn *hw, int freq, int nchannels, audfmt_e fmt)
+{
+ int bits16, mode;
+ FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+ if (conf.broken_adc) {
+ return -1;
+ }
+
+ mode = aud_to_fmodfmt (fmt, nchannels == 2 ? 1 : 0);
+ fmd->fmod_sample = FSOUND_Sample_Alloc (
+ FSOUND_FREE, /* index */
+ conf.nb_samples, /* length */
+ mode, /* mode */
+ freq, /* freq */
+ 255, /* volume */
+ 128, /* pan */
+ 255 /* priority */
+ );
+
+ if (!fmd->fmod_sample) {
+ fmod_logerr2 ("ADC", "Failed to allocate FMOD sample\n");
+ return -1;
+ }
+
+ /* FMOD always operates on little endian frames? */
+ audio_pcm_init_info (&hw->info, freq, nchannels, fmt,
+ audio_need_to_swap_endian (0));
+ bits16 = (mode & FSOUND_16BITS) != 0;
+ hw->bufsize = conf.nb_samples << (nchannels == 2) << bits16;
+ return 0;
+}
+
+static void fmod_fini_in (HWVoiceIn *hw)
+{
+ FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+ if (fmd->fmod_sample) {
+ FSOUND_Record_Stop ();
+ FSOUND_Sample_Free (fmd->fmod_sample);
+ fmd->fmod_sample = 0;
+ }
+}
+
+static int fmod_run_in (HWVoiceIn *hw)
+{
+ FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+ int hwshift = hw->info.shift;
+ int live, dead, new_pos, len;
+ unsigned int blen1 = 0, blen2 = 0;
+ unsigned int len1, len2;
+ unsigned int decr;
+ void *p1, *p2;
+
+ live = audio_pcm_hw_get_live_in (hw);
+ dead = hw->samples - live;
+ if (!dead) {
+ return 0;
+ }
+
+ new_pos = FSOUND_Record_GetPosition ();
+ if (new_pos < 0) {
+ fmod_logerr ("Can not get recording position\n");
+ return 0;
+ }
+
+ len = audio_ring_dist (new_pos, hw->wpos, hw->samples);
+ if (!len) {
+ return 0;
+ }
+ len = audio_MIN (len, dead);
+
+ if (fmod_lock_sample (fmd->fmod_sample, &fmd->hw.info,
+ hw->wpos, len,
+ &p1, &p2,
+ &blen1, &blen2)) {
+ return 0;
+ }
+
+ len1 = blen1 >> hwshift;
+ len2 = blen2 >> hwshift;
+ decr = len1 + len2;
+
+ if (p1 && blen1) {
+ hw->conv (hw->conv_buf + hw->wpos, p1, len1, &nominal_volume);
+ }
+ if (p2 && len2) {
+ hw->conv (hw->conv_buf, p2, len2, &nominal_volume);
+ }
+
+ fmod_unlock_sample (fmd->fmod_sample, p1, p2, blen1, blen2);
+ hw->wpos = (hw->wpos + decr) % hw->samples;
+ return decr;
+}
+
static struct {
const char *name;
int type;
@@ -378,16 +536,16 @@ static struct {
{"ps2", FSOUND_OUTPUT_PS2},
{"gcube", FSOUND_OUTPUT_GC},
#endif
- {"nort", FSOUND_OUTPUT_NOSOUND_NONREALTIME}
+ {"none-realtime", FSOUND_OUTPUT_NOSOUND_NONREALTIME}
};
static void *fmod_audio_init (void)
{
- int i;
+ size_t i;
double ver;
int status;
int output_type = -1;
- const char *drv = audio_get_conf_str (QC_FMOD_DRV, NULL);
+ const char *drv = conf.drvname;
ver = FSOUND_GetVersion ();
if (ver < FMOD_VERSION) {
@@ -395,6 +553,14 @@ static void *fmod_audio_init (void)
return NULL;
}
+#ifdef __linux__
+ if (ver < 3.75) {
+ dolog ("FMOD before 3.75 has bug preventing ADC from working\n"
+ "ADC will be disabled.\n");
+ conf.broken_adc = 1;
+ }
+#endif
+
if (drv) {
int found = 0;
for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
@@ -405,65 +571,115 @@ static void *fmod_audio_init (void)
}
}
if (!found) {
- dolog ("Unknown FMOD output driver `%s'\n", drv);
+ dolog ("Unknown FMOD driver `%s'\n", drv);
+ dolog ("Valid drivers:\n");
+ for (i = 0; i < sizeof (drvtab) / sizeof (drvtab[0]); i++) {
+ dolog (" %s\n", drvtab[i].name);
+ }
}
}
if (output_type != -1) {
status = FSOUND_SetOutput (output_type);
if (!status) {
- dolog ("FSOUND_SetOutput(%d) failed\nReason: %s\n",
- output_type, errstr ());
+ fmod_logerr ("FSOUND_SetOutput(%d) failed\n", output_type);
return NULL;
}
}
- conf.freq = audio_get_conf_int (QC_FMOD_FREQ, conf.freq);
- conf.nb_samples = audio_get_conf_int (QC_FMOD_SAMPLES, conf.nb_samples);
- conf.nb_channels =
- audio_get_conf_int (QC_FMOD_CHANNELS,
- (audio_state.nb_hw_voices > 1
- ? audio_state.nb_hw_voices
- : conf.nb_channels));
- conf.bufsize = audio_get_conf_int (QC_FMOD_BUFSIZE, conf.bufsize);
- conf.threshold = audio_get_conf_int (QC_FMOD_THRESHOLD, conf.threshold);
-
if (conf.bufsize) {
status = FSOUND_SetBufferSize (conf.bufsize);
if (!status) {
- dolog ("FSOUND_SetBufferSize (%d) failed\nReason: %s\n",
- conf.bufsize, errstr ());
+ fmod_logerr ("FSOUND_SetBufferSize (%d) failed\n", conf.bufsize);
}
}
status = FSOUND_Init (conf.freq, conf.nb_channels, 0);
if (!status) {
- dolog ("FSOUND_Init failed\nReason: %s\n", errstr ());
+ fmod_logerr ("FSOUND_Init failed\n");
return NULL;
}
return &conf;
}
+static int fmod_read (SWVoiceIn *sw, void *buf, int size)
+{
+ return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int fmod_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+ int status;
+ FMODVoiceIn *fmd = (FMODVoiceIn *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ status = FSOUND_Record_StartSample (fmd->fmod_sample, 1);
+ if (!status) {
+ fmod_logerr ("Failed to start recording\n");
+ }
+ break;
+
+ case VOICE_DISABLE:
+ status = FSOUND_Record_Stop ();
+ if (!status) {
+ fmod_logerr ("Failed to stop recording\n");
+ }
+ break;
+ }
+ return 0;
+}
+
static void fmod_audio_fini (void *opaque)
{
+ (void) opaque;
FSOUND_Close ();
}
-struct pcm_ops fmod_pcm_ops = {
- fmod_hw_init,
- fmod_hw_fini,
- fmod_hw_run,
- fmod_hw_write,
- fmod_hw_ctl
+static struct audio_option fmod_options[] = {
+ {"DRV", AUD_OPT_STR, &conf.drvname,
+ "FMOD driver", NULL, 0},
+ {"FREQ", AUD_OPT_INT, &conf.freq,
+ "Default frequency", NULL, 0},
+ {"SAMPLES", AUD_OPT_INT, &conf.nb_samples,
+ "Buffer size in samples", NULL, 0},
+ {"CHANNELS", AUD_OPT_INT, &conf.nb_channels,
+ "Number of default channels (1 - mono, 2 - stereo)", NULL, 0},
+ {"BUFSIZE", AUD_OPT_INT, &conf.bufsize,
+ "(undocumented)", NULL, 0},
+#if 0
+ {"THRESHOLD", AUD_OPT_INT, &conf.threshold,
+ "(undocumented)"},
+#endif
+
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops fmod_pcm_ops = {
+ fmod_init_out,
+ fmod_fini_out,
+ fmod_run_out,
+ fmod_write,
+ fmod_ctl_out,
+
+ fmod_init_in,
+ fmod_fini_in,
+ fmod_run_in,
+ fmod_read,
+ fmod_ctl_in
};
-struct audio_output_driver fmod_output_driver = {
- "fmod",
- fmod_audio_init,
- fmod_audio_fini,
- &fmod_pcm_ops,
- 1,
- INT_MAX,
- sizeof (FMODVoice)
+struct audio_driver fmod_audio_driver = {
+ INIT_FIELD (name = ) "fmod",
+ INIT_FIELD (descr = ) "FMOD 3.xx http://www.fmod.org",
+ INIT_FIELD (options = ) fmod_options,
+ INIT_FIELD (init = ) fmod_audio_init,
+ INIT_FIELD (fini = ) fmod_audio_fini,
+ INIT_FIELD (pcm_ops = ) &fmod_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) INT_MAX,
+ INIT_FIELD (max_voices_in = ) INT_MAX,
+ INIT_FIELD (voice_size_out = ) sizeof (FMODVoiceOut),
+ INIT_FIELD (voice_size_in = ) sizeof (FMODVoiceIn)
};
diff --git a/audio/mixeng.c b/audio/mixeng.c
index b0bb412c63..d43c5e59d7 100644
--- a/audio/mixeng.c
+++ b/audio/mixeng.c
@@ -1,7 +1,7 @@
/*
* QEMU Mixing engine
*
- * Copyright (c) 2004 Vassili Karpov (malc)
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
* Copyright (c) 1998 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
@@ -23,87 +23,174 @@
* THE SOFTWARE.
*/
#include "vl.h"
-//#define DEBUG_FP
-#include "audio/mixeng.h"
+#define AUDIO_CAP "mixeng"
+#include "audio_int.h"
+
+#define NOVOL
+
+/* 8 bit */
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+
+/* Signed 8 bit */
#define IN_T int8_t
-#define IN_MIN CHAR_MIN
-#define IN_MAX CHAR_MAX
+#define IN_MIN SCHAR_MIN
+#define IN_MAX SCHAR_MAX
#define SIGNED
+#define SHIFT 8
#include "mixeng_template.h"
#undef SIGNED
#undef IN_MAX
#undef IN_MIN
#undef IN_T
+#undef SHIFT
+/* Unsigned 8 bit */
#define IN_T uint8_t
#define IN_MIN 0
#define IN_MAX UCHAR_MAX
+#define SHIFT 8
#include "mixeng_template.h"
#undef IN_MAX
#undef IN_MIN
#undef IN_T
+#undef SHIFT
+
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+/* Signed 16 bit */
#define IN_T int16_t
#define IN_MIN SHRT_MIN
#define IN_MAX SHRT_MAX
#define SIGNED
+#define SHIFT 16
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap16 (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
#undef SIGNED
#undef IN_MAX
#undef IN_MIN
#undef IN_T
+#undef SHIFT
#define IN_T uint16_t
#define IN_MIN 0
#define IN_MAX USHRT_MAX
+#define SHIFT 16
+#define ENDIAN_CONVERSION natural
+#define ENDIAN_CONVERT(v) (v)
+#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
+#define ENDIAN_CONVERSION swap
+#define ENDIAN_CONVERT(v) bswap16 (v)
#include "mixeng_template.h"
+#undef ENDIAN_CONVERT
+#undef ENDIAN_CONVERSION
#undef IN_MAX
#undef IN_MIN
#undef IN_T
+#undef SHIFT
-t_sample *mixeng_conv[2][2][2] = {
+t_sample *mixeng_conv[2][2][2][2] = {
{
{
- conv_uint8_t_to_mono,
- conv_uint16_t_to_mono
+ {
+ conv_natural_uint8_t_to_mono,
+ conv_natural_uint16_t_to_mono
+ },
+ {
+ conv_natural_uint8_t_to_mono,
+ conv_swap_uint16_t_to_mono
+ }
},
{
- conv_int8_t_to_mono,
- conv_int16_t_to_mono
+ {
+ conv_natural_int8_t_to_mono,
+ conv_natural_int16_t_to_mono
+ },
+ {
+ conv_natural_int8_t_to_mono,
+ conv_swap_int16_t_to_mono
+ }
}
},
{
{
- conv_uint8_t_to_stereo,
- conv_uint16_t_to_stereo
+ {
+ conv_natural_uint8_t_to_stereo,
+ conv_natural_uint16_t_to_stereo
+ },
+ {
+ conv_natural_uint8_t_to_stereo,
+ conv_swap_uint16_t_to_stereo
+ }
},
{
- conv_int8_t_to_stereo,
- conv_int16_t_to_stereo
+ {
+ conv_natural_int8_t_to_stereo,
+ conv_natural_int16_t_to_stereo
+ },
+ {
+ conv_natural_int8_t_to_stereo,
+ conv_swap_int16_t_to_stereo
+ }
}
}
};
-f_sample *mixeng_clip[2][2][2] = {
+f_sample *mixeng_clip[2][2][2][2] = {
{
{
- clip_uint8_t_from_mono,
- clip_uint16_t_from_mono
+ {
+ clip_natural_uint8_t_from_mono,
+ clip_natural_uint16_t_from_mono
+ },
+ {
+ clip_natural_uint8_t_from_mono,
+ clip_swap_uint16_t_from_mono
+ }
},
{
- clip_int8_t_from_mono,
- clip_int16_t_from_mono
+ {
+ clip_natural_int8_t_from_mono,
+ clip_natural_int16_t_from_mono
+ },
+ {
+ clip_natural_int8_t_from_mono,
+ clip_swap_int16_t_from_mono
+ }
}
},
{
{
- clip_uint8_t_from_stereo,
- clip_uint16_t_from_stereo
+ {
+ clip_natural_uint8_t_from_stereo,
+ clip_natural_uint16_t_from_stereo
+ },
+ {
+ clip_natural_uint8_t_from_stereo,
+ clip_swap_uint16_t_from_stereo
+ }
},
{
- clip_int8_t_from_stereo,
- clip_int16_t_from_stereo
+ {
+ clip_natural_int8_t_from_stereo,
+ clip_natural_int16_t_from_stereo
+ },
+ {
+ clip_natural_int8_t_from_stereo,
+ clip_swap_int16_t_from_stereo
+ }
}
}
};
@@ -116,9 +203,9 @@ f_sample *mixeng_clip[2][2][2] = {
* Contributors with a more efficient algorithm.]
*
* This source code is freely redistributable and may be used for
- * any purpose. This copyright notice must be maintained.
- * Lance Norskog And Sundry Contributors are not responsible for
- * the consequences of using this software.
+ * any purpose. This copyright notice must be maintained.
+ * Lance Norskog And Sundry Contributors are not responsible for
+ * the consequences of using this software.
*/
/*
@@ -156,21 +243,13 @@ void *st_rate_start (int inrate, int outrate)
rate_t rate = (rate_t) qemu_mallocz (sizeof (struct ratestuff));
if (!rate) {
- exit (EXIT_FAILURE);
- }
-
- if (inrate == outrate) {
- // exit (EXIT_FAILURE);
- }
-
- if (inrate >= 65535 || outrate >= 65535) {
- // exit (EXIT_FAILURE);
+ return NULL;
}
rate->opos = 0;
/* increment */
- rate->opos_inc = (inrate * ((int64_t) UINT_MAX)) / outrate;
+ rate->opos_inc = ((uint64_t) inrate << 32) / outrate;
rate->ipos = 0;
rate->ilast.l = 0;
@@ -178,78 +257,20 @@ void *st_rate_start (int inrate, int outrate)
return rate;
}
-/*
- * Processed signed long samples from ibuf to obuf.
- * Return number of samples processed.
- */
-void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
- int *isamp, int *osamp)
-{
- rate_t rate = (rate_t) opaque;
- st_sample_t *istart, *iend;
- st_sample_t *ostart, *oend;
- st_sample_t ilast, icur, out;
- int64_t t;
-
- ilast = rate->ilast;
-
- istart = ibuf;
- iend = ibuf + *isamp;
+#define NAME st_rate_flow_mix
+#define OP(a, b) a += b
+#include "rate_template.h"
- ostart = obuf;
- oend = obuf + *osamp;
-
- if (rate->opos_inc == 1ULL << 32) {
- int i, n = *isamp > *osamp ? *osamp : *isamp;
- for (i = 0; i < n; i++) {
- obuf[i].l += ibuf[i].r;
- obuf[i].r += ibuf[i].r;
- }
- *isamp = n;
- *osamp = n;
- return;
- }
-
- while (obuf < oend) {
-
- /* Safety catch to make sure we have input samples. */
- if (ibuf >= iend)
- break;
-
- /* read as many input samples so that ipos > opos */
-
- while (rate->ipos <= (rate->opos >> 32)) {
- ilast = *ibuf++;
- rate->ipos++;
- /* See if we finished the input buffer yet */
- if (ibuf >= iend) goto the_end;
- }
-
- icur = *ibuf;
-
- /* interpolate */
- t = rate->opos & 0xffffffff;
- out.l = (ilast.l * (INT_MAX - t) + icur.l * t) / INT_MAX;
- out.r = (ilast.r * (INT_MAX - t) + icur.r * t) / INT_MAX;
-
- /* output sample & increment position */
-#if 0
- *obuf++ = out;
-#else
- obuf->l += out.l;
- obuf->r += out.r;
- obuf += 1;
-#endif
- rate->opos += rate->opos_inc;
- }
-
-the_end:
- *isamp = ibuf - istart;
- *osamp = obuf - ostart;
- rate->ilast = ilast;
-}
+#define NAME st_rate_flow
+#define OP(a, b) a = b
+#include "rate_template.h"
void st_rate_stop (void *opaque)
{
qemu_free (opaque);
}
+
+void mixeng_clear (st_sample_t *buf, int len)
+{
+ memset (buf, 0, len * sizeof (st_sample_t));
+}
diff --git a/audio/mixeng.h b/audio/mixeng.h
index 699435ea25..9e3bac1744 100644
--- a/audio/mixeng.h
+++ b/audio/mixeng.h
@@ -1,8 +1,8 @@
/*
* QEMU Mixing engine header
- *
- * Copyright (c) 2004 Vassili Karpov (malc)
- *
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -24,16 +24,28 @@
#ifndef QEMU_MIXENG_H
#define QEMU_MIXENG_H
-typedef void (t_sample) (void *dst, const void *src, int samples);
-typedef void (f_sample) (void *dst, const void *src, int samples);
+#ifdef FLOAT_MIXENG
+typedef float real_t;
+typedef struct { int mute; real_t r; real_t l; } volume_t;
+typedef struct { real_t l; real_t r; } st_sample_t;
+#else
+typedef struct { int mute; int64_t r; int64_t l; } volume_t;
typedef struct { int64_t l; int64_t r; } st_sample_t;
+#endif
+
+typedef void (t_sample) (st_sample_t *dst, const void *src,
+ int samples, volume_t *vol);
+typedef void (f_sample) (void *dst, const st_sample_t *src, int samples);
-extern t_sample *mixeng_conv[2][2][2];
-extern f_sample *mixeng_clip[2][2][2];
+extern t_sample *mixeng_conv[2][2][2][2];
+extern f_sample *mixeng_clip[2][2][2][2];
void *st_rate_start (int inrate, int outrate);
void st_rate_flow (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
int *isamp, int *osamp);
+void st_rate_flow_mix (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
+ int *isamp, int *osamp);
void st_rate_stop (void *opaque);
+void mixeng_clear (st_sample_t *buf, int len);
#endif /* mixeng.h */
diff --git a/audio/mixeng_template.h b/audio/mixeng_template.h
index f3b3f654fd..d726441e2e 100644
--- a/audio/mixeng_template.h
+++ b/audio/mixeng_template.h
@@ -1,8 +1,8 @@
/*
* QEMU Mixing engine
- *
- * Copyright (c) 2004 Vassili Karpov (malc)
- *
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -27,85 +27,151 @@
* dec++'ified by Dscho
*/
+#ifndef SIGNED
+#define HALF (IN_MAX >> 1)
+#endif
+
+#ifdef NOVOL
+#define VOL(a, b) a
+#else
+#ifdef FLOAT_MIXENG
+#define VOL(a, b) ((a) * (b))
+#else
+#define VOL(a, b) ((a) * (b)) >> 32
+#endif
+#endif
+
+#define ET glue (ENDIAN_CONVERSION, glue (_, IN_T))
+
+#ifdef FLOAT_MIXENG
+static real_t inline glue (conv_, ET) (IN_T v)
+{
+ IN_T nv = ENDIAN_CONVERT (v);
+
+#ifdef RECIPROCAL
+#ifdef SIGNED
+ return nv * (1.f / (real_t) (IN_MAX - IN_MIN));
+#else
+ return (nv - HALF) * (1.f / (real_t) IN_MAX);
+#endif
+#else /* !RECIPROCAL */
#ifdef SIGNED
-#define HALFT IN_MAX
-#define HALF IN_MAX
+ return nv / (real_t) (IN_MAX - IN_MIN);
#else
-#define HALFT ((IN_MAX)>>1)
-#define HALF HALFT
+ return (nv - HALF) / (real_t) IN_MAX;
#endif
+#endif
+}
-static int64_t inline glue(conv_,IN_T) (IN_T v)
+static IN_T inline glue (clip_, ET) (real_t v)
{
+ if (v >= 0.5) {
+ return IN_MAX;
+ }
+ else if (v < -0.5) {
+ return IN_MIN;
+ }
+
+#ifdef SIGNED
+ return ENDIAN_CONVERT ((IN_T) (v * (IN_MAX - IN_MIN)));
+#else
+ return ENDIAN_CONVERT ((IN_T) ((v * IN_MAX) + HALF));
+#endif
+}
+
+#else /* !FLOAT_MIXENG */
+
+static inline int64_t glue (conv_, ET) (IN_T v)
+{
+ IN_T nv = ENDIAN_CONVERT (v);
#ifdef SIGNED
- return (INT_MAX*(int64_t)v)/HALF;
+ return ((int64_t) nv) << (32 - SHIFT);
#else
- return (INT_MAX*((int64_t)v-HALFT))/HALF;
+ return ((int64_t) nv - HALF) << (32 - SHIFT);
#endif
}
-static IN_T inline glue(clip_,IN_T) (int64_t v)
+static inline IN_T glue (clip_, ET) (int64_t v)
{
- if (v >= INT_MAX)
+ if (v >= 0x7f000000) {
return IN_MAX;
- else if (v < -INT_MAX)
+ }
+ else if (v < -2147483648LL) {
return IN_MIN;
+ }
#ifdef SIGNED
- return (IN_T) (v*HALF/INT_MAX);
+ return ENDIAN_CONVERT ((IN_T) (v >> (32 - SHIFT)));
#else
- return (IN_T) (v+INT_MAX/2)*HALF/INT_MAX;
+ return ENDIAN_CONVERT ((IN_T) ((v >> (32 - SHIFT)) + HALF));
#endif
}
+#endif
-static void glue(glue(conv_,IN_T),_to_stereo) (void *dst, const void *src,
- int samples)
+static void glue (glue (conv_, ET), _to_stereo)
+ (st_sample_t *dst, const void *src, int samples, volume_t *vol)
{
- st_sample_t *out = (st_sample_t *) dst;
+ st_sample_t *out = dst;
IN_T *in = (IN_T *) src;
+#ifndef NOVOL
+ if (vol->mute) {
+ mixeng_clear (dst, samples);
+ return;
+ }
+#else
+ (void) vol;
+#endif
while (samples--) {
- out->l = glue(conv_,IN_T) (*in++);
- out->r = glue(conv_,IN_T) (*in++);
+ out->l = VOL (glue (conv_, ET) (*in++), vol->l);
+ out->r = VOL (glue (conv_, ET) (*in++), vol->r);
out += 1;
}
}
-static void glue(glue(conv_,IN_T),_to_mono) (void *dst, const void *src,
- int samples)
+static void glue (glue (conv_, ET), _to_mono)
+ (st_sample_t *dst, const void *src, int samples, volume_t *vol)
{
- st_sample_t *out = (st_sample_t *) dst;
+ st_sample_t *out = dst;
IN_T *in = (IN_T *) src;
+#ifndef NOVOL
+ if (vol->mute) {
+ mixeng_clear (dst, samples);
+ return;
+ }
+#else
+ (void) vol;
+#endif
while (samples--) {
- out->l = glue(conv_,IN_T) (in[0]);
+ out->l = VOL (glue (conv_, ET) (in[0]), vol->l);
out->r = out->l;
out += 1;
in += 1;
}
}
-static void glue(glue(clip_,IN_T),_from_stereo) (void *dst, const void *src,
- int samples)
+static void glue (glue (clip_, ET), _from_stereo)
+ (void *dst, const st_sample_t *src, int samples)
{
- st_sample_t *in = (st_sample_t *) src;
+ const st_sample_t *in = src;
IN_T *out = (IN_T *) dst;
while (samples--) {
- *out++ = glue(clip_,IN_T) (in->l);
- *out++ = glue(clip_,IN_T) (in->r);
+ *out++ = glue (clip_, ET) (in->l);
+ *out++ = glue (clip_, ET) (in->r);
in += 1;
}
}
-static void glue(glue(clip_,IN_T),_from_mono) (void *dst, const void *src,
- int samples)
+static void glue (glue (clip_, ET), _from_mono)
+ (void *dst, const st_sample_t *src, int samples)
{
- st_sample_t *in = (st_sample_t *) src;
+ const st_sample_t *in = src;
IN_T *out = (IN_T *) dst;
while (samples--) {
- *out++ = glue(clip_,IN_T) (in->l + in->r);
+ *out++ = glue (clip_, ET) (in->l + in->r);
in += 1;
}
}
+#undef ET
#undef HALF
-#undef HALFT
-
+#undef VOL
diff --git a/audio/noaudio.c b/audio/noaudio.c
index a192885a72..e7936cc7b9 100644
--- a/audio/noaudio.c
+++ b/audio/noaudio.c
@@ -1,8 +1,8 @@
/*
- * QEMU NULL audio output driver
- *
- * Copyright (c) 2004 Vassili Karpov (malc)
- *
+ * QEMU Timer based audio emulation
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -23,77 +23,110 @@
*/
#include "vl.h"
-#include "audio/audio_int.h"
+#define AUDIO_CAP "noaudio"
+#include "audio_int.h"
-typedef struct NoVoice {
- HWVoice hw;
+typedef struct NoVoiceOut {
+ HWVoiceOut hw;
int64_t old_ticks;
-} NoVoice;
+} NoVoiceOut;
-#define dolog(...) AUD_log ("noaudio", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+typedef struct NoVoiceIn {
+ HWVoiceIn hw;
+ int64_t old_ticks;
+} NoVoiceIn;
-static void no_hw_run (HWVoice *hw)
+static int no_run_out (HWVoiceOut *hw)
{
- NoVoice *no = (NoVoice *) hw;
- int rpos, live, decr, samples;
- st_sample_t *src;
+ NoVoiceOut *no = (NoVoiceOut *) hw;
+ int live, decr, samples;
int64_t now = qemu_get_clock (vm_clock);
int64_t ticks = now - no->old_ticks;
- int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec;
+ int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
- if (bytes > INT_MAX)
- samples = INT_MAX >> hw->shift;
- else
- samples = bytes >> hw->shift;
+ if (bytes > INT_MAX) {
+ samples = INT_MAX >> hw->info.shift;
+ }
+ else {
+ samples = bytes >> hw->info.shift;
+ }
- live = pcm_hw_get_live (hw);
- if (live <= 0)
- return;
+ live = audio_pcm_hw_get_live_out (&no->hw);
+ if (!live) {
+ return 0;
+ }
no->old_ticks = now;
decr = audio_MIN (live, samples);
- samples = decr;
- rpos = hw->rpos;
- while (samples) {
- int left_till_end_samples = hw->samples - rpos;
- int convert_samples = audio_MIN (samples, left_till_end_samples);
+ hw->rpos = (hw->rpos + decr) % hw->samples;
+ return decr;
+}
- src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
- memset (src, 0, convert_samples * sizeof (st_sample_t));
+static int no_write (SWVoiceOut *sw, void *buf, int len)
+{
+ return audio_pcm_sw_write (sw, buf, len);
+}
- rpos = (rpos + convert_samples) % hw->samples;
- samples -= convert_samples;
- }
+static int no_init_out (HWVoiceOut *hw, int freq,
+ int nchannels, audfmt_e fmt)
+{
+ audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
+ hw->bufsize = 4096;
+ return 0;
+}
- pcm_hw_dec_live (hw, decr);
- hw->rpos = rpos;
+static void no_fini_out (HWVoiceOut *hw)
+{
+ (void) hw;
}
-static int no_hw_write (SWVoice *sw, void *buf, int len)
+static int no_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
- return pcm_hw_write (sw, buf, len);
+ (void) hw;
+ (void) cmd;
+ return 0;
}
-static int no_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int no_init_in (HWVoiceIn *hw, int freq,
+ int nchannels, audfmt_e fmt)
{
- hw->freq = freq;
- hw->nchannels = nchannels;
- hw->fmt = fmt;
+ audio_pcm_init_info (&hw->info, freq, nchannels, fmt, 0);
hw->bufsize = 4096;
return 0;
}
-static void no_hw_fini (HWVoice *hw)
+static void no_fini_in (HWVoiceIn *hw)
{
(void) hw;
}
-static int no_hw_ctl (HWVoice *hw, int cmd, ...)
+static int no_run_in (HWVoiceIn *hw)
+{
+ NoVoiceIn *no = (NoVoiceIn *) hw;
+ int64_t now = qemu_get_clock (vm_clock);
+ int64_t ticks = now - no->old_ticks;
+ int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
+ int live = audio_pcm_hw_get_live_in (hw);
+ int dead = hw->samples - live;
+ int samples;
+
+ bytes = audio_MIN (bytes, INT_MAX);
+ samples = bytes >> hw->info.shift;
+ samples = audio_MIN (samples, dead);
+
+ return samples;
+}
+
+static int no_read (SWVoiceIn *sw, void *buf, int size)
+{
+ int samples = size >> sw->info.shift;
+ int total = sw->hw->total_samples_captured - sw->total_hw_samples_acquired;
+ int to_clear = audio_MIN (samples, total);
+ audio_pcm_info_clear_buf (&sw->info, buf, to_clear);
+ return to_clear;
+}
+
+static int no_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
(void) hw;
(void) cmd;
@@ -107,22 +140,33 @@ static void *no_audio_init (void)
static void no_audio_fini (void *opaque)
{
+ (void) opaque;
}
-struct pcm_ops no_pcm_ops = {
- no_hw_init,
- no_hw_fini,
- no_hw_run,
- no_hw_write,
- no_hw_ctl
+static struct audio_pcm_ops no_pcm_ops = {
+ no_init_out,
+ no_fini_out,
+ no_run_out,
+ no_write,
+ no_ctl_out,
+
+ no_init_in,
+ no_fini_in,
+ no_run_in,
+ no_read,
+ no_ctl_in
};
-struct audio_output_driver no_output_driver = {
- "none",
- no_audio_init,
- no_audio_fini,
- &no_pcm_ops,
- 1,
- 1,
- sizeof (NoVoice)
+struct audio_driver no_audio_driver = {
+ INIT_FIELD (name = ) "none",
+ INIT_FIELD (descr = ) "Timer based audio emulation",
+ INIT_FIELD (options = ) NULL,
+ INIT_FIELD (init = ) no_audio_init,
+ INIT_FIELD (fini = ) no_audio_fini,
+ INIT_FIELD (pcm_ops = ) &no_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) INT_MAX,
+ INIT_FIELD (max_voices_in = ) INT_MAX,
+ INIT_FIELD (voice_size_out = ) sizeof (NoVoiceOut),
+ INIT_FIELD (voice_size_in = ) sizeof (NoVoiceIn)
};
diff --git a/audio/ossaudio.c b/audio/ossaudio.c
index 5246ebb785..ff1a034945 100644
--- a/audio/ossaudio.c
+++ b/audio/ossaudio.c
@@ -1,8 +1,8 @@
/*
- * QEMU OSS audio output driver
- *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- *
+ * QEMU OSS audio driver
+ *
+ * Copyright (c) 2003-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -25,45 +25,42 @@
#include <sys/types.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
-#include <assert.h>
#include "vl.h"
-#include "audio/audio_int.h"
+#define AUDIO_CAP "oss"
+#include "audio_int.h"
-typedef struct OSSVoice {
- HWVoice hw;
+typedef struct OSSVoiceOut {
+ HWVoiceOut hw;
void *pcm_buf;
int fd;
int nfrags;
int fragsize;
int mmapped;
int old_optr;
-} OSSVoice;
-
-#define dolog(...) AUD_log ("oss", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+} OSSVoiceOut;
-#define QC_OSS_FRAGSIZE "QEMU_OSS_FRAGSIZE"
-#define QC_OSS_NFRAGS "QEMU_OSS_NFRAGS"
-#define QC_OSS_MMAP "QEMU_OSS_MMAP"
-#define QC_OSS_DEV "QEMU_OSS_DEV"
-
-#define errstr() strerror (errno)
+typedef struct OSSVoiceIn {
+ HWVoiceIn hw;
+ void *pcm_buf;
+ int fd;
+ int nfrags;
+ int fragsize;
+ int old_optr;
+} OSSVoiceIn;
static struct {
int try_mmap;
int nfrags;
int fragsize;
- const char *dspname;
+ const char *devpath_out;
+ const char *devpath_in;
} conf = {
.try_mmap = 0,
.nfrags = 4,
.fragsize = 4096,
- .dspname = "/dev/dsp"
+ .devpath_out = "/dev/dsp",
+ .devpath_in = "/dev/dsp"
};
struct oss_params {
@@ -74,65 +71,141 @@ struct oss_params {
int fragsize;
};
-static int oss_hw_write (SWVoice *sw, void *buf, int len)
+static void GCC_FMT_ATTR (2, 3) oss_logerr (int err, const char *fmt, ...)
{
- return pcm_hw_write (sw, buf, len);
+ va_list ap;
+
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+
+ va_start (ap, fmt);
+ AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
+ va_end (ap);
}
-static int AUD_to_ossfmt (audfmt_e fmt)
+static void GCC_FMT_ATTR (3, 4) oss_logerr2 (
+ int err,
+ const char *typ,
+ const char *fmt,
+ ...
+ )
+{
+ va_list ap;
+
+ AUD_log (AUDIO_CAP, "Can not initialize %s\n", typ);
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", strerror (err));
+}
+
+static void oss_anal_close (int *fdp)
+{
+ int err = close (*fdp);
+ if (err) {
+ oss_logerr (errno, "Failed to close file(fd=%d)\n", *fdp);
+ }
+ *fdp = -1;
+}
+
+static int oss_write (SWVoiceOut *sw, void *buf, int len)
+{
+ return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int aud_to_ossfmt (audfmt_e fmt)
{
switch (fmt) {
- case AUD_FMT_S8: return AFMT_S8;
- case AUD_FMT_U8: return AFMT_U8;
- case AUD_FMT_S16: return AFMT_S16_LE;
- case AUD_FMT_U16: return AFMT_U16_LE;
+ case AUD_FMT_S8:
+ return AFMT_S8;
+
+ case AUD_FMT_U8:
+ return AFMT_U8;
+
+ case AUD_FMT_S16:
+ return AFMT_S16_LE;
+
+ case AUD_FMT_U16:
+ return AFMT_U16_LE;
+
default:
- dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
- exit (EXIT_FAILURE);
+ dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+ abort ();
+#endif
+ return AFMT_U8;
}
}
-static int oss_to_audfmt (int fmt)
+static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness)
{
- switch (fmt) {
- case AFMT_S8: return AUD_FMT_S8;
- case AFMT_U8: return AUD_FMT_U8;
- case AFMT_S16_LE: return AUD_FMT_S16;
- case AFMT_U16_LE: return AUD_FMT_U16;
+ switch (ossfmt) {
+ case AFMT_S8:
+ *endianness =0;
+ *fmt = AUD_FMT_S8;
+ break;
+
+ case AFMT_U8:
+ *endianness = 0;
+ *fmt = AUD_FMT_U8;
+ break;
+
+ case AFMT_S16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case AFMT_U16_LE:
+ *endianness = 0;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ case AFMT_S16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case AFMT_U16_BE:
+ *endianness = 1;
+ *fmt = AUD_FMT_U16;
+ break;
+
default:
- dolog ("Internal logic error: Unrecognized OSS audio format %d\n"
- "Aborting\n",
- fmt);
- exit (EXIT_FAILURE);
+ dolog ("Unrecognized audio format %d\n", ossfmt);
+ return -1;
}
+
+ return 0;
}
-#ifdef DEBUG_PCM
-static void oss_dump_pcm_info (struct oss_params *req, struct oss_params *obt)
+#ifdef DEBUG_MISMATCHES
+static void oss_dump_info (struct oss_params *req, struct oss_params *obt)
{
dolog ("parameter | requested value | obtained value\n");
dolog ("format | %10d | %10d\n", req->fmt, obt->fmt);
- dolog ("channels | %10d | %10d\n", req->nchannels, obt->nchannels);
+ dolog ("channels | %10d | %10d\n",
+ req->nchannels, obt->nchannels);
dolog ("frequency | %10d | %10d\n", req->freq, obt->freq);
dolog ("nfrags | %10d | %10d\n", req->nfrags, obt->nfrags);
- dolog ("fragsize | %10d | %10d\n", req->fragsize, obt->fragsize);
+ dolog ("fragsize | %10d | %10d\n",
+ req->fragsize, obt->fragsize);
}
#endif
-static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
+static int oss_open (int in, struct oss_params *req,
+ struct oss_params *obt, int *pfd)
{
int fd;
int mmmmssss;
audio_buf_info abinfo;
int fmt, freq, nchannels;
- const char *dspname = conf.dspname;
+ const char *dspname = in ? conf.devpath_in : conf.devpath_out;
+ const char *typ = in ? "ADC" : "DAC";
- fd = open (dspname, O_WRONLY | O_NONBLOCK);
+ fd = open (dspname, (in ? O_RDONLY : O_WRONLY) | O_NONBLOCK);
if (-1 == fd) {
- dolog ("Could not initialize audio hardware. Failed to open `%s':\n"
- "Reason:%s\n",
- dspname,
- errstr ());
+ oss_logerr2 (errno, typ, "Failed to open `%s'\n", dspname);
return -1;
}
@@ -141,52 +214,35 @@ static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
fmt = req->fmt;
if (ioctl (fd, SNDCTL_DSP_SAMPLESIZE, &fmt)) {
- dolog ("Could not initialize audio hardware\n"
- "Failed to set sample size\n"
- "Reason: %s\n",
- errstr ());
+ oss_logerr2 (errno, typ, "Failed to set sample size %d\n", req->fmt);
goto err;
}
if (ioctl (fd, SNDCTL_DSP_CHANNELS, &nchannels)) {
- dolog ("Could not initialize audio hardware\n"
- "Failed to set number of channels\n"
- "Reason: %s\n",
- errstr ());
+ oss_logerr2 (errno, typ, "Failed to set number of channels %d\n",
+ req->nchannels);
goto err;
}
if (ioctl (fd, SNDCTL_DSP_SPEED, &freq)) {
- dolog ("Could not initialize audio hardware\n"
- "Failed to set frequency\n"
- "Reason: %s\n",
- errstr ());
+ oss_logerr2 (errno, typ, "Failed to set frequency %d\n", req->freq);
goto err;
}
if (ioctl (fd, SNDCTL_DSP_NONBLOCK)) {
- dolog ("Could not initialize audio hardware\n"
- "Failed to set non-blocking mode\n"
- "Reason: %s\n",
- errstr ());
+ oss_logerr2 (errno, typ, "Failed to set non-blocking mode\n");
goto err;
}
mmmmssss = (req->nfrags << 16) | lsbindex (req->fragsize);
if (ioctl (fd, SNDCTL_DSP_SETFRAGMENT, &mmmmssss)) {
- dolog ("Could not initialize audio hardware\n"
- "Failed to set buffer length (%d, %d)\n"
- "Reason:%s\n",
- conf.nfrags, conf.fragsize,
- errstr ());
+ oss_logerr2 (errno, typ, "Failed to set buffer length (%d, %d)\n",
+ req->nfrags, req->fragsize);
goto err;
}
- if (ioctl (fd, SNDCTL_DSP_GETOSPACE, &abinfo)) {
- dolog ("Could not initialize audio hardware\n"
- "Failed to get buffer length\n"
- "Reason:%s\n",
- errstr ());
+ if (ioctl (fd, in ? SNDCTL_DSP_GETISPACE : SNDCTL_DSP_GETOSPACE, &abinfo)) {
+ oss_logerr2 (errno, typ, "Failed to get buffer length\n");
goto err;
}
@@ -202,25 +258,25 @@ static int oss_open (struct oss_params *req, struct oss_params *obt, int *pfd)
(req->freq != obt->freq) ||
(req->fragsize != obt->fragsize) ||
(req->nfrags != obt->nfrags)) {
-#ifdef DEBUG_PCM
+#ifdef DEBUG_MISMATCHES
dolog ("Audio parameters mismatch\n");
- oss_dump_pcm_info (req, obt);
+ oss_dump_info (req, obt);
#endif
}
-#ifdef DEBUG_PCM
- oss_dump_pcm_info (req, obt);
+#ifdef DEBUG
+ oss_dump_info (req, obt);
#endif
return 0;
-err:
- close (fd);
+ err:
+ oss_anal_close (&fd);
return -1;
}
-static void oss_hw_run (HWVoice *hw)
+static int oss_run_out (HWVoiceOut *hw)
{
- OSSVoice *oss = (OSSVoice *) hw;
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
int err, rpos, live, decr;
int samples;
uint8_t *dst;
@@ -228,23 +284,25 @@ static void oss_hw_run (HWVoice *hw)
struct audio_buf_info abinfo;
struct count_info cntinfo;
- live = pcm_hw_get_live (hw);
- if (live <= 0)
- return;
+ live = audio_pcm_hw_get_live_out (hw);
+ if (!live) {
+ return 0;
+ }
if (oss->mmapped) {
int bytes;
err = ioctl (oss->fd, SNDCTL_DSP_GETOPTR, &cntinfo);
if (err < 0) {
- dolog ("SNDCTL_DSP_GETOPTR failed\nReason: %s\n", errstr ());
- return;
+ oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
+ return 0;
}
if (cntinfo.ptr == oss->old_optr) {
- if (abs (hw->samples - live) < 64)
- dolog ("overrun\n");
- return;
+ if (abs (hw->samples - live) < 64) {
+ dolog ("warning: overrun\n");
+ }
+ return 0;
}
if (cntinfo.ptr > oss->old_optr) {
@@ -254,18 +312,25 @@ static void oss_hw_run (HWVoice *hw)
bytes = hw->bufsize + cntinfo.ptr - oss->old_optr;
}
- decr = audio_MIN (bytes >> hw->shift, live);
+ decr = audio_MIN (bytes >> hw->info.shift, live);
}
else {
err = ioctl (oss->fd, SNDCTL_DSP_GETOSPACE, &abinfo);
if (err < 0) {
- dolog ("SNDCTL_DSP_GETOSPACE failed\nReason: %s\n", errstr ());
- return;
+ oss_logerr (errno, "SNDCTL_DSP_GETOPTR failed\n");
+ return 0;
+ }
+
+ if (abinfo.bytes < 0 || abinfo.bytes > hw->bufsize) {
+ ldebug ("warning: invalid available size, size=%d bufsize=%d\n",
+ abinfo.bytes, hw->bufsize);
+ return 0;
}
- decr = audio_MIN (abinfo.bytes >> hw->shift, live);
- if (decr <= 0)
- return;
+ decr = audio_MIN (abinfo.bytes >> hw->info.shift, live);
+ if (!decr) {
+ return 0;
+ }
}
samples = decr;
@@ -274,33 +339,41 @@ static void oss_hw_run (HWVoice *hw)
int left_till_end_samples = hw->samples - rpos;
int convert_samples = audio_MIN (samples, left_till_end_samples);
- src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
- dst = advance (oss->pcm_buf, rpos << hw->shift);
+ src = hw->mix_buf + rpos;
+ dst = advance (oss->pcm_buf, rpos << hw->info.shift);
hw->clip (dst, src, convert_samples);
if (!oss->mmapped) {
int written;
- written = write (oss->fd, dst, convert_samples << hw->shift);
+ written = write (oss->fd, dst, convert_samples << hw->info.shift);
/* XXX: follow errno recommendations ? */
if (written == -1) {
- dolog ("Failed to write audio\nReason: %s\n", errstr ());
+ oss_logerr (
+ errno,
+ "Failed to write %d bytes of audio data from %p\n",
+ convert_samples << hw->info.shift,
+ dst
+ );
continue;
}
- if (written != convert_samples << hw->shift) {
- int wsamples = written >> hw->shift;
- int wbytes = wsamples << hw->shift;
+ if (written != convert_samples << hw->info.shift) {
+ int wsamples = written >> hw->info.shift;
+ int wbytes = wsamples << hw->info.shift;
if (wbytes != written) {
- dolog ("Unaligned write %d, %d\n", wbytes, written);
+ dolog ("warning: misaligned write %d (requested %d), "
+ "alignment %d\n",
+ wbytes, written, hw->info.align + 1);
}
- memset (src, 0, wbytes);
- decr -= samples;
+ mixeng_clear (src, wsamples);
+ decr -= wsamples;
rpos = (rpos + wsamples) % hw->samples;
break;
}
}
- memset (src, 0, convert_samples * sizeof (st_sample_t));
+
+ mixeng_clear (src, convert_samples);
rpos = (rpos + convert_samples) % hw->samples;
samples -= convert_samples;
@@ -309,28 +382,24 @@ static void oss_hw_run (HWVoice *hw)
oss->old_optr = cntinfo.ptr;
}
- pcm_hw_dec_live (hw, decr);
hw->rpos = rpos;
+ return decr;
}
-static void oss_hw_fini (HWVoice *hw)
+static void oss_fini_out (HWVoiceOut *hw)
{
int err;
- OSSVoice *oss = (OSSVoice *) hw;
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- ldebug ("oss_hw_fini\n");
- err = close (oss->fd);
- if (err) {
- dolog ("Failed to close OSS descriptor\nReason: %s\n", errstr ());
- }
- oss->fd = -1;
+ ldebug ("oss_fini\n");
+ oss_anal_close (&oss->fd);
if (oss->pcm_buf) {
if (oss->mmapped) {
err = munmap (oss->pcm_buf, hw->bufsize);
if (err) {
- dolog ("Failed to unmap OSS buffer\nReason: %s\n",
- errstr ());
+ oss_logerr (errno, "Failed to unmap buffer %p, size %d\n",
+ oss->pcm_buf, hw->bufsize);
}
}
else {
@@ -340,25 +409,38 @@ static void oss_hw_fini (HWVoice *hw)
}
}
-static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int oss_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
{
- OSSVoice *oss = (OSSVoice *) hw;
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
struct oss_params req, obt;
+ int endianness;
+ int err;
+ int fd;
+ audfmt_e effective_fmt;
- assert (!oss->fd);
- req.fmt = AUD_to_ossfmt (fmt);
+ req.fmt = aud_to_ossfmt (fmt);
req.freq = freq;
req.nchannels = nchannels;
req.fragsize = conf.fragsize;
req.nfrags = conf.nfrags;
- if (oss_open (&req, &obt, &oss->fd))
+ if (oss_open (0, &req, &obt, &fd)) {
return -1;
+ }
- hw->freq = obt.freq;
- hw->fmt = oss_to_audfmt (obt.fmt);
- hw->nchannels = obt.nchannels;
+ err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+ if (err) {
+ oss_anal_close (&fd);
+ return -1;
+ }
+ audio_pcm_init_info (
+ &hw->info,
+ obt.freq,
+ obt.nchannels,
+ effective_fmt,
+ audio_need_to_swap_endian (endianness)
+ );
oss->nfrags = obt.nfrags;
oss->fragsize = obt.fragsize;
hw->bufsize = obt.nfrags * obt.fragsize;
@@ -366,22 +448,23 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
oss->mmapped = 0;
if (conf.try_mmap) {
oss->pcm_buf = mmap (0, hw->bufsize, PROT_READ | PROT_WRITE,
- MAP_SHARED, oss->fd, 0);
+ MAP_SHARED, fd, 0);
if (oss->pcm_buf == MAP_FAILED) {
- dolog ("Failed to mmap OSS device\nReason: %s\n",
- errstr ());
+ oss_logerr (errno, "Failed to map %d bytes of DAC\n",
+ hw->bufsize);
} else {
int err;
int trig = 0;
- if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
- dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
- errstr ());
+ if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+ oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
}
else {
trig = PCM_ENABLE_OUTPUT;
- if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
- dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
- "Reason: %s\n", errstr ());
+ if (ioctl (fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
+ oss_logerr (
+ errno,
+ "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+ );
}
else {
oss->mmapped = 1;
@@ -391,8 +474,8 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
if (!oss->mmapped) {
err = munmap (oss->pcm_buf, hw->bufsize);
if (err) {
- dolog ("Failed to unmap OSS device\nReason: %s\n",
- errstr ());
+ oss_logerr (errno, "Failed to unmap buffer %p size %d\n",
+ oss->pcm_buf, hw->bufsize);
}
}
}
@@ -401,31 +484,34 @@ static int oss_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
if (!oss->mmapped) {
oss->pcm_buf = qemu_mallocz (hw->bufsize);
if (!oss->pcm_buf) {
- close (oss->fd);
- oss->fd = -1;
+ oss_anal_close (&fd);
return -1;
}
}
+ oss->fd = fd;
return 0;
}
-static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
+static int oss_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
int trig;
- OSSVoice *oss = (OSSVoice *) hw;
+ OSSVoiceOut *oss = (OSSVoiceOut *) hw;
- if (!oss->mmapped)
+ if (!oss->mmapped) {
return 0;
+ }
switch (cmd) {
case VOICE_ENABLE:
ldebug ("enabling voice\n");
- pcm_hw_clear (hw, oss->pcm_buf, hw->samples);
+ audio_pcm_info_clear_buf (&hw->info, oss->pcm_buf, hw->samples);
trig = PCM_ENABLE_OUTPUT;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
- dolog ("SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
- "Reason: %s\n", errstr ());
+ oss_logerr (
+ errno,
+ "SNDCTL_DSP_SETTRIGGER PCM_ENABLE_OUTPUT failed\n"
+ );
return -1;
}
break;
@@ -434,8 +520,7 @@ static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
ldebug ("disabling voice\n");
trig = 0;
if (ioctl (oss->fd, SNDCTL_DSP_SETTRIGGER, &trig) < 0) {
- dolog ("SNDCTL_DSP_SETTRIGGER 0 failed\nReason: %s\n",
- errstr ());
+ oss_logerr (errno, "SNDCTL_DSP_SETTRIGGER 0 failed\n");
return -1;
}
break;
@@ -443,33 +528,194 @@ static int oss_hw_ctl (HWVoice *hw, int cmd, ...)
return 0;
}
+static int oss_init_in (HWVoiceIn *hw,
+ int freq, int nchannels, audfmt_e fmt)
+{
+ OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+ struct oss_params req, obt;
+ int endianness;
+ int err;
+ int fd;
+ audfmt_e effective_fmt;
+
+ req.fmt = aud_to_ossfmt (fmt);
+ req.freq = freq;
+ req.nchannels = nchannels;
+ req.fragsize = conf.fragsize;
+ req.nfrags = conf.nfrags;
+ if (oss_open (1, &req, &obt, &fd)) {
+ return -1;
+ }
+
+ err = oss_to_audfmt (obt.fmt, &effective_fmt, &endianness);
+ if (err) {
+ oss_anal_close (&fd);
+ return -1;
+ }
+
+ audio_pcm_init_info (
+ &hw->info,
+ obt.freq,
+ obt.nchannels,
+ effective_fmt,
+ audio_need_to_swap_endian (endianness)
+ );
+ oss->nfrags = obt.nfrags;
+ oss->fragsize = obt.fragsize;
+ hw->bufsize = obt.nfrags * obt.fragsize;
+ oss->pcm_buf = qemu_mallocz (hw->bufsize);
+ if (!oss->pcm_buf) {
+ oss_anal_close (&fd);
+ return -1;
+ }
+
+ oss->fd = fd;
+ return 0;
+}
+
+static void oss_fini_in (HWVoiceIn *hw)
+{
+ OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+
+ oss_anal_close (&oss->fd);
+
+ if (oss->pcm_buf) {
+ qemu_free (oss->pcm_buf);
+ oss->pcm_buf = NULL;
+ }
+}
+
+static int oss_run_in (HWVoiceIn *hw)
+{
+ OSSVoiceIn *oss = (OSSVoiceIn *) hw;
+ int hwshift = hw->info.shift;
+ int i;
+ int live = audio_pcm_hw_get_live_in (hw);
+ int dead = hw->samples - live;
+ size_t read_samples = 0;
+ struct {
+ int add;
+ int len;
+ } bufs[2] = {
+ { hw->wpos, 0 },
+ { 0, 0 }
+ };
+
+ if (!dead) {
+ return 0;
+ }
+
+ if (hw->wpos + dead > hw->samples) {
+ bufs[0].len = (hw->samples - hw->wpos) << hwshift;
+ bufs[1].len = (dead - (hw->samples - hw->wpos)) << hwshift;
+ }
+ else {
+ bufs[0].len = dead << hwshift;
+ }
+
+
+ for (i = 0; i < 2; ++i) {
+ ssize_t nread;
+
+ if (bufs[i].len) {
+ void *p = advance (oss->pcm_buf, bufs[i].add << hwshift);
+ nread = read (oss->fd, p, bufs[i].len);
+
+ if (nread > 0) {
+ if (nread & hw->info.align) {
+ dolog ("warning: misaligned read %d (requested %d), "
+ "alignment %d\n", nread, bufs[i].add << hwshift,
+ hw->info.align + 1);
+ }
+ read_samples += nread >> hwshift;
+ hw->conv (hw->conv_buf + bufs[i].add, p, nread >> hwshift,
+ &nominal_volume);
+ }
+
+ if (bufs[i].len - nread) {
+ if (nread == -1) {
+ switch (errno) {
+ case EINTR:
+ case EAGAIN:
+ break;
+ default:
+ oss_logerr (
+ errno,
+ "Failed to read %d bytes of audio (to %p)\n",
+ bufs[i].len, p
+ );
+ break;
+ }
+ }
+ break;
+ }
+ }
+ }
+
+ hw->wpos = (hw->wpos + read_samples) % hw->samples;
+ return read_samples;
+}
+
+static int oss_read (SWVoiceIn *sw, void *buf, int size)
+{
+ return audio_pcm_sw_read (sw, buf, size);
+}
+
+static int oss_ctl_in (HWVoiceIn *hw, int cmd, ...)
+{
+ (void) hw;
+ (void) cmd;
+ return 0;
+}
+
static void *oss_audio_init (void)
{
- conf.fragsize = audio_get_conf_int (QC_OSS_FRAGSIZE, conf.fragsize);
- conf.nfrags = audio_get_conf_int (QC_OSS_NFRAGS, conf.nfrags);
- conf.try_mmap = audio_get_conf_int (QC_OSS_MMAP, conf.try_mmap);
- conf.dspname = audio_get_conf_str (QC_OSS_DEV, conf.dspname);
return &conf;
}
static void oss_audio_fini (void *opaque)
{
+ (void) opaque;
}
-struct pcm_ops oss_pcm_ops = {
- oss_hw_init,
- oss_hw_fini,
- oss_hw_run,
- oss_hw_write,
- oss_hw_ctl
+static struct audio_option oss_options[] = {
+ {"FRAGSIZE", AUD_OPT_INT, &conf.fragsize,
+ "Fragment size in bytes", NULL, 0},
+ {"NFRAGS", AUD_OPT_INT, &conf.nfrags,
+ "Number of fragments", NULL, 0},
+ {"MMAP", AUD_OPT_BOOL, &conf.try_mmap,
+ "Try using memory mapped access", NULL, 0},
+ {"DAC_DEV", AUD_OPT_STR, &conf.devpath_out,
+ "Path to DAC device", NULL, 0},
+ {"ADC_DEV", AUD_OPT_STR, &conf.devpath_in,
+ "Path to ADC device", NULL, 0},
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops oss_pcm_ops = {
+ oss_init_out,
+ oss_fini_out,
+ oss_run_out,
+ oss_write,
+ oss_ctl_out,
+
+ oss_init_in,
+ oss_fini_in,
+ oss_run_in,
+ oss_read,
+ oss_ctl_in
};
-struct audio_output_driver oss_output_driver = {
- "oss",
- oss_audio_init,
- oss_audio_fini,
- &oss_pcm_ops,
- 1,
- INT_MAX,
- sizeof (OSSVoice)
+struct audio_driver oss_audio_driver = {
+ INIT_FIELD (name = ) "oss",
+ INIT_FIELD (descr = ) "OSS http://www.opensound.com",
+ INIT_FIELD (options = ) oss_options,
+ INIT_FIELD (init = ) oss_audio_init,
+ INIT_FIELD (fini = ) oss_audio_fini,
+ INIT_FIELD (pcm_ops = ) &oss_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) INT_MAX,
+ INIT_FIELD (max_voices_in = ) INT_MAX,
+ INIT_FIELD (voice_size_out = ) sizeof (OSSVoiceOut),
+ INIT_FIELD (voice_size_in = ) sizeof (OSSVoiceIn)
};
diff --git a/audio/rate_template.h b/audio/rate_template.h
new file mode 100644
index 0000000000..5cc95c829a
--- /dev/null
+++ b/audio/rate_template.h
@@ -0,0 +1,111 @@
+/*
+ * QEMU Mixing engine
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ * Copyright (c) 1998 Fabrice Bellard
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/*
+ * Processed signed long samples from ibuf to obuf.
+ * Return number of samples processed.
+ */
+void NAME (void *opaque, st_sample_t *ibuf, st_sample_t *obuf,
+ int *isamp, int *osamp)
+{
+ rate_t rate = (rate_t) opaque;
+ st_sample_t *istart, *iend;
+ st_sample_t *ostart, *oend;
+ st_sample_t ilast, icur, out;
+#ifdef FLOAT_MIXENG
+ real_t t;
+#else
+ int64_t t;
+#endif
+
+ ilast = rate->ilast;
+
+ istart = ibuf;
+ iend = ibuf + *isamp;
+
+ ostart = obuf;
+ oend = obuf + *osamp;
+
+ if (rate->opos_inc == (1ULL + UINT_MAX)) {
+ int i, n = *isamp > *osamp ? *osamp : *isamp;
+ for (i = 0; i < n; i++) {
+ OP (obuf[i].l, ibuf[i].r);
+ OP (obuf[i].r, ibuf[i].r);
+ }
+ *isamp = n;
+ *osamp = n;
+ return;
+ }
+
+ while (obuf < oend) {
+
+ /* Safety catch to make sure we have input samples. */
+ if (ibuf >= iend) {
+ break;
+ }
+
+ /* read as many input samples so that ipos > opos */
+
+ while (rate->ipos <= (rate->opos >> 32)) {
+ ilast = *ibuf++;
+ rate->ipos++;
+ /* See if we finished the input buffer yet */
+ if (ibuf >= iend) {
+ goto the_end;
+ }
+ }
+
+ icur = *ibuf;
+
+ /* interpolate */
+#ifdef FLOAT_MIXENG
+#ifdef RECIPROCAL
+ t = (rate->opos & UINT_MAX) * (1.f / UINT_MAX);
+#else
+ t = (rate->opos & UINT_MAX) / (real_t) UINT_MAX;
+#endif
+ out.l = (ilast.l * (1.0 - t)) + icur.l * t;
+ out.r = (ilast.r * (1.0 - t)) + icur.r * t;
+#else
+ t = rate->opos & 0xffffffff;
+ out.l = (ilast.l * ((int64_t) UINT_MAX - t) + icur.l * t) >> 32;
+ out.r = (ilast.r * ((int64_t) UINT_MAX - t) + icur.r * t) >> 32;
+#endif
+
+ /* output sample & increment position */
+ OP (obuf->l, out.l);
+ OP (obuf->r, out.r);
+ obuf += 1;
+ rate->opos += rate->opos_inc;
+ }
+
+the_end:
+ *isamp = ibuf - istart;
+ *osamp = obuf - ostart;
+ rate->ilast = ilast;
+}
+
+#undef NAME
+#undef OP
diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c
index 978686a071..673e2a1125 100644
--- a/audio/sdlaudio.c
+++ b/audio/sdlaudio.c
@@ -1,8 +1,8 @@
/*
- * QEMU SDL audio output driver
- *
- * Copyright (c) 2004 Vassili Karpov (malc)
- *
+ * QEMU SDL audio driver
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -25,22 +25,15 @@
#include <SDL_thread.h>
#include "vl.h"
-#include "audio/audio_int.h"
-
-typedef struct SDLVoice {
- HWVoice hw;
-} SDLVoice;
-
-#define dolog(...) AUD_log ("sdl", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+#define AUDIO_CAP "sdl"
+#include "audio_int.h"
-#define QC_SDL_SAMPLES "QEMU_SDL_SAMPLES"
-
-#define errstr() SDL_GetError ()
+typedef struct SDLVoiceOut {
+ HWVoiceOut hw;
+ int live;
+ int rpos;
+ int decr;
+} SDLVoiceOut;
static struct {
int nb_samples;
@@ -56,91 +49,129 @@ struct SDLAudioState {
} glob_sdl;
typedef struct SDLAudioState SDLAudioState;
-static void sdl_hw_run (HWVoice *hw)
+static void GCC_FMT_ATTR (1, 2) sdl_logerr (const char *fmt, ...)
{
- (void) hw;
+ va_list ap;
+
+ va_start (ap, fmt);
+ AUD_vlog (AUDIO_CAP, fmt, ap);
+ va_end (ap);
+
+ AUD_log (AUDIO_CAP, "Reason: %s\n", SDL_GetError ());
}
-static int sdl_lock (SDLAudioState *s)
+static int sdl_lock (SDLAudioState *s, const char *forfn)
{
if (SDL_LockMutex (s->mutex)) {
- dolog ("SDL_LockMutex failed\nReason: %s\n", errstr ());
+ sdl_logerr ("SDL_LockMutex for %s failed\n", forfn);
return -1;
}
return 0;
}
-static int sdl_unlock (SDLAudioState *s)
+static int sdl_unlock (SDLAudioState *s, const char *forfn)
{
if (SDL_UnlockMutex (s->mutex)) {
- dolog ("SDL_UnlockMutex failed\nReason: %s\n", errstr ());
+ sdl_logerr ("SDL_UnlockMutex for %s failed\n", forfn);
return -1;
}
return 0;
}
-static int sdl_post (SDLAudioState *s)
+static int sdl_post (SDLAudioState *s, const char *forfn)
{
if (SDL_SemPost (s->sem)) {
- dolog ("SDL_SemPost failed\nReason: %s\n", errstr ());
+ sdl_logerr ("SDL_SemPost for %s failed\n", forfn);
return -1;
}
return 0;
}
-static int sdl_wait (SDLAudioState *s)
+static int sdl_wait (SDLAudioState *s, const char *forfn)
{
if (SDL_SemWait (s->sem)) {
- dolog ("SDL_SemWait failed\nReason: %s\n", errstr ());
+ sdl_logerr ("SDL_SemWait for %s failed\n", forfn);
return -1;
}
return 0;
}
-static int sdl_unlock_and_post (SDLAudioState *s)
+static int sdl_unlock_and_post (SDLAudioState *s, const char *forfn)
{
- if (sdl_unlock (s))
+ if (sdl_unlock (s, forfn)) {
return -1;
+ }
- return sdl_post (s);
-}
-
-static int sdl_hw_write (SWVoice *sw, void *buf, int len)
-{
- int ret;
- SDLAudioState *s = &glob_sdl;
- sdl_lock (s);
- ret = pcm_hw_write (sw, buf, len);
- sdl_unlock_and_post (s);
- return ret;
+ return sdl_post (s, forfn);
}
-static int AUD_to_sdlfmt (audfmt_e fmt, int *shift)
+static int aud_to_sdlfmt (audfmt_e fmt, int *shift)
{
- *shift = 0;
switch (fmt) {
- case AUD_FMT_S8: return AUDIO_S8;
- case AUD_FMT_U8: return AUDIO_U8;
- case AUD_FMT_S16: *shift = 1; return AUDIO_S16LSB;
- case AUD_FMT_U16: *shift = 1; return AUDIO_U16LSB;
+ case AUD_FMT_S8:
+ *shift = 0;
+ return AUDIO_S8;
+
+ case AUD_FMT_U8:
+ *shift = 0;
+ return AUDIO_U8;
+
+ case AUD_FMT_S16:
+ *shift = 1;
+ return AUDIO_S16LSB;
+
+ case AUD_FMT_U16:
+ *shift = 1;
+ return AUDIO_U16LSB;
+
default:
- dolog ("Internal logic error: Bad audio format %d\nAborting\n", fmt);
- exit (EXIT_FAILURE);
+ dolog ("Internal logic error: Bad audio format %d\n", fmt);
+#ifdef DEBUG_AUDIO
+ abort ();
+#endif
+ return AUDIO_U8;
}
}
-static int sdl_to_audfmt (int fmt)
+static int sdl_to_audfmt (int sdlfmt, audfmt_e *fmt, int *endianess)
{
- switch (fmt) {
- case AUDIO_S8: return AUD_FMT_S8;
- case AUDIO_U8: return AUD_FMT_U8;
- case AUDIO_S16LSB: return AUD_FMT_S16;
- case AUDIO_U16LSB: return AUD_FMT_U16;
+ switch (sdlfmt) {
+ case AUDIO_S8:
+ *endianess = 0;
+ *fmt = AUD_FMT_S8;
+ break;
+
+ case AUDIO_U8:
+ *endianess = 0;
+ *fmt = AUD_FMT_U8;
+ break;
+
+ case AUDIO_S16LSB:
+ *endianess = 0;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case AUDIO_U16LSB:
+ *endianess = 0;
+ *fmt = AUD_FMT_U16;
+ break;
+
+ case AUDIO_S16MSB:
+ *endianess = 1;
+ *fmt = AUD_FMT_S16;
+ break;
+
+ case AUDIO_U16MSB:
+ *endianess = 1;
+ *fmt = AUD_FMT_U16;
+ break;
+
default:
- dolog ("Internal logic error: Unrecognized SDL audio format %d\n"
- "Aborting\n", fmt);
- exit (EXIT_FAILURE);
+ dolog ("Unrecognized SDL audio format %d\n", sdlfmt);
+ return -1;
}
+
+ return 0;
}
static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
@@ -149,7 +180,7 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
status = SDL_OpenAudio (req, obt);
if (status) {
- dolog ("SDL_OpenAudio failed\nReason: %s\n", errstr ());
+ sdl_logerr ("SDL_OpenAudio failed\n");
}
return status;
}
@@ -157,9 +188,9 @@ static int sdl_open (SDL_AudioSpec *req, SDL_AudioSpec *obt)
static void sdl_close (SDLAudioState *s)
{
if (s->initialized) {
- sdl_lock (s);
+ sdl_lock (s, "sdl_close");
s->exit = 1;
- sdl_unlock_and_post (s);
+ sdl_unlock_and_post (s, "sdl_close");
SDL_PauseAudio (1);
SDL_CloseAudio ();
s->initialized = 0;
@@ -168,31 +199,40 @@ static void sdl_close (SDLAudioState *s)
static void sdl_callback (void *opaque, Uint8 *buf, int len)
{
- SDLVoice *sdl = opaque;
+ SDLVoiceOut *sdl = opaque;
SDLAudioState *s = &glob_sdl;
- HWVoice *hw = &sdl->hw;
- int samples = len >> hw->shift;
+ HWVoiceOut *hw = &sdl->hw;
+ int samples = len >> hw->info.shift;
if (s->exit) {
return;
}
while (samples) {
- int to_mix, live, decr;
+ int to_mix, decr;
/* dolog ("in callback samples=%d\n", samples); */
- sdl_wait (s);
+ sdl_wait (s, "sdl_callback");
if (s->exit) {
return;
}
- sdl_lock (s);
- live = pcm_hw_get_live (hw);
- if (live <= 0)
+ if (sdl_lock (s, "sdl_callback")) {
+ return;
+ }
+
+ if (audio_bug (AUDIO_FUNC, sdl->live < 0 || sdl->live > hw->samples)) {
+ dolog ("sdl->live=%d hw->samples=%d\n",
+ sdl->live, hw->samples);
+ return;
+ }
+
+ if (!sdl->live) {
goto again;
+ }
/* dolog ("in callback live=%d\n", live); */
- to_mix = audio_MIN (samples, live);
+ to_mix = audio_MIN (samples, sdl->live);
decr = to_mix;
while (to_mix) {
int chunk = audio_MIN (to_mix, hw->samples - hw->rpos);
@@ -200,44 +240,86 @@ static void sdl_callback (void *opaque, Uint8 *buf, int len)
/* dolog ("in callback to_mix %d, chunk %d\n", to_mix, chunk); */
hw->clip (buf, src, chunk);
- memset (src, 0, chunk * sizeof (st_sample_t));
- hw->rpos = (hw->rpos + chunk) % hw->samples;
+ mixeng_clear (src, chunk);
+ sdl->rpos = (sdl->rpos + chunk) % hw->samples;
to_mix -= chunk;
- buf += chunk << hw->shift;
+ buf += chunk << hw->info.shift;
}
samples -= decr;
- pcm_hw_dec_live (hw, decr);
+ sdl->live -= decr;
+ sdl->decr += decr;
again:
- sdl_unlock (s);
+ if (sdl_unlock (s, "sdl_callback")) {
+ return;
+ }
}
/* dolog ("done len=%d\n", len); */
}
-static void sdl_hw_fini (HWVoice *hw)
+static int sdl_write_out (SWVoiceOut *sw, void *buf, int len)
{
- ldebug ("sdl_hw_fini %d fixed=%d\n",
- glob_sdl.initialized, audio_state.fixed_format);
+ return audio_pcm_sw_write (sw, buf, len);
+}
+
+static int sdl_run_out (HWVoiceOut *hw)
+{
+ int decr, live;
+ SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
+ SDLAudioState *s = &glob_sdl;
+
+ if (sdl_lock (s, "sdl_callback")) {
+ return 0;
+ }
+
+ live = audio_pcm_hw_get_live_out (hw);
+
+ if (sdl->decr > live) {
+ ldebug ("sdl->decr %d live %d sdl->live %d\n",
+ sdl->decr,
+ live,
+ sdl->live);
+ }
+
+ decr = audio_MIN (sdl->decr, live);
+ sdl->decr -= decr;
+
+ sdl->live = live - decr;
+ hw->rpos = sdl->rpos;
+
+ if (sdl->live > 0) {
+ sdl_unlock_and_post (s, "sdl_callback");
+ }
+ else {
+ sdl_unlock (s, "sdl_callback");
+ }
+ return decr;
+}
+
+static void sdl_fini_out (HWVoiceOut *hw)
+{
+ (void) hw;
+
sdl_close (&glob_sdl);
}
-static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int sdl_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
{
- SDLVoice *sdl = (SDLVoice *) hw;
+ SDLVoiceOut *sdl = (SDLVoiceOut *) hw;
SDLAudioState *s = &glob_sdl;
SDL_AudioSpec req, obt;
int shift;
-
- ldebug ("sdl_hw_init %d freq=%d fixed=%d\n",
- s->initialized, freq, audio_state.fixed_format);
+ int endianess;
+ int err;
+ audfmt_e effective_fmt;
if (nchannels != 2) {
- dolog ("Bogus channel count %d\n", nchannels);
+ dolog ("Can not init DAC. Bogus channel count %d\n", nchannels);
return -1;
}
req.freq = freq;
- req.format = AUD_to_sdlfmt (fmt, &shift);
+ req.format = aud_to_sdlfmt (fmt, &shift);
req.channels = nchannels;
req.samples = conf.nb_samples;
shift <<= nchannels == 2;
@@ -245,12 +327,23 @@ static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
req.callback = sdl_callback;
req.userdata = sdl;
- if (sdl_open (&req, &obt))
+ if (sdl_open (&req, &obt)) {
+ return -1;
+ }
+
+ err = sdl_to_audfmt (obt.format, &effective_fmt, &endianess);
+ if (err) {
+ sdl_close (s);
return -1;
+ }
- hw->freq = obt.freq;
- hw->fmt = sdl_to_audfmt (obt.format);
- hw->nchannels = obt.channels;
+ audio_pcm_init_info (
+ &hw->info,
+ obt.freq,
+ obt.channels,
+ effective_fmt,
+ audio_need_to_swap_endian (endianess)
+ );
hw->bufsize = obt.samples << shift;
s->initialized = 1;
@@ -259,7 +352,7 @@ static int sdl_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
return 0;
}
-static int sdl_hw_ctl (HWVoice *hw, int cmd, ...)
+static int sdl_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
(void) hw;
@@ -278,24 +371,22 @@ static int sdl_hw_ctl (HWVoice *hw, int cmd, ...)
static void *sdl_audio_init (void)
{
SDLAudioState *s = &glob_sdl;
- conf.nb_samples = audio_get_conf_int (QC_SDL_SAMPLES, conf.nb_samples);
if (SDL_InitSubSystem (SDL_INIT_AUDIO)) {
- dolog ("SDL failed to initialize audio subsystem\nReason: %s\n",
- errstr ());
+ sdl_logerr ("SDL failed to initialize audio subsystem\n");
return NULL;
}
s->mutex = SDL_CreateMutex ();
if (!s->mutex) {
- dolog ("Failed to create SDL mutex\nReason: %s\n", errstr ());
+ sdl_logerr ("Failed to create SDL mutex\n");
SDL_QuitSubSystem (SDL_INIT_AUDIO);
return NULL;
}
s->sem = SDL_CreateSemaphore (0);
if (!s->sem) {
- dolog ("Failed to create SDL semaphore\nReason: %s\n", errstr ());
+ sdl_logerr ("Failed to create SDL semaphore\n");
SDL_DestroyMutex (s->mutex);
SDL_QuitSubSystem (SDL_INIT_AUDIO);
return NULL;
@@ -313,20 +404,36 @@ static void sdl_audio_fini (void *opaque)
SDL_QuitSubSystem (SDL_INIT_AUDIO);
}
-struct pcm_ops sdl_pcm_ops = {
- sdl_hw_init,
- sdl_hw_fini,
- sdl_hw_run,
- sdl_hw_write,
- sdl_hw_ctl
+static struct audio_option sdl_options[] = {
+ {"SAMPLES", AUD_OPT_INT, &conf.nb_samples,
+ "Size of SDL buffer in samples", NULL, 0},
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+static struct audio_pcm_ops sdl_pcm_ops = {
+ sdl_init_out,
+ sdl_fini_out,
+ sdl_run_out,
+ sdl_write_out,
+ sdl_ctl_out,
+
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ NULL
};
-struct audio_output_driver sdl_output_driver = {
- "sdl",
- sdl_audio_init,
- sdl_audio_fini,
- &sdl_pcm_ops,
- 1,
- 1,
- sizeof (SDLVoice)
+struct audio_driver sdl_audio_driver = {
+ INIT_FIELD (name = ) "sdl",
+ INIT_FIELD (descr = ) "SDL http://www.libsdl.org",
+ INIT_FIELD (options = ) sdl_options,
+ INIT_FIELD (init = ) sdl_audio_init,
+ INIT_FIELD (fini = ) sdl_audio_fini,
+ INIT_FIELD (pcm_ops = ) &sdl_pcm_ops,
+ INIT_FIELD (can_be_default = ) 1,
+ INIT_FIELD (max_voices_out = ) 1,
+ INIT_FIELD (max_voices_in = ) 0,
+ INIT_FIELD (voice_size_out = ) sizeof (SDLVoiceOut),
+ INIT_FIELD (voice_size_in = ) 0
};
diff --git a/audio/sys-queue.h b/audio/sys-queue.h
new file mode 100644
index 0000000000..5b6e2a0a23
--- /dev/null
+++ b/audio/sys-queue.h
@@ -0,0 +1,241 @@
+/*
+ * Copyright (c) 1991, 1993
+ * The Regents of the University of California. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ * 4. Neither the name of the University nor the names of its contributors
+ * may be used to endorse or promote products derived from this software
+ * without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
+ * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
+ * SUCH DAMAGE.
+ *
+ * @(#)queue.h 8.3 (Berkeley) 12/13/93
+ */
+
+#ifndef _SYS_QUEUE_H
+#define _SYS_QUEUE_H 1
+
+/*
+ * This file defines three types of data structures: lists, tail queues,
+ * and circular queues.
+ *
+ * A list is headed by a single forward pointer (or an array of forward
+ * pointers for a hash table header). The elements are doubly linked
+ * so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list after
+ * an existing element or at the head of the list. A list may only be
+ * traversed in the forward direction.
+ *
+ * A tail queue is headed by a pair of pointers, one to the head of the
+ * list and the other to the tail of the list. The elements are doubly
+ * linked so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list after
+ * an existing element, at the head of the list, or at the end of the
+ * list. A tail queue may only be traversed in the forward direction.
+ *
+ * A circle queue is headed by a pair of pointers, one to the head of the
+ * list and the other to the tail of the list. The elements are doubly
+ * linked so that an arbitrary element can be removed without a need to
+ * traverse the list. New elements can be added to the list before or after
+ * an existing element, at the head of the list, or at the end of the list.
+ * A circle queue may be traversed in either direction, but has a more
+ * complex end of list detection.
+ *
+ * For details on the use of these macros, see the queue(3) manual page.
+ */
+
+/*
+ * List definitions.
+ */
+#define LIST_HEAD(name, type) \
+struct name { \
+ struct type *lh_first; /* first element */ \
+}
+
+#define LIST_ENTRY(type) \
+struct { \
+ struct type *le_next; /* next element */ \
+ struct type **le_prev; /* address of previous next element */ \
+}
+
+/*
+ * List functions.
+ */
+#define LIST_INIT(head) { \
+ (head)->lh_first = NULL; \
+}
+
+#define LIST_INSERT_AFTER(listelm, elm, field) { \
+ if (((elm)->field.le_next = (listelm)->field.le_next) != NULL) \
+ (listelm)->field.le_next->field.le_prev = \
+ &(elm)->field.le_next; \
+ (listelm)->field.le_next = (elm); \
+ (elm)->field.le_prev = &(listelm)->field.le_next; \
+}
+
+#define LIST_INSERT_HEAD(head, elm, field) { \
+ if (((elm)->field.le_next = (head)->lh_first) != NULL) \
+ (head)->lh_first->field.le_prev = &(elm)->field.le_next;\
+ (head)->lh_first = (elm); \
+ (elm)->field.le_prev = &(head)->lh_first; \
+}
+
+#define LIST_REMOVE(elm, field) { \
+ if ((elm)->field.le_next != NULL) \
+ (elm)->field.le_next->field.le_prev = \
+ (elm)->field.le_prev; \
+ *(elm)->field.le_prev = (elm)->field.le_next; \
+}
+
+/*
+ * Tail queue definitions.
+ */
+#define TAILQ_HEAD(name, type) \
+struct name { \
+ struct type *tqh_first; /* first element */ \
+ struct type **tqh_last; /* addr of last next element */ \
+}
+
+#define TAILQ_ENTRY(type) \
+struct { \
+ struct type *tqe_next; /* next element */ \
+ struct type **tqe_prev; /* address of previous next element */ \
+}
+
+/*
+ * Tail queue functions.
+ */
+#define TAILQ_INIT(head) { \
+ (head)->tqh_first = NULL; \
+ (head)->tqh_last = &(head)->tqh_first; \
+}
+
+#define TAILQ_INSERT_HEAD(head, elm, field) { \
+ if (((elm)->field.tqe_next = (head)->tqh_first) != NULL) \
+ (elm)->field.tqe_next->field.tqe_prev = \
+ &(elm)->field.tqe_next; \
+ else \
+ (head)->tqh_last = &(elm)->field.tqe_next; \
+ (head)->tqh_first = (elm); \
+ (elm)->field.tqe_prev = &(head)->tqh_first; \
+}
+
+#define TAILQ_INSERT_TAIL(head, elm, field) { \
+ (elm)->field.tqe_next = NULL; \
+ (elm)->field.tqe_prev = (head)->tqh_last; \
+ *(head)->tqh_last = (elm); \
+ (head)->tqh_last = &(elm)->field.tqe_next; \
+}
+
+#define TAILQ_INSERT_AFTER(head, listelm, elm, field) { \
+ if (((elm)->field.tqe_next = (listelm)->field.tqe_next) != NULL)\
+ (elm)->field.tqe_next->field.tqe_prev = \
+ &(elm)->field.tqe_next; \
+ else \
+ (head)->tqh_last = &(elm)->field.tqe_next; \
+ (listelm)->field.tqe_next = (elm); \
+ (elm)->field.tqe_prev = &(listelm)->field.tqe_next; \
+}
+
+#define TAILQ_REMOVE(head, elm, field) { \
+ if (((elm)->field.tqe_next) != NULL) \
+ (elm)->field.tqe_next->field.tqe_prev = \
+ (elm)->field.tqe_prev; \
+ else \
+ (head)->tqh_last = (elm)->field.tqe_prev; \
+ *(elm)->field.tqe_prev = (elm)->field.tqe_next; \
+}
+
+/*
+ * Circular queue definitions.
+ */
+#define CIRCLEQ_HEAD(name, type) \
+struct name { \
+ struct type *cqh_first; /* first element */ \
+ struct type *cqh_last; /* last element */ \
+}
+
+#define CIRCLEQ_ENTRY(type) \
+struct { \
+ struct type *cqe_next; /* next element */ \
+ struct type *cqe_prev; /* previous element */ \
+}
+
+/*
+ * Circular queue functions.
+ */
+#define CIRCLEQ_INIT(head) { \
+ (head)->cqh_first = (void *)(head); \
+ (head)->cqh_last = (void *)(head); \
+}
+
+#define CIRCLEQ_INSERT_AFTER(head, listelm, elm, field) { \
+ (elm)->field.cqe_next = (listelm)->field.cqe_next; \
+ (elm)->field.cqe_prev = (listelm); \
+ if ((listelm)->field.cqe_next == (void *)(head)) \
+ (head)->cqh_last = (elm); \
+ else \
+ (listelm)->field.cqe_next->field.cqe_prev = (elm); \
+ (listelm)->field.cqe_next = (elm); \
+}
+
+#define CIRCLEQ_INSERT_BEFORE(head, listelm, elm, field) { \
+ (elm)->field.cqe_next = (listelm); \
+ (elm)->field.cqe_prev = (listelm)->field.cqe_prev; \
+ if ((listelm)->field.cqe_prev == (void *)(head)) \
+ (head)->cqh_first = (elm); \
+ else \
+ (listelm)->field.cqe_prev->field.cqe_next = (elm); \
+ (listelm)->field.cqe_prev = (elm); \
+}
+
+#define CIRCLEQ_INSERT_HEAD(head, elm, field) { \
+ (elm)->field.cqe_next = (head)->cqh_first; \
+ (elm)->field.cqe_prev = (void *)(head); \
+ if ((head)->cqh_last == (void *)(head)) \
+ (head)->cqh_last = (elm); \
+ else \
+ (head)->cqh_first->field.cqe_prev = (elm); \
+ (head)->cqh_first = (elm); \
+}
+
+#define CIRCLEQ_INSERT_TAIL(head, elm, field) { \
+ (elm)->field.cqe_next = (void *)(head); \
+ (elm)->field.cqe_prev = (head)->cqh_last; \
+ if ((head)->cqh_first == (void *)(head)) \
+ (head)->cqh_first = (elm); \
+ else \
+ (head)->cqh_last->field.cqe_next = (elm); \
+ (head)->cqh_last = (elm); \
+}
+
+#define CIRCLEQ_REMOVE(head, elm, field) { \
+ if ((elm)->field.cqe_next == (void *)(head)) \
+ (head)->cqh_last = (elm)->field.cqe_prev; \
+ else \
+ (elm)->field.cqe_next->field.cqe_prev = \
+ (elm)->field.cqe_prev; \
+ if ((elm)->field.cqe_prev == (void *)(head)) \
+ (head)->cqh_first = (elm)->field.cqe_next; \
+ else \
+ (elm)->field.cqe_prev->field.cqe_next = \
+ (elm)->field.cqe_next; \
+}
+#endif /* sys/queue.h */
diff --git a/audio/wavaudio.c b/audio/wavaudio.c
index 5680161c72..e9bd87872f 100644
--- a/audio/wavaudio.c
+++ b/audio/wavaudio.c
@@ -1,8 +1,8 @@
/*
- * QEMU WAV audio output driver
- *
- * Copyright (c) 2004 Vassili Karpov (malc)
- *
+ * QEMU WAV audio driver
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -23,22 +23,16 @@
*/
#include "vl.h"
-#include "audio/audio_int.h"
+#define AUDIO_CAP "wav"
+#include "audio_int.h"
-typedef struct WAVVoice {
- HWVoice hw;
+typedef struct WAVVoiceOut {
+ HWVoiceOut hw;
QEMUFile *f;
int64_t old_ticks;
void *pcm_buf;
int total_samples;
-} WAVVoice;
-
-#define dolog(...) AUD_log ("wav", __VA_ARGS__)
-#ifdef DEBUG
-#define ldebug(...) dolog (__VA_ARGS__)
-#else
-#define ldebug(...)
-#endif
+} WAVVoiceOut;
static struct {
const char *wav_path;
@@ -46,24 +40,27 @@ static struct {
.wav_path = "qemu.wav"
};
-static void wav_hw_run (HWVoice *hw)
+static int wav_run_out (HWVoiceOut *hw)
{
- WAVVoice *wav = (WAVVoice *) hw;
+ WAVVoiceOut *wav = (WAVVoiceOut *) hw;
int rpos, live, decr, samples;
uint8_t *dst;
st_sample_t *src;
int64_t now = qemu_get_clock (vm_clock);
int64_t ticks = now - wav->old_ticks;
- int64_t bytes = (ticks * hw->bytes_per_second) / ticks_per_sec;
+ int64_t bytes = (ticks * hw->info.bytes_per_second) / ticks_per_sec;
- if (bytes > INT_MAX)
- samples = INT_MAX >> hw->shift;
- else
- samples = bytes >> hw->shift;
+ if (bytes > INT_MAX) {
+ samples = INT_MAX >> hw->info.shift;
+ }
+ else {
+ samples = bytes >> hw->info.shift;
+ }
- live = pcm_hw_get_live (hw);
- if (live <= 0)
- return;
+ live = audio_pcm_hw_get_live_out (hw);
+ if (!live) {
+ return 0;
+ }
wav->old_ticks = now;
decr = audio_MIN (live, samples);
@@ -73,25 +70,25 @@ static void wav_hw_run (HWVoice *hw)
int left_till_end_samples = hw->samples - rpos;
int convert_samples = audio_MIN (samples, left_till_end_samples);
- src = advance (hw->mix_buf, rpos * sizeof (st_sample_t));
- dst = advance (wav->pcm_buf, rpos << hw->shift);
+ src = hw->mix_buf + rpos;
+ dst = advance (wav->pcm_buf, rpos << hw->info.shift);
hw->clip (dst, src, convert_samples);
- qemu_put_buffer (wav->f, dst, convert_samples << hw->shift);
- memset (src, 0, convert_samples * sizeof (st_sample_t));
+ qemu_put_buffer (wav->f, dst, convert_samples << hw->info.shift);
+ mixeng_clear (src, convert_samples);
rpos = (rpos + convert_samples) % hw->samples;
samples -= convert_samples;
wav->total_samples += convert_samples;
}
- pcm_hw_dec_live (hw, decr);
hw->rpos = rpos;
+ return decr;
}
-static int wav_hw_write (SWVoice *sw, void *buf, int len)
+static int wav_write_out (SWVoiceOut *sw, void *buf, int len)
{
- return pcm_hw_write (sw, buf, len);
+ return audio_pcm_sw_write (sw, buf, len);
}
/* VICE code: Store number as little endian. */
@@ -104,10 +101,10 @@ static void le_store (uint8_t *buf, uint32_t val, int len)
}
}
-static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
+static int wav_init_out (HWVoiceOut *hw, int freq, int nchannels, audfmt_e fmt)
{
- WAVVoice *wav = (WAVVoice *) hw;
- int bits16 = 0, stereo = audio_state.fixed_channels == 2;
+ WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+ int bits16;
uint8_t hdr[] = {
0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56,
0x45, 0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00,
@@ -115,34 +112,50 @@ static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
0x00, 0x10, 0x00, 0x64, 0x61, 0x74, 0x61, 0x00, 0x00, 0x00, 0x00
};
- switch (audio_state.fixed_fmt) {
+ freq = audio_state.fixed_freq_out;
+ fmt = audio_state.fixed_fmt_out;
+ nchannels = audio_state.fixed_channels_out;
+
+ switch (fmt) {
case AUD_FMT_S8:
case AUD_FMT_U8:
+ bits16 = 0;
break;
case AUD_FMT_S16:
case AUD_FMT_U16:
bits16 = 1;
break;
+
+ default:
+ dolog ("Internal logic error bad format %d\n", fmt);
+ return -1;
}
hdr[34] = bits16 ? 0x10 : 0x08;
- hw->freq = 44100;
- hw->nchannels = stereo ? 2 : 1;
- hw->fmt = bits16 ? AUD_FMT_S16 : AUD_FMT_U8;
+ audio_pcm_init_info (
+ &hw->info,
+ freq,
+ nchannels,
+ bits16 ? AUD_FMT_S16 : AUD_FMT_U8,
+ audio_need_to_swap_endian (0)
+ );
hw->bufsize = 4096;
wav->pcm_buf = qemu_mallocz (hw->bufsize);
- if (!wav->pcm_buf)
+ if (!wav->pcm_buf) {
+ dolog ("Can not initialize WAV buffer of %d bytes\n",
+ hw->bufsize);
return -1;
+ }
- le_store (hdr + 22, hw->nchannels, 2);
- le_store (hdr + 24, hw->freq, 4);
- le_store (hdr + 28, hw->freq << (bits16 + stereo), 4);
- le_store (hdr + 32, 1 << (bits16 + stereo), 2);
+ le_store (hdr + 22, hw->info.nchannels, 2);
+ le_store (hdr + 24, hw->info.freq, 4);
+ le_store (hdr + 28, hw->info.freq << (bits16 + (nchannels == 2)), 4);
+ le_store (hdr + 32, 1 << (bits16 + (nchannels == 2)), 2);
wav->f = fopen (conf.wav_path, "wb");
if (!wav->f) {
- dolog ("failed to open wave file `%s'\nReason: %s\n",
+ dolog ("Failed to open wave file `%s'\nReason: %s\n",
conf.wav_path, strerror (errno));
qemu_free (wav->pcm_buf);
wav->pcm_buf = NULL;
@@ -153,17 +166,18 @@ static int wav_hw_init (HWVoice *hw, int freq, int nchannels, audfmt_e fmt)
return 0;
}
-static void wav_hw_fini (HWVoice *hw)
+static void wav_fini_out (HWVoiceOut *hw)
{
- WAVVoice *wav = (WAVVoice *) hw;
- int stereo = hw->nchannels == 2;
+ WAVVoiceOut *wav = (WAVVoiceOut *) hw;
+ int stereo = hw->info.nchannels == 2;
uint8_t rlen[4];
uint8_t dlen[4];
uint32_t rifflen = (wav->total_samples << stereo) + 36;
uint32_t datalen = wav->total_samples << stereo;
- if (!wav->f || !hw->active)
+ if (!wav->f || !hw->active) {
return;
+ }
le_store (rlen, rifflen, 4);
le_store (dlen, datalen, 4);
@@ -181,7 +195,7 @@ static void wav_hw_fini (HWVoice *hw)
wav->pcm_buf = NULL;
}
-static int wav_hw_ctl (HWVoice *hw, int cmd, ...)
+static int wav_ctl_out (HWVoiceOut *hw, int cmd, ...)
{
(void) hw;
(void) cmd;
@@ -195,23 +209,41 @@ static void *wav_audio_init (void)
static void wav_audio_fini (void *opaque)
{
+ (void) opaque;
ldebug ("wav_fini");
}
-struct pcm_ops wav_pcm_ops = {
- wav_hw_init,
- wav_hw_fini,
- wav_hw_run,
- wav_hw_write,
- wav_hw_ctl
+struct audio_option wav_options[] = {
+ {"PATH", AUD_OPT_STR, &conf.wav_path,
+ "Path to wave file", NULL, 0},
+ {NULL, 0, NULL, NULL, NULL, 0}
+};
+
+struct audio_pcm_ops wav_pcm_ops = {
+ wav_init_out,
+ wav_fini_out,
+ wav_run_out,
+ wav_write_out,
+ wav_ctl_out,
+
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ NULL
};
-struct audio_output_driver wav_output_driver = {
- "wav",
- wav_audio_init,
- wav_audio_fini,
- &wav_pcm_ops,
- 1,
- 1,
- sizeof (WAVVoice)
+struct audio_driver wav_audio_driver = {
+ INIT_FIELD (name = ) "wav",
+ INIT_FIELD (descr = )
+ "WAV renderer http://wikipedia.org/wiki/WAV",
+ INIT_FIELD (options = ) wav_options,
+ INIT_FIELD (init = ) wav_audio_init,
+ INIT_FIELD (fini = ) wav_audio_fini,
+ INIT_FIELD (pcm_ops = ) &wav_pcm_ops,
+ INIT_FIELD (can_be_default = ) 0,
+ INIT_FIELD (max_voices_out = ) 1,
+ INIT_FIELD (max_voices_in = ) 0,
+ INIT_FIELD (voice_size_out = ) sizeof (WAVVoiceOut),
+ INIT_FIELD (voice_size_in = ) 0
};
diff --git a/configure b/configure
index 19a0ec900f..98ef82b6e1 100755
--- a/configure
+++ b/configure
@@ -77,6 +77,9 @@ gdbstub="yes"
slirp="yes"
adlib="no"
oss="no"
+dsound="no"
+coreaudio="no"
+alsa="no"
fmod="no"
fmod_lib=""
fmod_inc=""
@@ -115,7 +118,7 @@ Darwin)
bsd="yes"
darwin="yes"
;;
-*)
+*)
oss="yes"
linux="yes"
if [ "$cpu" = "i386" -o "$cpu" = "x86_64" ] ; then
@@ -131,7 +134,7 @@ if [ "$bsd" = "yes" ] ; then
fi
# find source path
-# XXX: we assume an absolute path is given when launching configure,
+# XXX: we assume an absolute path is given when launching configure,
# except in './configure' case.
source_path=${0%configure}
source_path=${source_path%/}
@@ -171,6 +174,12 @@ for opt do
;;
--disable-sdl) sdl="no"
;;
+ --enable-coreaudio) coreaudio="yes"
+ ;;
+ --enable-alsa) alsa="yes"
+ ;;
+ --enable-dsound) dsound="yes"
+ ;;
--enable-fmod) fmod="yes"
;;
--fmod-lib=*) fmod_lib=${opt#--fmod-lib=}
@@ -178,17 +187,17 @@ for opt do
--fmod-inc=*) fmod_inc=${opt#--fmod-inc=}
;;
--enable-mingw32) mingw32="yes" ; cross_prefix="i386-mingw32-"
- ;;
+ ;;
--disable-slirp) slirp="no"
- ;;
+ ;;
--enable-adlib) adlib="yes"
- ;;
+ ;;
--disable-kqemu) kqemu="no"
- ;;
+ ;;
--kernel-path=*) kernel_path=${opt#--kernel-path=}
- ;;
- --enable-cocoa) cocoa="yes" ; sdl="no"
- ;;
+ ;;
+ --enable-cocoa) cocoa="yes" ; coreaudio="yes" ; sdl="no"
+ ;;
--disable-gfx-check) check_gfx="no"
;;
esac
@@ -231,8 +240,8 @@ if test -z "$cross_prefix" ; then
cat > $TMPC << EOF
#include <inttypes.h>
int main(int argc, char ** argv){
- volatile uint32_t i=0x01234567;
- return (*((uint8_t*)(&i))) == 0x67;
+ volatile uint32_t i=0x01234567;
+ return (*((uint8_t*)(&i))) == 0x67;
}
EOF
@@ -346,7 +355,10 @@ echo " --make=MAKE use specified make [$make]"
echo " --static enable static build [$static]"
echo " --enable-mingw32 enable Win32 cross compilation with mingw32"
echo " --enable-adlib enable Adlib emulation"
-echo " --enable-fmod enable FMOD audio output driver"
+echo " --enable-coreaudio enable Coreaudio audio driver"
+echo " --enable-alsa enable ALSA audio driver"
+echo " --enable-fmod enable FMOD audio driver"
+echo " --enabled-dsound enable DirectSound audio driver"
echo " --fmod-lib path to FMOD library"
echo " --fmod-inc path to FMOD includes"
echo ""
@@ -375,20 +387,20 @@ fi
# kqemu support
if test $kqemu = "yes" ; then
# test if the source code is installed
- if test '!' -f "kqemu/Makefile" ; then
+ if test '!' -f "kqemu/Makefile" ; then
kqemu="no"
fi
fi
-
+
# Linux specific kqemu configuration
if test $kqemu = "yes" -a $linux = "yes" ; then
# find the kernel path
if test -z "$kernel_path" ; then
kernel_version=`uname -r`
kernel_path="/lib/modules/$kernel_version/build"
-if test '!' -d "$kernel_path/include" ; then
+if test '!' -d "$kernel_path/include" ; then
kernel_path="/usr/src/linux"
- if test '!' -d "$kernel_path/include" ; then
+ if test '!' -d "$kernel_path/include" ; then
echo "Could not find kernel includes in /lib/modules or /usr/src/linux - cannot build the kqemu module"
kqemu="no"
fi
@@ -401,7 +413,7 @@ if test $kqemu = "yes" ; then
if test '!' -f "$kernel_path/Makefile" ; then
echo "No Makefile file present in $kernel_path - kqemu cannot be built"
kqemu="no"
-fi
+fi
# find build system (2.6 or legacy)
kbuild26="yes"
@@ -439,8 +451,18 @@ if test "$sdl" != "no" ; then
fi
echo "mingw32 support $mingw32"
echo "Adlib support $adlib"
+echo "CoreAudio support $coreaudio"
+echo "ALSA support $alsa"
+echo "DSound support $dsound"
echo -n "FMOD support $fmod"
-if test $fmod = "yes"; then
+if test "$fmod" = "yes"; then
+ if test -z $fmod_lib || test -z $fmod_inc; then
+ echo
+ echo "Error: You must specify path to FMOD library and headers"
+ echo "Example: --fmod-inc=/path/include/fmod --fmod-lib=/path/lib/libfmod-3.74.so"
+ echo
+ exit 1
+ fi
echo -n " (lib='$fmod_lib' include='$fmod_inc')"
fi
echo ""
@@ -568,6 +590,18 @@ if test "$oss" = "yes" ; then
echo "CONFIG_OSS=yes" >> $config_mak
echo "#define CONFIG_OSS 1" >> $config_h
fi
+if test "$coreaudio" = "yes" ; then
+ echo "CONFIG_COREAUDIO=yes" >> $config_mak
+ echo "#define CONFIG_COREAUDIO 1" >> $config_h
+fi
+if test "$alsa" = "yes" ; then
+ echo "CONFIG_ALSA=yes" >> $config_mak
+ echo "#define CONFIG_ALSA 1" >> $config_h
+fi
+if test "$dsound" = "yes" ; then
+ echo "CONFIG_DSOUND=yes" >> $config_mak
+ echo "#define CONFIG_DSOUND 1" >> $config_h
+fi
if test "$fmod" = "yes" ; then
echo "CONFIG_FMOD=yes" >> $config_mak
echo "CONFIG_FMOD_LIB=$fmod_lib" >> $config_mak
@@ -600,7 +634,7 @@ if [ "$bsd" = "yes" ] ; then
echo "#define _BSD 1" >> $config_h
fi
-for target in $target_list; do
+for target in $target_list; do
target_dir="$target"
config_mak=$target_dir/config.mak
@@ -623,7 +657,7 @@ if expr $target : '.*-user' > /dev/null ; then
fi
if test "$target_user_only" = "no" -a "$check_gfx" = "yes" \
- -a "$sdl" = "no" -a "$cocoa" = "no" ; then
+ -a "$sdl" = "no" -a "$cocoa" = "no" ; then
echo "ERROR: QEMU requires SDL or Cocoa for graphical output"
echo "To build QEMU with graphical output configure with --disable-gfx-check"
echo "Note that this will disable all output from the virtual graphics card."
diff --git a/hw/adlib.c b/hw/adlib.c
index 939a7ed036..70de4ffab1 100644
--- a/hw/adlib.c
+++ b/hw/adlib.c
@@ -1,8 +1,8 @@
/*
- * QEMU Adlib emulation
- *
- * Copyright (c) 2004 Vassili Karpov (malc)
- *
+ * QEMU Proxy for OPL2/3 emulation by MAME team
+ *
+ * Copyright (c) 2004-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -21,8 +21,11 @@
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
+#include <assert.h>
#include "vl.h"
+#define ADLIB_KILL_TIMERS 1
+
#define dolog(...) AUD_log ("adlib", __VA_ARGS__)
#ifdef DEBUG
#define ldebug(...) dolog (__VA_ARGS__)
@@ -30,23 +33,15 @@
#define ldebug(...)
#endif
-#ifdef USE_YMF262
-#define HAS_YMF262 1
+#ifdef HAS_YMF262
#include "ymf262.h"
-void YMF262UpdateOneQEMU(int which, INT16 *dst, int length);
+void YMF262UpdateOneQEMU (int which, INT16 *dst, int length);
#define SHIFT 2
#else
#include "fmopl.h"
#define SHIFT 1
#endif
-#ifdef _WIN32
-#include <windows.h>
-#define small_delay() Sleep (1)
-#else
-#define small_delay() usleep (1)
-#endif
-
#define IO_READ_PROTO(name) \
uint32_t name (void *opaque, uint32_t nport)
#define IO_WRITE_PROTO(name) \
@@ -58,35 +53,70 @@ static struct {
} conf = {0x220, 44100};
typedef struct {
+ int ticking[2];
int enabled;
int active;
- int cparam;
- int64_t ticks;
int bufpos;
+#ifdef DEBUG
+ int64_t exp[2];
+#endif
int16_t *mixbuf;
- double interval;
- QEMUTimer *ts, *opl_ts;
- SWVoice *voice;
- int left, pos, samples, bytes_per_second, old_free;
- int refcount;
-#ifndef USE_YMF262
+ uint64_t dexp[2];
+ SWVoiceOut *voice;
+ int left, pos, samples;
+ QEMUAudioTimeStamp ats;
+#ifndef HAS_YMF262
FM_OPL *opl;
#endif
} AdlibState;
static AdlibState adlib;
+static void adlib_stop_opl_timer (AdlibState *s, size_t n)
+{
+#ifdef HAS_YMF262
+ YMF262TimerOver (0, n);
+#else
+ OPLTimerOver (s->opl, n);
+#endif
+ s->ticking[n] = 0;
+}
+
+static void adlib_kill_timers (AdlibState *s)
+{
+ size_t i;
+
+ for (i = 0; i < 2; ++i) {
+ if (s->ticking[i]) {
+ uint64_t delta;
+
+ delta = AUD_time_stamp_get_elapsed_usec_out (s->voice, &s->ats);
+ ldebug (
+ "delta = %f dexp = %f expired => %d\n",
+ delta / 1000000.0,
+ s->dexp[i] / 1000000.0,
+ delta >= s->dexp[i]
+ );
+ if (ADLIB_KILL_TIMERS || delta >= s->dexp[i]) {
+ adlib_stop_opl_timer (s, i);
+ AUD_init_time_stamp_out (s->voice, &s->ats);
+ }
+ }
+ }
+}
+
static IO_WRITE_PROTO(adlib_write)
{
AdlibState *s = opaque;
int a = nport & 3;
int status;
- s->ticks = qemu_get_clock (vm_clock);
s->active = 1;
- AUD_enable (s->voice, 1);
+ AUD_set_active_out (s->voice, 1);
-#ifdef USE_YMF262
+ adlib_kill_timers (s);
+
+#ifdef HAS_YMF262
status = YMF262Write (0, a, val);
#else
status = OPLWrite (s->opl, a, val);
@@ -99,8 +129,9 @@ static IO_READ_PROTO(adlib_read)
uint8_t data;
int a = nport & 3;
-#ifdef USE_YMF262
- (void) s;
+ adlib_kill_timers (s);
+
+#ifdef HAS_YMF262
data = YMF262Read (0, a);
#else
data = OPLRead (s->opl, a);
@@ -108,119 +139,115 @@ static IO_READ_PROTO(adlib_read)
return data;
}
-static void OPL_timer (void *opaque)
+static void timer_handler (int c, double interval_Sec)
{
- AdlibState *s = opaque;
-#ifdef USE_YMF262
- YMF262TimerOver (s->cparam >> 1, s->cparam & 1);
-#else
- OPLTimerOver (s->opl, s->cparam);
+ AdlibState *s = &adlib;
+ unsigned n = c & 1;
+#ifdef DEBUG
+ double interval;
#endif
- qemu_mod_timer (s->opl_ts, qemu_get_clock (vm_clock) + s->interval);
-}
-static void YMF262TimerHandler (int c, double interval_Sec)
-{
- AdlibState *s = &adlib;
if (interval_Sec == 0.0) {
- qemu_del_timer (s->opl_ts);
+ s->ticking[n] = 0;
return;
}
- s->cparam = c;
- s->interval = ticks_per_sec * interval_Sec;
- qemu_mod_timer (s->opl_ts, qemu_get_clock (vm_clock) + s->interval);
- small_delay ();
+
+ s->ticking[n] = 1;
+#ifdef DEBUG
+ interval = ticks_per_sec * interval_Sec;
+ exp = qemu_get_clock (vm_clock) + interval;
+ s->exp[n] = exp;
+#endif
+
+ s->dexp[n] = interval_Sec * 1000000.0;
+ AUD_init_time_stamp_out (s->voice, &s->ats);
}
static int write_audio (AdlibState *s, int samples)
{
int net = 0;
- int ss = samples;
+ int pos = s->pos;
+
while (samples) {
- int nbytes = samples << SHIFT;
- int wbytes = AUD_write (s->voice,
- s->mixbuf + (s->pos << (SHIFT - 1)),
- nbytes);
- int wsampl = wbytes >> SHIFT;
- samples -= wsampl;
- s->pos = (s->pos + wsampl) % s->samples;
- net += wsampl;
- if (!wbytes)
+ int nbytes, wbytes, wsampl;
+
+ nbytes = samples << SHIFT;
+ wbytes = AUD_write (
+ s->voice,
+ s->mixbuf + (pos << (SHIFT - 1)),
+ nbytes
+ );
+
+ if (wbytes) {
+ wsampl = wbytes >> SHIFT;
+
+ samples -= wsampl;
+ pos = (pos + wsampl) % s->samples;
+
+ net += wsampl;
+ }
+ else {
break;
+ }
}
- if (net > ss) {
- dolog ("WARNING: net > ss\n");
- }
+
return net;
}
-static void timer (void *opaque)
+static void adlib_callback (void *opaque, int free)
{
AdlibState *s = opaque;
- int elapsed, samples, net = 0;
-
- if (s->refcount)
- dolog ("refcount=%d\n", s->refcount);
+ int samples, net = 0, to_play, written;
- s->refcount += 1;
- if (!(s->active && s->enabled))
- goto reset;
-
- AUD_run ();
-
- while (s->left) {
- int written = write_audio (s, s->left);
- net += written;
- if (!written)
- goto reset2;
- s->left -= written;
+ samples = free >> SHIFT;
+ if (!(s->active && s->enabled) || !samples) {
+ return;
}
- s->pos = 0;
-
- elapsed = AUD_calc_elapsed (s->voice);
- if (!elapsed)
- goto reset2;
- /* elapsed = AUD_get_free (s->voice); */
- samples = elapsed >> SHIFT;
- if (!samples)
- goto reset2;
+ to_play = audio_MIN (s->left, samples);
+ while (to_play) {
+ written = write_audio (s, to_play);
+
+ if (written) {
+ s->left -= written;
+ samples -= written;
+ to_play -= written;
+ s->pos = (s->pos + written) % s->samples;
+ }
+ else {
+ return;
+ }
+ }
samples = audio_MIN (samples, s->samples - s->pos);
- if (s->left)
- dolog ("left=%d samples=%d elapsed=%d free=%d\n",
- s->left, samples, elapsed, AUD_get_free (s->voice));
-
- if (!samples)
- goto reset2;
+ if (!samples) {
+ return;
+ }
-#ifdef USE_YMF262
+#ifdef HAS_YMF262
YMF262UpdateOneQEMU (0, s->mixbuf + s->pos * 2, samples);
#else
YM3812UpdateOne (s->opl, s->mixbuf + s->pos, samples);
#endif
while (samples) {
- int written = write_audio (s, samples);
- net += written;
- if (!written)
- break;
- samples -= written;
+ written = write_audio (s, samples);
+
+ if (written) {
+ net += written;
+ samples -= written;
+ s->pos = (s->pos + written) % s->samples;
+ }
+ else {
+ s->left = samples;
+ return;
+ }
}
- if (!samples)
- s->pos = 0;
- s->left = samples;
-
-reset2:
- AUD_adjust (s->voice, net << SHIFT);
-reset:
- qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + ticks_per_sec / 1024);
- s->refcount -= 1;
}
static void Adlib_fini (AdlibState *s)
{
-#ifdef USE_YMF262
+#ifdef HAS_YMF262
YMF262Shutdown ();
#else
if (s->opl) {
@@ -229,15 +256,9 @@ static void Adlib_fini (AdlibState *s)
}
#endif
- if (s->opl_ts)
- qemu_free_timer (s->opl_ts);
-
- if (s->ts)
- qemu_free_timer (s->ts);
-
-#define maybe_free(p) if (p) qemu_free (p)
- maybe_free (s->mixbuf);
-#undef maybe_free
+ if (s->mixbuf) {
+ qemu_free (s->mixbuf);
+ }
s->active = 0;
s->enabled = 0;
@@ -247,15 +268,13 @@ void Adlib_init (void)
{
AdlibState *s = &adlib;
- memset (s, 0, sizeof (*s));
-
-#ifdef USE_YMF262
+#ifdef HAS_YMF262
if (YMF262Init (1, 14318180, conf.freq)) {
dolog ("YMF262Init %d failed\n", conf.freq);
return;
}
else {
- YMF262SetTimerHandler (0, YMF262TimerHandler, 0);
+ YMF262SetTimerHandler (0, timer_handler, 0);
s->enabled = 1;
}
#else
@@ -265,33 +284,26 @@ void Adlib_init (void)
return;
}
else {
- OPLSetTimerHandler (s->opl, YMF262TimerHandler, 0);
+ OPLSetTimerHandler (s->opl, timer_handler, 0);
s->enabled = 1;
}
#endif
- s->opl_ts = qemu_new_timer (vm_clock, OPL_timer, s);
- if (!s->opl_ts) {
- dolog ("Can not get timer for adlib emulation\n");
- Adlib_fini (s);
- return;
- }
-
- s->ts = qemu_new_timer (vm_clock, timer, s);
- if (!s->opl_ts) {
- dolog ("Can not get timer for adlib emulation\n");
- Adlib_fini (s);
- return;
- }
-
- s->voice = AUD_open (s->voice, "adlib", conf.freq, SHIFT, AUD_FMT_S16);
+ s->voice = AUD_open_out (
+ s->voice,
+ "adlib",
+ s,
+ adlib_callback,
+ conf.freq,
+ SHIFT,
+ AUD_FMT_S16
+ );
if (!s->voice) {
Adlib_fini (s);
return;
}
- s->bytes_per_second = conf.freq << SHIFT;
- s->samples = AUD_get_buffer_size (s->voice) >> SHIFT;
+ s->samples = AUD_get_buffer_size_out (s->voice) >> SHIFT;
s->mixbuf = qemu_mallocz (s->samples << SHIFT);
if (!s->mixbuf) {
@@ -300,6 +312,7 @@ void Adlib_init (void)
Adlib_fini (s);
return;
}
+
register_ioport_read (0x388, 4, 1, adlib_read, s);
register_ioport_write (0x388, 4, 1, adlib_write, s);
@@ -308,6 +321,4 @@ void Adlib_init (void)
register_ioport_read (conf.port + 8, 2, 1, adlib_read, s);
register_ioport_write (conf.port + 8, 2, 1, adlib_write, s);
-
- qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1);
}
diff --git a/hw/es1370.c b/hw/es1370.c
new file mode 100644
index 0000000000..0191f5fdf8
--- /dev/null
+++ b/hw/es1370.c
@@ -0,0 +1,1007 @@
+/*
+ * QEMU ES1370 emulation
+ *
+ * Copyright (c) 2005 Vassili Karpov (malc)
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/* #define DEBUG_ES1370 */
+/* #define VERBOSE_ES1370 */
+#define SILENT_ES1370
+
+#include "vl.h"
+
+/* Missing stuff:
+ SCTRL_P[12](END|ST)INC
+ SCTRL_P1SCTRLD
+ SCTRL_P2DACSEN
+ CTRL_DAC_SYNC
+ MIDI
+ non looped mode
+ surely more
+*/
+
+/*
+ Following macros and samplerate array were copied verbatim from
+ Linux kernel 2.4.30: drivers/sound/es1370.c
+
+ Copyright (C) 1998-2001, 2003 Thomas Sailer (t.sailer@alumni.ethz.ch)
+*/
+
+/* Start blatant GPL violation */
+
+#define ES1370_REG_CONTROL 0x00
+#define ES1370_REG_STATUS 0x04
+#define ES1370_REG_UART_DATA 0x08
+#define ES1370_REG_UART_STATUS 0x09
+#define ES1370_REG_UART_CONTROL 0x09
+#define ES1370_REG_UART_TEST 0x0a
+#define ES1370_REG_MEMPAGE 0x0c
+#define ES1370_REG_CODEC 0x10
+#define ES1370_REG_SERIAL_CONTROL 0x20
+#define ES1370_REG_DAC1_SCOUNT 0x24
+#define ES1370_REG_DAC2_SCOUNT 0x28
+#define ES1370_REG_ADC_SCOUNT 0x2c
+
+#define ES1370_REG_DAC1_FRAMEADR 0xc30
+#define ES1370_REG_DAC1_FRAMECNT 0xc34
+#define ES1370_REG_DAC2_FRAMEADR 0xc38
+#define ES1370_REG_DAC2_FRAMECNT 0xc3c
+#define ES1370_REG_ADC_FRAMEADR 0xd30
+#define ES1370_REG_ADC_FRAMECNT 0xd34
+#define ES1370_REG_PHANTOM_FRAMEADR 0xd38
+#define ES1370_REG_PHANTOM_FRAMECNT 0xd3c
+
+static const unsigned dac1_samplerate[] = { 5512, 11025, 22050, 44100 };
+
+#define DAC2_SRTODIV(x) (((1411200+(x)/2)/(x))-2)
+#define DAC2_DIVTOSR(x) (1411200/((x)+2))
+
+#define CTRL_ADC_STOP 0x80000000 /* 1 = ADC stopped */
+#define CTRL_XCTL1 0x40000000 /* electret mic bias */
+#define CTRL_OPEN 0x20000000 /* no function, can be read and written */
+#define CTRL_PCLKDIV 0x1fff0000 /* ADC/DAC2 clock divider */
+#define CTRL_SH_PCLKDIV 16
+#define CTRL_MSFMTSEL 0x00008000 /* MPEG serial data fmt: 0 = Sony, 1 = I2S */
+#define CTRL_M_SBB 0x00004000 /* DAC2 clock: 0 = PCLKDIV, 1 = MPEG */
+#define CTRL_WTSRSEL 0x00003000 /* DAC1 clock freq: 0=5512, 1=11025, 2=22050, 3=44100 */
+#define CTRL_SH_WTSRSEL 12
+#define CTRL_DAC_SYNC 0x00000800 /* 1 = DAC2 runs off DAC1 clock */
+#define CTRL_CCB_INTRM 0x00000400 /* 1 = CCB "voice" ints enabled */
+#define CTRL_M_CB 0x00000200 /* recording source: 0 = ADC, 1 = MPEG */
+#define CTRL_XCTL0 0x00000100 /* 0 = Line in, 1 = Line out */
+#define CTRL_BREQ 0x00000080 /* 1 = test mode (internal mem test) */
+#define CTRL_DAC1_EN 0x00000040 /* enable DAC1 */
+#define CTRL_DAC2_EN 0x00000020 /* enable DAC2 */
+#define CTRL_ADC_EN 0x00000010 /* enable ADC */
+#define CTRL_UART_EN 0x00000008 /* enable MIDI uart */
+#define CTRL_JYSTK_EN 0x00000004 /* enable Joystick port (presumably at address 0x200) */
+#define CTRL_CDC_EN 0x00000002 /* enable serial (CODEC) interface */
+#define CTRL_SERR_DIS 0x00000001 /* 1 = disable PCI SERR signal */
+
+#define STAT_INTR 0x80000000 /* wired or of all interrupt bits */
+#define STAT_CSTAT 0x00000400 /* 1 = codec busy or codec write in progress */
+#define STAT_CBUSY 0x00000200 /* 1 = codec busy */
+#define STAT_CWRIP 0x00000100 /* 1 = codec write in progress */
+#define STAT_VC 0x00000060 /* CCB int source, 0=DAC1, 1=DAC2, 2=ADC, 3=undef */
+#define STAT_SH_VC 5
+#define STAT_MCCB 0x00000010 /* CCB int pending */
+#define STAT_UART 0x00000008 /* UART int pending */
+#define STAT_DAC1 0x00000004 /* DAC1 int pending */
+#define STAT_DAC2 0x00000002 /* DAC2 int pending */
+#define STAT_ADC 0x00000001 /* ADC int pending */
+
+#define USTAT_RXINT 0x80 /* UART rx int pending */
+#define USTAT_TXINT 0x04 /* UART tx int pending */
+#define USTAT_TXRDY 0x02 /* UART tx ready */
+#define USTAT_RXRDY 0x01 /* UART rx ready */
+
+#define UCTRL_RXINTEN 0x80 /* 1 = enable RX ints */
+#define UCTRL_TXINTEN 0x60 /* TX int enable field mask */
+#define UCTRL_ENA_TXINT 0x20 /* enable TX int */
+#define UCTRL_CNTRL 0x03 /* control field */
+#define UCTRL_CNTRL_SWR 0x03 /* software reset command */
+
+#define SCTRL_P2ENDINC 0x00380000 /* */
+#define SCTRL_SH_P2ENDINC 19
+#define SCTRL_P2STINC 0x00070000 /* */
+#define SCTRL_SH_P2STINC 16
+#define SCTRL_R1LOOPSEL 0x00008000 /* 0 = loop mode */
+#define SCTRL_P2LOOPSEL 0x00004000 /* 0 = loop mode */
+#define SCTRL_P1LOOPSEL 0x00002000 /* 0 = loop mode */
+#define SCTRL_P2PAUSE 0x00001000 /* 1 = pause mode */
+#define SCTRL_P1PAUSE 0x00000800 /* 1 = pause mode */
+#define SCTRL_R1INTEN 0x00000400 /* enable interrupt */
+#define SCTRL_P2INTEN 0x00000200 /* enable interrupt */
+#define SCTRL_P1INTEN 0x00000100 /* enable interrupt */
+#define SCTRL_P1SCTRLD 0x00000080 /* reload sample count register for DAC1 */
+#define SCTRL_P2DACSEN 0x00000040 /* 1 = DAC2 play back last sample when disabled */
+#define SCTRL_R1SEB 0x00000020 /* 1 = 16bit */
+#define SCTRL_R1SMB 0x00000010 /* 1 = stereo */
+#define SCTRL_R1FMT 0x00000030 /* format mask */
+#define SCTRL_SH_R1FMT 4
+#define SCTRL_P2SEB 0x00000008 /* 1 = 16bit */
+#define SCTRL_P2SMB 0x00000004 /* 1 = stereo */
+#define SCTRL_P2FMT 0x0000000c /* format mask */
+#define SCTRL_SH_P2FMT 2
+#define SCTRL_P1SEB 0x00000002 /* 1 = 16bit */
+#define SCTRL_P1SMB 0x00000001 /* 1 = stereo */
+#define SCTRL_P1FMT 0x00000003 /* format mask */
+#define SCTRL_SH_P1FMT 0
+
+/* End blatant GPL violation */
+
+#define NB_CHANNELS 3
+#define DAC1_CHANNEL 0
+#define DAC2_CHANNEL 1
+#define ADC_CHANNEL 2
+
+#define IO_READ_PROTO(n) \
+static uint32_t n (void *opaque, uint32_t addr)
+#define IO_WRITE_PROTO(n) \
+static void n (void *opaque, uint32_t addr, uint32_t val)
+
+static void es1370_dac1_callback (void *opaque, int free);
+static void es1370_dac2_callback (void *opaque, int free);
+static void es1370_adc_callback (void *opaque, int avail);
+
+#ifdef DEBUG_ES1370
+
+#define ldebug(...) AUD_log ("es1370", __VA_ARGS__)
+
+static void print_ctl (uint32_t val)
+{
+ char buf[1024];
+
+ buf[0] = '\0';
+#define a(n) if (val & CTRL_##n) strcat (buf, " "#n)
+ a (ADC_STOP);
+ a (XCTL1);
+ a (OPEN);
+ a (MSFMTSEL);
+ a (M_SBB);
+ a (DAC_SYNC);
+ a (CCB_INTRM);
+ a (M_CB);
+ a (XCTL0);
+ a (BREQ);
+ a (DAC1_EN);
+ a (DAC2_EN);
+ a (ADC_EN);
+ a (UART_EN);
+ a (JYSTK_EN);
+ a (CDC_EN);
+ a (SERR_DIS);
+#undef a
+ AUD_log ("es1370", "ctl - PCLKDIV %d(DAC2 freq %d), freq %d,%s\n",
+ (val & CTRL_PCLKDIV) >> CTRL_SH_PCLKDIV,
+ DAC2_DIVTOSR ((val & CTRL_PCLKDIV) >> CTRL_SH_PCLKDIV),
+ dac1_samplerate[(val & CTRL_WTSRSEL) >> CTRL_SH_WTSRSEL],
+ buf);
+}
+
+static void print_sctl (uint32_t val)
+{
+ static const char *fmt_names[] = {"8M", "8S", "16M", "16S"};
+ char buf[1024];
+
+ buf[0] = '\0';
+
+#define a(n) if (val & SCTRL_##n) strcat (buf, " "#n)
+#define b(n) if (!(val & SCTRL_##n)) strcat (buf, " "#n)
+ b (R1LOOPSEL);
+ b (P2LOOPSEL);
+ b (P1LOOPSEL);
+ a (P2PAUSE);
+ a (P1PAUSE);
+ a (R1INTEN);
+ a (P2INTEN);
+ a (P1INTEN);
+ a (P1SCTRLD);
+ a (P2DACSEN);
+ if (buf[0]) {
+ strcat (buf, "\n ");
+ }
+ else {
+ buf[0] = ' ';
+ buf[1] = '\0';
+ }
+#undef b
+#undef a
+ AUD_log ("es1370",
+ "%s"
+ "p2_end_inc %d, p2_st_inc %d, r1_fmt %s, p2_fmt %s, p1_fmt %s\n",
+ buf,
+ (val & SCTRL_P2ENDINC) >> SCTRL_SH_P2ENDINC,
+ (val & SCTRL_P2STINC) >> SCTRL_SH_P2STINC,
+ fmt_names [(val >> SCTRL_SH_R1FMT) & 3],
+ fmt_names [(val >> SCTRL_SH_P2FMT) & 3],
+ fmt_names [(val >> SCTRL_SH_P1FMT) & 3]
+ );
+}
+#else
+#define ldebug(...)
+#define print_ctl(...)
+#define print_sctl(...)
+#endif
+
+#ifdef VERBOSE_ES1370
+#define dolog(...) AUD_log ("es1370", __VA_ARGS__)
+#else
+#define dolog(...)
+#endif
+
+#ifndef SILENT_ES1370
+#define lwarn(...) AUD_log ("es1370: warning:", __VA_ARGS__)
+#else
+#define lwarn(...)
+#endif
+
+struct chan {
+ uint32_t shift;
+ uint32_t leftover;
+ uint32_t scount;
+ uint32_t frame_addr;
+ uint32_t frame_cnt;
+};
+
+typedef struct ES1370State {
+ PCIDevice *pci_dev;
+
+ struct chan chan[NB_CHANNELS];
+ SWVoiceOut *dac_voice[2];
+ SWVoiceIn *adc_voice;
+
+ uint32_t ctl;
+ uint32_t status;
+ uint32_t mempage;
+ uint32_t codec;
+ uint32_t sctl;
+} ES1370State;
+
+typedef struct PCIES1370State {
+ PCIDevice dev;
+ ES1370State es1370;
+} PCIES1370State;
+
+struct chan_bits {
+ uint32_t ctl_en;
+ uint32_t stat_int;
+ uint32_t sctl_pause;
+ uint32_t sctl_inten;
+ uint32_t sctl_fmt;
+ uint32_t sctl_sh_fmt;
+ uint32_t sctl_loopsel;
+ void (*calc_freq) (ES1370State *s, uint32_t ctl,
+ uint32_t *old_freq, uint32_t *new_freq);
+};
+
+static void es1370_dac1_calc_freq (ES1370State *s, uint32_t ctl,
+ uint32_t *old_freq, uint32_t *new_freq);
+static void es1370_dac2_and_adc_calc_freq (ES1370State *s, uint32_t ctl,
+ uint32_t *old_freq,
+ uint32_t *new_freq);
+
+static const struct chan_bits es1370_chan_bits[] = {
+ {CTRL_DAC1_EN, STAT_DAC1, SCTRL_P1PAUSE, SCTRL_P1INTEN,
+ SCTRL_P1FMT, SCTRL_SH_P1FMT, SCTRL_P1LOOPSEL,
+ es1370_dac1_calc_freq},
+
+ {CTRL_DAC2_EN, STAT_DAC2, SCTRL_P2PAUSE, SCTRL_P2INTEN,
+ SCTRL_P2FMT, SCTRL_SH_P2FMT, SCTRL_P2LOOPSEL,
+ es1370_dac2_and_adc_calc_freq},
+
+ {CTRL_ADC_EN, STAT_ADC, 0, SCTRL_R1INTEN,
+ SCTRL_R1FMT, SCTRL_SH_R1FMT, SCTRL_R1LOOPSEL,
+ es1370_dac2_and_adc_calc_freq}
+};
+
+static void es1370_update_status (ES1370State *s, uint32_t new_status)
+{
+ uint32_t level = new_status & (STAT_DAC1 | STAT_DAC2 | STAT_ADC);
+
+ if (level) {
+ s->status = new_status | STAT_INTR;
+ }
+ else {
+ s->status = new_status & ~STAT_INTR;
+ }
+ pci_set_irq (s->pci_dev, 0, !!level);
+}
+
+static void es1370_reset (ES1370State *s)
+{
+ size_t i;
+
+ s->ctl = 1;
+ s->status = 0x60;
+ s->mempage = 0;
+ s->codec = 0;
+ s->sctl = 0;
+
+ for (i = 0; i < NB_CHANNELS; ++i) {
+ struct chan *d = &s->chan[i];
+ d->scount = 0;
+ d->leftover = 0;
+ if (i == ADC_CHANNEL) {
+ AUD_close_in (s->adc_voice);
+ s->adc_voice = NULL;
+ }
+ else {
+ AUD_close_out (s->dac_voice[i]);
+ s->dac_voice[i] = NULL;
+ }
+ }
+ pci_set_irq (s->pci_dev, 0, 0);
+}
+
+static void es1370_maybe_lower_irq (ES1370State *s, uint32_t sctl)
+{
+ uint32_t new_status = s->status;
+
+ if (!(sctl & SCTRL_P1INTEN) && (s->sctl & SCTRL_P1INTEN)) {
+ new_status &= ~STAT_DAC1;
+ }
+
+ if (!(sctl & SCTRL_P2INTEN) && (s->sctl & SCTRL_P2INTEN)) {
+ new_status &= ~STAT_DAC2;
+ }
+
+ if (!(sctl & SCTRL_R1INTEN) && (s->sctl & SCTRL_R1INTEN)) {
+ new_status &= ~STAT_ADC;
+ }
+
+ if (new_status != s->status) {
+ es1370_update_status (s, new_status);
+ }
+}
+
+static void es1370_dac1_calc_freq (ES1370State *s, uint32_t ctl,
+ uint32_t *old_freq, uint32_t *new_freq)
+
+{
+ *old_freq = dac1_samplerate[(s->ctl & CTRL_WTSRSEL) >> CTRL_SH_WTSRSEL];
+ *new_freq = dac1_samplerate[(ctl & CTRL_WTSRSEL) >> CTRL_SH_WTSRSEL];
+}
+
+static void es1370_dac2_and_adc_calc_freq (ES1370State *s, uint32_t ctl,
+ uint32_t *old_freq,
+ uint32_t *new_freq)
+
+{
+ uint32_t old_pclkdiv, new_pclkdiv;
+
+ new_pclkdiv = (ctl & CTRL_PCLKDIV) >> CTRL_SH_PCLKDIV;
+ old_pclkdiv = (s->ctl & CTRL_PCLKDIV) >> CTRL_SH_PCLKDIV;
+ *new_freq = DAC2_DIVTOSR (new_pclkdiv);
+ *old_freq = DAC2_DIVTOSR (old_pclkdiv);
+}
+
+static void es1370_update_voices (ES1370State *s, uint32_t ctl, uint32_t sctl)
+{
+ size_t i;
+ uint32_t old_freq, new_freq, old_fmt, new_fmt;
+
+ for (i = 0; i < NB_CHANNELS; ++i) {
+ struct chan *d = &s->chan[i];
+ const struct chan_bits *b = &es1370_chan_bits[i];
+
+ new_fmt = (sctl & b->sctl_fmt) >> b->sctl_sh_fmt;
+ old_fmt = (s->sctl & b->sctl_fmt) >> b->sctl_sh_fmt;
+
+ b->calc_freq (s, ctl, &old_freq, &new_freq);
+
+ if ((old_fmt != new_fmt) || (old_freq != new_freq)) {
+ d->shift = (new_fmt & 1) + (new_fmt >> 1);
+ ldebug ("channel %d, freq = %d, nchannels %d, fmt %d, shift %d\n",
+ i,
+ new_freq,
+ 1 << (new_fmt & 1),
+ (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8,
+ d->shift);
+ if (new_freq) {
+ if (i == ADC_CHANNEL) {
+ s->adc_voice =
+ AUD_open_in (
+ s->adc_voice,
+ "es1370.adc",
+ s,
+ es1370_adc_callback,
+ new_freq,
+ 1 << (new_fmt & 1),
+ (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8
+ );
+ }
+ else {
+ s->dac_voice[i] =
+ AUD_open_out (
+ s->dac_voice[i],
+ i ? "es1370.dac2" : "es1370.dac1",
+ s,
+ i ? es1370_dac2_callback : es1370_dac1_callback,
+ new_freq,
+ 1 << (new_fmt & 1),
+ (new_fmt & 2) ? AUD_FMT_S16 : AUD_FMT_U8
+ );
+ }
+ }
+ }
+
+ if (((ctl ^ s->ctl) & b->ctl_en)
+ || ((sctl ^ s->sctl) & b->sctl_pause)) {
+ int on = (ctl & b->ctl_en) && !(sctl & b->sctl_pause);
+
+ if (i == ADC_CHANNEL) {
+ AUD_set_active_in (s->adc_voice, on);
+ }
+ else {
+ AUD_set_active_out (s->dac_voice[i], on);
+ }
+ }
+ }
+
+ s->ctl = ctl;
+ s->sctl = sctl;
+}
+
+static inline uint32_t es1370_fixup (ES1370State *s, uint32_t addr)
+{
+ addr &= 0xff;
+ if (addr >= 0x30 && addr <= 0x3f)
+ addr |= s->mempage << 8;
+ return addr;
+}
+
+IO_WRITE_PROTO (es1370_writeb)
+{
+ ES1370State *s = opaque;
+ addr = es1370_fixup (s, addr);
+ uint32_t shift, mask;
+
+ switch (addr) {
+ case ES1370_REG_CONTROL:
+ case ES1370_REG_CONTROL + 1:
+ case ES1370_REG_CONTROL + 2:
+ case ES1370_REG_CONTROL + 3:
+ shift = (addr - ES1370_REG_CONTROL) << 3;
+ mask = 0xff << shift;
+ val = (s->ctl & ~mask) | ((val & 0xff) << shift);
+ es1370_update_voices (s, val, s->sctl);
+ print_ctl (val);
+ break;
+ case ES1370_REG_MEMPAGE:
+ s->mempage = val;
+ break;
+ case ES1370_REG_SERIAL_CONTROL:
+ case ES1370_REG_SERIAL_CONTROL + 1:
+ case ES1370_REG_SERIAL_CONTROL + 2:
+ case ES1370_REG_SERIAL_CONTROL + 3:
+ shift = (addr - ES1370_REG_SERIAL_CONTROL) << 3;
+ mask = 0xff << shift;
+ val = (s->sctl & ~mask) | ((val & 0xff) << shift);
+ es1370_maybe_lower_irq (s, val);
+ es1370_update_voices (s, s->ctl, val);
+ print_sctl (val);
+ break;
+ default:
+ lwarn ("writeb %#x <- %#x\n", addr, val);
+ break;
+ }
+}
+
+IO_WRITE_PROTO (es1370_writew)
+{
+ ES1370State *s = opaque;
+ addr = es1370_fixup (s, addr);
+ uint32_t shift, mask;
+ struct chan *d = &s->chan[0];
+
+ switch (addr) {
+ case ES1370_REG_CODEC:
+ dolog ("ignored codec write address %#x, data %#x\n",
+ (val >> 8) & 0xff, val & 0xff);
+ s->codec = val;
+ break;
+
+ case ES1370_REG_CONTROL:
+ case ES1370_REG_CONTROL + 2:
+ shift = (addr != ES1370_REG_CONTROL) << 4;
+ mask = 0xffff << shift;
+ val = (s->ctl & ~mask) | ((val & 0xffff) << shift);
+ es1370_update_voices (s, val, s->sctl);
+ print_ctl (val);
+ break;
+
+ case ES1370_REG_ADC_SCOUNT:
+ d++;
+ case ES1370_REG_DAC2_SCOUNT:
+ d++;
+ case ES1370_REG_DAC1_SCOUNT:
+ d->scount = (d->scount & ~0xffff) | (val & 0xffff);
+ break;
+
+ default:
+ lwarn ("writew %#x <- %#x\n", addr, val);
+ break;
+ }
+}
+
+IO_WRITE_PROTO (es1370_writel)
+{
+ ES1370State *s = opaque;
+ struct chan *d = &s->chan[0];
+
+ addr = es1370_fixup (s, addr);
+
+ switch (addr) {
+ case ES1370_REG_CONTROL:
+ es1370_update_voices (s, val, s->sctl);
+ print_ctl (val);
+ break;
+
+ case ES1370_REG_MEMPAGE:
+ s->mempage = val & 0xf;
+ break;
+
+ case ES1370_REG_SERIAL_CONTROL:
+ es1370_maybe_lower_irq (s, val);
+ es1370_update_voices (s, s->ctl, val);
+ print_sctl (val);
+ break;
+
+ case ES1370_REG_ADC_SCOUNT:
+ d++;
+ case ES1370_REG_DAC2_SCOUNT:
+ d++;
+ case ES1370_REG_DAC1_SCOUNT:
+ d->scount = (val & 0xffff) | (d->scount & ~0xffff);
+ ldebug ("chan %d CURR_SAMP_CT %d, SAMP_CT %d\n",
+ d - &s->chan[0], val >> 16, (val & 0xffff));
+ break;
+
+ case ES1370_REG_ADC_FRAMEADR:
+ d++;
+ case ES1370_REG_DAC2_FRAMEADR:
+ d++;
+ case ES1370_REG_DAC1_FRAMEADR:
+ d->frame_addr = val;
+ ldebug ("chan %d frame address %#x\n", d - &s->chan[0], val);
+ break;
+
+ case ES1370_REG_ADC_FRAMECNT:
+ d++;
+ case ES1370_REG_DAC2_FRAMECNT:
+ d++;
+ case ES1370_REG_DAC1_FRAMECNT:
+ d->frame_cnt = val;
+ d->leftover = 0;
+ ldebug ("chan %d frame count %d, buffer size %d\n",
+ d - &s->chan[0], val >> 16, val & 0xffff);
+ break;
+
+ default:
+ lwarn ("writel %#x <- %#x\n", addr, val);
+ break;
+ }
+}
+
+IO_READ_PROTO (es1370_readb)
+{
+ ES1370State *s = opaque;
+ uint32_t val;
+
+ addr = es1370_fixup (s, addr);
+
+ switch (addr) {
+ case 0x1b: /* Legacy */
+ lwarn ("Attempt to read from legacy register\n");
+ val = 5;
+ break;
+ case ES1370_REG_MEMPAGE:
+ val = s->mempage;
+ break;
+ case ES1370_REG_CONTROL + 0:
+ case ES1370_REG_CONTROL + 1:
+ case ES1370_REG_CONTROL + 2:
+ case ES1370_REG_CONTROL + 3:
+ val = s->ctl >> ((addr - ES1370_REG_CONTROL) << 3);
+ break;
+ case ES1370_REG_STATUS + 0:
+ case ES1370_REG_STATUS + 1:
+ case ES1370_REG_STATUS + 2:
+ case ES1370_REG_STATUS + 3:
+ val = s->status >> ((addr - ES1370_REG_STATUS) << 3);
+ break;
+ default:
+ val = ~0;
+ lwarn ("readb %#x -> %#x\n", addr, val);
+ break;
+ }
+ return val;
+}
+
+IO_READ_PROTO (es1370_readw)
+{
+ ES1370State *s = opaque;
+ struct chan *d = &s->chan[0];
+ uint32_t val;
+
+ addr = es1370_fixup (s, addr);
+
+ switch (addr) {
+ case ES1370_REG_ADC_SCOUNT + 2:
+ d++;
+ case ES1370_REG_DAC2_SCOUNT + 2:
+ d++;
+ case ES1370_REG_DAC1_SCOUNT + 2:
+ val = d->scount >> 16;
+ break;
+
+ default:
+ val = ~0;
+ lwarn ("readw %#x -> %#x\n", addr, val);
+ break;
+ }
+
+ return val;
+}
+
+IO_READ_PROTO (es1370_readl)
+{
+ ES1370State *s = opaque;
+ uint32_t val;
+ struct chan *d = &s->chan[0];
+
+ addr = es1370_fixup (s, addr);
+
+ switch (addr) {
+ case ES1370_REG_CONTROL:
+ val = s->ctl;
+ break;
+ case ES1370_REG_STATUS:
+ val = s->status;
+ break;
+ case ES1370_REG_MEMPAGE:
+ val = s->mempage;
+ break;
+ case ES1370_REG_CODEC:
+ val = s->codec;
+ break;
+ case ES1370_REG_SERIAL_CONTROL:
+ val = s->sctl;
+ break;
+
+ case ES1370_REG_ADC_SCOUNT:
+ d++;
+ case ES1370_REG_DAC2_SCOUNT:
+ d++;
+ case ES1370_REG_DAC1_SCOUNT:
+ val = d->scount;
+#ifdef DEBUG_ES1370
+ {
+ uint32_t curr_count = d->scount >> 16;
+ uint32_t count = d->scount & 0xffff;
+
+ curr_count <<= d->shift;
+ count <<= d->shift;
+ dolog ("read scount curr %d, total %d\n", curr_count, count);
+ }
+#endif
+ break;
+
+ case ES1370_REG_ADC_FRAMECNT:
+ d++;
+ case ES1370_REG_DAC2_FRAMECNT:
+ d++;
+ case ES1370_REG_DAC1_FRAMECNT:
+ val = d->frame_cnt;
+#ifdef DEBUG_ES1370
+ {
+ uint32_t size = ((d->frame_cnt & 0xffff) + 1) << 2;
+ uint32_t curr = ((d->frame_cnt >> 16) + 1) << 2;
+ if (curr > size)
+ dolog ("read framecnt curr %d, size %d %d\n", curr, size,
+ curr > size);
+ }
+#endif
+ break;
+
+ case ES1370_REG_ADC_FRAMEADR:
+ d++;
+ case ES1370_REG_DAC2_FRAMEADR:
+ d++;
+ case ES1370_REG_DAC1_FRAMEADR:
+ val = d->frame_addr;
+ break;
+
+ default:
+ val = ~0U;
+ lwarn ("readl %#x -> %#x\n", addr, val);
+ break;
+ }
+ return val;
+}
+
+
+static void es1370_transfer_audio (ES1370State *s, struct chan *d, int loop_sel,
+ int max, int *irq)
+{
+ uint8_t tmpbuf[4096];
+ uint32_t addr = d->frame_addr;
+ int sc = d->scount & 0xffff;
+ int csc = d->scount >> 16;
+ int csc_bytes = (csc + 1) << d->shift;
+ int cnt = d->frame_cnt >> 16;
+ int size = d->frame_cnt & 0xffff;
+ int left = ((size - cnt + 1) << 2) + d->leftover;
+ int transfered = 0;
+ int temp = audio_MIN (max, audio_MIN (left, csc_bytes));
+ int index = d - &s->chan[0];
+
+ addr += (cnt << 2) + d->leftover;
+
+ if (index == ADC_CHANNEL) {
+ while (temp) {
+ int acquired, to_copy;
+
+ to_copy = audio_MIN (temp, sizeof (tmpbuf));
+ acquired = AUD_read (s->adc_voice, tmpbuf, to_copy);
+ if (!acquired)
+ break;
+
+ cpu_physical_memory_write (addr, tmpbuf, acquired);
+
+ temp -= acquired;
+ addr += acquired;
+ transfered += acquired;
+ }
+ }
+ else {
+ SWVoiceOut *voice = s->dac_voice[index];
+
+ while (temp) {
+ int copied, to_copy;
+
+ to_copy = audio_MIN (temp, sizeof (tmpbuf));
+ cpu_physical_memory_read (addr, tmpbuf, to_copy);
+ copied = AUD_write (voice, tmpbuf, to_copy);
+ if (!copied)
+ break;
+ temp -= copied;
+ addr += copied;
+ transfered += copied;
+ }
+ }
+
+ if (csc_bytes == transfered) {
+ *irq = 1;
+ d->scount = sc | (sc << 16);
+ ldebug ("sc = %d, rate = %f\n",
+ (sc + 1) << d->shift,
+ (sc + 1) / (double) 44100);
+ }
+ else {
+ *irq = 0;
+ d->scount = sc | (((csc_bytes - transfered - 1) >> d->shift) << 16);
+ }
+
+ cnt += (transfered + d->leftover) >> 2;
+
+ if (s->sctl & loop_sel) {
+ /* Bah, how stupid is that having a 0 represent true value?
+ i just spent few hours on this shit */
+ lwarn ("whoops non looping mode\n");
+ }
+ else {
+ d->frame_cnt = size;
+
+ if (cnt <= d->frame_cnt)
+ d->frame_cnt |= cnt << 16;
+ }
+
+ d->leftover = (transfered + d->leftover) & 3;
+}
+
+static void es1370_run_channel (ES1370State *s, size_t chan, int free_or_avail)
+{
+ uint32_t new_status = s->status;
+ int max_bytes, irq;
+ struct chan *d = &s->chan[chan];
+ const struct chan_bits *b = &es1370_chan_bits[chan];
+
+ if (!(s->ctl & b->ctl_en) || (s->sctl & b->sctl_pause)) {
+ return;
+ }
+
+ max_bytes = free_or_avail;
+ max_bytes &= ~((1 << d->shift) - 1);
+ if (!max_bytes) {
+ return;
+ }
+
+ es1370_transfer_audio (s, d, b->sctl_loopsel, max_bytes, &irq);
+
+ if (irq) {
+ if (s->sctl & b->sctl_inten) {
+ new_status |= b->stat_int;
+ }
+ }
+
+ if (new_status != s->status) {
+ es1370_update_status (s, new_status);
+ }
+}
+
+static void es1370_dac1_callback (void *opaque, int free)
+{
+ ES1370State *s = opaque;
+
+ es1370_run_channel (s, DAC1_CHANNEL, free);
+}
+
+static void es1370_dac2_callback (void *opaque, int free)
+{
+ ES1370State *s = opaque;
+
+ es1370_run_channel (s, DAC2_CHANNEL, free);
+}
+
+static void es1370_adc_callback (void *opaque, int avail)
+{
+ ES1370State *s = opaque;
+
+ es1370_run_channel (s, ADC_CHANNEL, avail);
+}
+
+static void es1370_map (PCIDevice *pci_dev, int region_num,
+ uint32_t addr, uint32_t size, int type)
+{
+ PCIES1370State *d = (PCIES1370State *) pci_dev;
+ ES1370State *s = &d->es1370;
+
+ register_ioport_write (addr, 0x40 * 4, 1, es1370_writeb, s);
+ register_ioport_write (addr, 0x40 * 2, 2, es1370_writew, s);
+ register_ioport_write (addr, 0x40, 4, es1370_writel, s);
+
+ register_ioport_read (addr, 0x40 * 4, 1, es1370_readb, s);
+ register_ioport_read (addr, 0x40 * 2, 2, es1370_readw, s);
+ register_ioport_read (addr, 0x40, 4, es1370_readl, s);
+}
+
+static void es1370_save (QEMUFile *f, void *opaque)
+{
+ ES1370State *s = opaque;
+ size_t i;
+
+ for (i = 0; i < NB_CHANNELS; ++i) {
+ struct chan *d = &s->chan[i];
+ qemu_put_be32s (f, &d->shift);
+ qemu_put_be32s (f, &d->leftover);
+ qemu_put_be32s (f, &d->scount);
+ qemu_put_be32s (f, &d->frame_addr);
+ qemu_put_be32s (f, &d->frame_cnt);
+ }
+ qemu_put_be32s (f, &s->ctl);
+ qemu_put_be32s (f, &s->status);
+ qemu_put_be32s (f, &s->mempage);
+ qemu_put_be32s (f, &s->codec);
+ qemu_put_be32s (f, &s->sctl);
+}
+
+static int es1370_load (QEMUFile *f, void *opaque, int version_id)
+{
+ uint32_t ctl, sctl;
+ ES1370State *s = opaque;
+ size_t i;
+
+ if (version_id != 1)
+ return -EINVAL;
+
+ for (i = 0; i < NB_CHANNELS; ++i) {
+ struct chan *d = &s->chan[i];
+ qemu_get_be32s (f, &d->shift);
+ qemu_get_be32s (f, &d->leftover);
+ qemu_get_be32s (f, &d->scount);
+ qemu_get_be32s (f, &d->frame_addr);
+ qemu_get_be32s (f, &d->frame_cnt);
+ if (i == ADC_CHANNEL) {
+ if (s->adc_voice) {
+ AUD_close_in (s->adc_voice);
+ s->adc_voice = NULL;
+ }
+ }
+ else {
+ if (s->dac_voice[i]) {
+ AUD_close_out (s->dac_voice[i]);
+ s->dac_voice[i] = NULL;
+ }
+ }
+ }
+
+ qemu_get_be32s (f, &ctl);
+ qemu_get_be32s (f, &s->status);
+ qemu_get_be32s (f, &s->mempage);
+ qemu_get_be32s (f, &s->codec);
+ qemu_get_be32s (f, &sctl);
+
+ s->ctl = 0;
+ s->sctl = 0;
+ es1370_update_voices (s, ctl, sctl);
+ return 0;
+}
+
+static void es1370_on_reset (void *opaque)
+{
+ ES1370State *s = opaque;
+ es1370_reset (s);
+}
+
+int es1370_init (PCIBus *bus)
+{
+ PCIES1370State *d;
+ ES1370State *s;
+ uint8_t *c;
+
+ d = (PCIES1370State *) pci_register_device (bus, "ES1370",
+ sizeof (PCIES1370State),
+ -1, NULL, NULL);
+
+ if (!d) {
+ fprintf (stderr, "Failed to register PCI device for ES1370\n");
+ return -1;
+ }
+
+ c = d->dev.config;
+ c[0x00] = 0x74;
+ c[0x01] = 0x12;
+ c[0x02] = 0x00;
+ c[0x03] = 0x50;
+ c[0x07] = 2 << 1;
+ c[0x0a] = 0x01;
+ c[0x0b] = 0x04;
+
+#if 1
+ c[0x2c] = 0x42;
+ c[0x2d] = 0x49;
+ c[0x2e] = 0x4c;
+ c[0x2f] = 0x4c;
+#else
+ c[0x2c] = 0x74;
+ c[0x2d] = 0x12;
+ c[0x2e] = 0x71;
+ c[0x2f] = 0x13;
+ c[0x34] = 0xdc;
+ c[0x3c] = 10;
+ c[0xdc] = 0x00;
+#endif
+
+ c[0x3d] = 1;
+ c[0x3e] = 0x0c;
+ c[0x3f] = 0x80;
+
+ s = &d->es1370;
+ s->pci_dev = &d->dev;
+
+ pci_register_io_region (&d->dev, 0, 256, PCI_ADDRESS_SPACE_IO, es1370_map);
+ register_savevm ("es1370", 0, 1, es1370_save, es1370_load, s);
+ qemu_register_reset (es1370_on_reset, s);
+ es1370_reset (s);
+ return 0;
+}
diff --git a/hw/pc.c b/hw/pc.c
index f0ae1b964f..ad3d5e032b 100644
--- a/hw/pc.c
+++ b/hw/pc.c
@@ -602,18 +602,18 @@ static void pc_init1(int ram_size, int vga_ram_size, int boot_device,
if (audio_enabled) {
AUD_init();
-#ifdef USE_SB16
if (sb16_enabled)
SB16_init ();
-#endif
#ifdef CONFIG_ADLIB
if (adlib_enabled)
Adlib_init ();
#endif
-#ifdef USE_GUS
+#ifdef CONFIG_GUS
if (gus_enabled)
GUS_init ();
#endif
+ if (pci_enabled && es1370_enabled)
+ es1370_init (pci_bus);
}
floppy_controller = fdctrl_init(6, 2, 0, 0x3f0, fd_table);
diff --git a/hw/sb16.c b/hw/sb16.c
index bca5795eac..e79d192789 100644
--- a/hw/sb16.c
+++ b/hw/sb16.c
@@ -1,8 +1,8 @@
/*
* QEMU Soundblaster 16 emulation
- *
- * Copyright (c) 2003-2004 Vassili Karpov (malc)
- *
+ *
+ * Copyright (c) 2003-2005 Vassili Karpov (malc)
+ *
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
@@ -99,9 +99,10 @@ typedef struct SB16State {
int dma_running;
int bytes_per_second;
int align;
- SWVoice *voice;
+ int audio_free;
+ SWVoiceOut *voice;
- QEMUTimer *ts, *aux_ts;
+ QEMUTimer *aux_ts;
/* mixer state */
int mixer_nreg;
uint8_t mixer_regs[256];
@@ -110,6 +111,8 @@ typedef struct SB16State {
/* XXX: suppress that and use a context */
static struct SB16State dsp;
+static void SB_audio_callback (void *opaque, int free);
+
static int magic_of_irq (int irq)
{
switch (irq) {
@@ -174,11 +177,11 @@ static void control (SB16State *s, int hold)
if (hold) {
DMA_hold_DREQ (dma);
- AUD_enable (s->voice, 1);
+ AUD_set_active_out (s->voice, 1);
}
else {
DMA_release_DREQ (dma);
- AUD_enable (s->voice, 0);
+ AUD_set_active_out (s->voice, 0);
}
}
@@ -207,8 +210,9 @@ static void dma_cmd8 (SB16State *s, int mask, int dma_len)
s->freq = (1000000 + (tmp / 2)) / tmp;
}
- if (dma_len != -1)
+ if (dma_len != -1) {
s->block_size = dma_len << s->fmt_stereo;
+ }
else {
/* This is apparently the only way to make both Act1/PL
and SecondReality/FC work
@@ -227,17 +231,28 @@ static void dma_cmd8 (SB16State *s, int mask, int dma_len)
s->dma_auto = (mask & DMA8_AUTO) != 0;
s->align = (1 << s->fmt_stereo) - 1;
- if (s->block_size & s->align)
- dolog ("warning: unaligned buffer\n");
+ if (s->block_size & s->align) {
+ dolog ("warning: misaligned block size %d, alignment %d\n",
+ s->block_size, s->align + 1);
+ }
ldebug ("freq %d, stereo %d, sign %d, bits %d, "
"dma %d, auto %d, fifo %d, high %d\n",
s->freq, s->fmt_stereo, s->fmt_signed, s->fmt_bits,
s->block_size, s->dma_auto, s->fifo, s->highspeed);
- if (s->freq)
- s->voice = AUD_open (s->voice, "sb16", s->freq,
- 1 << s->fmt_stereo, s->fmt);
+ if (s->freq) {
+ s->audio_free = 0;
+ s->voice = AUD_open_out (
+ s->voice,
+ "sb16",
+ s,
+ SB_audio_callback,
+ s->freq,
+ 1 << s->fmt_stereo,
+ s->fmt
+ );
+ }
control (s, 1);
speaker (s, 1);
@@ -309,12 +324,23 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
s->bytes_per_second = (s->freq << s->fmt_stereo) << (s->fmt_bits == 16);
s->highspeed = 0;
s->align = (1 << (s->fmt_stereo + (s->fmt_bits == 16))) - 1;
- if (s->block_size & s->align)
- dolog ("warning: unaligned buffer\n");
+ if (s->block_size & s->align) {
+ dolog ("warning: misaligned block size %d, alignment %d\n",
+ s->block_size, s->align + 1);
+ }
- if (s->freq)
- s->voice = AUD_open (s->voice, "sb16", s->freq,
- 1 << s->fmt_stereo, s->fmt);
+ if (s->freq) {
+ s->audio_free = 0;
+ s->voice = AUD_open_out (
+ s->voice,
+ "sb16",
+ s,
+ SB_audio_callback,
+ s->freq,
+ 1 << s->fmt_stereo,
+ s->fmt
+ );
+ }
control (s, 1);
speaker (s, 1);
@@ -323,14 +349,16 @@ static void dma_cmd (SB16State *s, uint8_t cmd, uint8_t d0, int dma_len)
static inline void dsp_out_data (SB16State *s, uint8_t val)
{
ldebug ("outdata %#x\n", val);
- if (s->out_data_len < sizeof (s->out_data))
+ if (s->out_data_len < sizeof (s->out_data)) {
s->out_data[s->out_data_len++] = val;
+ }
}
static inline uint8_t dsp_get_data (SB16State *s)
{
- if (s->in_index)
+ if (s->in_index) {
return s->in2_data[--s->in_index];
+ }
else {
dolog ("buffer underflow\n");
return 0;
@@ -356,6 +384,8 @@ static void command (SB16State *s, uint8_t cmd)
s->needed_bytes = 3;
}
else {
+ s->needed_bytes = 0;
+
switch (cmd) {
case 0x03:
dsp_out_data (s, 0x10); /* s->csp_param); */
@@ -403,7 +433,7 @@ static void command (SB16State *s, uint8_t cmd)
goto warn;
case 0x35:
- dolog ("MIDI command(0x35) not implemented\n");
+ dolog ("0x35 - MIDI command not implemented\n");
break;
case 0x40:
@@ -435,6 +465,38 @@ static void command (SB16State *s, uint8_t cmd)
s->needed_bytes = 2;
break;
+ case 0x74:
+ s->needed_bytes = 2; /* DMA DAC, 4-bit ADPCM */
+ dolog ("0x75 - DMA DAC, 4-bit ADPCM not implemented\n");
+ break;
+
+ case 0x75: /* DMA DAC, 4-bit ADPCM Reference */
+ s->needed_bytes = 2;
+ dolog ("0x74 - DMA DAC, 4-bit ADPCM Reference not implemented\n");
+ break;
+
+ case 0x76: /* DMA DAC, 2.6-bit ADPCM */
+ s->needed_bytes = 2;
+ dolog ("0x74 - DMA DAC, 2.6-bit ADPCM not implemented\n");
+ break;
+
+ case 0x77: /* DMA DAC, 2.6-bit ADPCM Reference */
+ s->needed_bytes = 2;
+ dolog ("0x74 - DMA DAC, 2.6-bit ADPCM Reference not implemented\n");
+ break;
+
+ case 0x7d:
+ dolog ("0x7d - Autio-Initialize DMA DAC, 4-bit ADPCM Reference\n");
+ dolog ("not implemented\n");
+ break;
+
+ case 0x7f:
+ dolog (
+ "0x7d - Autio-Initialize DMA DAC, 2.6-bit ADPCM Reference\n"
+ );
+ dolog ("not implemented\n");
+ break;
+
case 0x80:
s->needed_bytes = 2;
break;
@@ -476,9 +538,9 @@ static void command (SB16State *s, uint8_t cmd)
s->dma_auto = 0;
break;
- case 0xe0:
+ case 0xe0: /* DSP identification */
s->needed_bytes = 1;
- goto warn;
+ break;
case 0xe1:
dsp_out_data (s, s->ver & 0xff);
@@ -503,7 +565,7 @@ static void command (SB16State *s, uint8_t cmd)
case 0xe7:
dolog ("Attempt to probe for ESS (0xe7)?\n");
- return;
+ break;
case 0xe8: /* read test reg */
dsp_out_data (s, s->test_reg);
@@ -529,21 +591,29 @@ static void command (SB16State *s, uint8_t cmd)
goto warn;
default:
- dolog ("unrecognized command %#x\n", cmd);
- return;
+ dolog ("Unrecognized command %#x\n", cmd);
+ break;
}
}
- s->cmd = cmd;
- if (!s->needed_bytes)
+ if (!s->needed_bytes) {
ldebug ("\n");
+ }
+
+ exit:
+ if (!s->needed_bytes) {
+ s->cmd = -1;
+ }
+ else {
+ s->cmd = cmd;
+ }
return;
warn:
dolog ("warning: command %#x,%d is not truly understood yet\n",
cmd, s->needed_bytes);
- s->cmd = cmd;
- return;
+ goto exit;
+
}
static uint16_t dsp_get_lohi (SB16State *s)
@@ -607,8 +677,9 @@ static void complete (SB16State *s)
s->csp_reg83[s->csp_reg83r % 4] = d0;
s->csp_reg83r += 1;
}
- else
+ else {
s->csp_regs[d1] = d0;
+ }
break;
case 0x0f:
@@ -622,8 +693,9 @@ static void complete (SB16State *s)
dsp_out_data (s, s->csp_reg83[s->csp_reg83w % 4]);
s->csp_reg83w += 1;
}
- else
+ else {
dsp_out_data (s, s->csp_regs[d0]);
+ }
break;
case 0x10:
@@ -641,8 +713,9 @@ static void complete (SB16State *s)
break;
case 0x42: /* FT2 sets output freq with this, go figure */
+#if 0
dolog ("cmd 0x42 might not do what it think it should\n");
-
+#endif
case 0x41:
s->freq = dsp_get_hilo (s);
ldebug ("set freq %d\n", s->freq);
@@ -653,6 +726,13 @@ static void complete (SB16State *s)
ldebug ("set dma block len %d\n", s->block_size);
break;
+ case 0x74:
+ case 0x75:
+ case 0x76:
+ case 0x77:
+ /* ADPCM stuff, ignore */
+ break;
+
case 0x80:
{
int freq, samples, bytes;
@@ -662,10 +742,17 @@ static void complete (SB16State *s)
samples = dsp_get_lohi (s) + 1;
bytes = samples << s->fmt_stereo << (s->fmt_bits == 16);
ticks = (bytes * ticks_per_sec) / freq;
- if (ticks < ticks_per_sec / 1024)
+ if (ticks < ticks_per_sec / 1024) {
pic_set_irq (s->irq, 1);
- else
- qemu_mod_timer (s->aux_ts, qemu_get_clock (vm_clock) + ticks);
+ }
+ else {
+ if (s->aux_ts) {
+ qemu_mod_timer (
+ s->aux_ts,
+ qemu_get_clock (vm_clock) + ticks
+ );
+ }
+ }
ldebug ("mix silence %d %d %lld\n", samples, bytes, ticks);
}
break;
@@ -674,7 +761,7 @@ static void complete (SB16State *s)
d0 = dsp_get_data (s);
s->out_data_len = 0;
ldebug ("E0 data = %#x\n", d0);
- dsp_out_data(s, ~d0);
+ dsp_out_data (s, ~d0);
break;
case 0xe2:
@@ -737,6 +824,7 @@ static void reset (SB16State *s)
s->nzero = 0;
s->highspeed = 0;
s->v2x6 = 0;
+ s->cmd = -1;
dsp_out_data(s, 0xaa);
speaker (s, 0);
@@ -761,8 +849,9 @@ static IO_WRITE_PROTO (dsp_write)
pic_set_irq (s->irq, 0);
control (s, 0);
}
- else
+ else {
reset (s);
+ }
}
s->v2x6 = 0;
break;
@@ -845,7 +934,10 @@ static IO_READ_PROTO (dsp_read)
s->last_read_byte = retval;
}
else {
- dolog ("empty output buffer\n");
+ if (s->cmd != -1) {
+ dolog ("empty output buffer for command %#x\n",
+ s->cmd);
+ }
retval = s->last_read_byte;
/* goto error; */
}
@@ -882,13 +974,14 @@ static IO_READ_PROTO (dsp_read)
goto error;
}
- if (!ack)
+ if (!ack) {
ldebug ("read %#x -> %#x\n", nport, retval);
+ }
return retval;
error:
- dolog ("WARNING dsp_read %#x error\n", nport);
+ dolog ("warning: dsp_read %#x error\n", nport);
return 0xff;
}
@@ -933,8 +1026,9 @@ static IO_WRITE_PROTO(mixer_write_datab)
SB16State *s = opaque;
ldebug ("mixer_write [%#x] <- %#x\n", s->mixer_nreg, val);
- if (s->mixer_nreg > sizeof (s->mixer_regs))
+ if (s->mixer_nreg > sizeof (s->mixer_regs)) {
return;
+ }
switch (s->mixer_nreg) {
case 0x00:
@@ -945,8 +1039,9 @@ static IO_WRITE_PROTO(mixer_write_datab)
{
int irq = irq_of_magic (val);
ldebug ("setting irq to %d (val=%#x)\n", irq, val);
- if (irq > 0)
+ if (irq > 0) {
s->irq = irq;
+ }
}
break;
@@ -956,8 +1051,12 @@ static IO_WRITE_PROTO(mixer_write_datab)
dma = lsbindex (val & 0xf);
hdma = lsbindex (val & 0xf0);
- dolog ("attempt to set DMA register 8bit %d, 16bit %d (val=%#x)\n",
- dma, hdma, val);
+ if (dma != s->dma || hdma != s->hdma) {
+ dolog (
+ "attempt to change DMA "
+ "8bit %d(%d), 16bit %d(%d) (val=%#x)\n",
+ dma, s->dma, hdma, s->hdma, val);
+ }
#if 0
s->dma = dma;
s->hdma = hdma;
@@ -971,8 +1070,9 @@ static IO_WRITE_PROTO(mixer_write_datab)
return;
default:
- if (s->mixer_nreg >= 0x80)
- dolog ("attempt to write mixer[%#x] <- %#x\n", s->mixer_nreg, val);
+ if (s->mixer_nreg >= 0x80) {
+ ldebug ("attempt to write mixer[%#x] <- %#x\n", s->mixer_nreg, val);
+ }
break;
}
@@ -989,10 +1089,14 @@ static IO_READ_PROTO(mixer_read)
{
SB16State *s = opaque;
#ifndef DEBUG_SB16_MOST
- if (s->mixer_nreg != 0x82)
-#endif
+ if (s->mixer_nreg != 0x82) {
+ ldebug ("mixer_read[%#x] -> %#x\n",
+ s->mixer_nreg, s->mixer_regs[s->mixer_nreg]);
+ }
+#else
ldebug ("mixer_read[%#x] -> %#x\n",
s->mixer_nreg, s->mixer_regs[s->mixer_nreg]);
+#endif
return s->mixer_regs[s->mixer_nreg];
}
@@ -1010,8 +1114,9 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
int to_copy, copied;
to_copy = audio_MIN (temp, left);
- if (to_copy > sizeof(tmpbuf))
+ if (to_copy > sizeof(tmpbuf)) {
to_copy = sizeof(tmpbuf);
+ }
copied = DMA_read_memory (nchan, tmpbuf, dma_pos, to_copy);
copied = AUD_write (s->voice, tmpbuf, copied);
@@ -1020,8 +1125,9 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
dma_pos = (dma_pos + copied) % dma_len;
net += copied;
- if (!copied)
+ if (!copied) {
break;
+ }
}
return net;
@@ -1030,27 +1136,28 @@ static int write_audio (SB16State *s, int nchan, int dma_pos,
static int SB_read_DMA (void *opaque, int nchan, int dma_pos, int dma_len)
{
SB16State *s = opaque;
- int free, rfree, till, copy, written, elapsed;
+ int till, copy, written, free;
if (s->left_till_irq < 0) {
s->left_till_irq = s->block_size;
}
- elapsed = AUD_calc_elapsed (s->voice);
- free = elapsed;/* AUD_get_free (s->voice); */
- rfree = free;
- free = audio_MIN (free, elapsed) & ~s->align;
-
- if ((free <= 0) || !dma_len) {
- return dma_pos;
+ if (s->voice) {
+ free = s->audio_free & ~s->align;
+ if ((free <= 0) || !dma_len) {
+ return dma_pos;
+ }
+ }
+ else {
+ free = dma_len;
}
copy = free;
till = s->left_till_irq;
#ifdef DEBUG_SB16_MOST
- dolog ("pos:%06d free:%d,%d till:%d len:%d\n",
- dma_pos, free, AUD_get_free (s->voice), till, dma_len);
+ dolog ("pos:%06d %d till:%d len:%d\n",
+ dma_pos, free, till, dma_len);
#endif
if (till <= copy) {
@@ -1082,15 +1189,13 @@ static int SB_read_DMA (void *opaque, int nchan, int dma_pos, int dma_len)
s->left_till_irq = s->block_size + s->left_till_irq;
}
- AUD_adjust (s->voice, written);
return dma_pos;
}
-void SB_timer (void *opaque)
+static void SB_audio_callback (void *opaque, int free)
{
SB16State *s = opaque;
- AUD_run ();
- qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1);
+ s->audio_free = free;
}
static void SB_save (QEMUFile *f, void *opaque)
@@ -1150,8 +1255,9 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
{
SB16State *s = opaque;
- if (version_id != 1)
+ if (version_id != 1) {
return -EINVAL;
+ }
qemu_get_be32s (f, &s->irq);
qemu_get_be32s (f, &s->dma);
@@ -1202,14 +1308,23 @@ static int SB_load (QEMUFile *f, void *opaque, int version_id)
qemu_get_buffer (f, s->mixer_regs, 256);
if (s->voice) {
- AUD_close (s->voice);
+ AUD_close_out (s->voice);
s->voice = NULL;
}
if (s->dma_running) {
- if (s->freq)
- s->voice = AUD_open (s->voice, "sb16", s->freq,
- 1 << s->fmt_stereo, s->fmt);
+ if (s->freq) {
+ s->audio_free = 0;
+ s->voice = AUD_open_out (
+ s->voice,
+ "sb16",
+ s,
+ SB_audio_callback,
+ s->freq,
+ 1 << s->fmt_stereo,
+ s->fmt
+ );
+ }
control (s, 1);
speaker (s, s->speaker);
@@ -1224,10 +1339,7 @@ void SB16_init (void)
static const uint8_t dsp_write_ports[] = {0x6, 0xc};
static const uint8_t dsp_read_ports[] = {0x6, 0xa, 0xc, 0xd, 0xe, 0xf};
- s->ts = qemu_new_timer (vm_clock, SB_timer, s);
- if (!s->ts)
- return;
-
+ s->cmd = -1;
s->irq = conf.irq;
s->dma = conf.dma;
s->hdma = conf.hdma;
@@ -1243,8 +1355,9 @@ void SB16_init (void)
reset_mixer (s);
s->aux_ts = qemu_new_timer (vm_clock, aux_timer, s);
- if (!s->aux_ts)
- return;
+ if (!s->aux_ts) {
+ dolog ("Can not create auxiliary timer\n");
+ }
for (i = 0; i < LENOFA (dsp_write_ports); i++) {
register_ioport_write (s->port + dsp_write_ports[i], 1, 1, dsp_write, s);
@@ -1263,6 +1376,5 @@ void SB16_init (void)
DMA_register_channel (s->dma, SB_read_DMA, s);
s->can_write = 1;
- qemu_mod_timer (s->ts, qemu_get_clock (vm_clock) + 1);
register_savevm ("sb16", 0, 1, SB_save, SB_load, s);
}
diff --git a/qemu-doc.texi b/qemu-doc.texi
index 526bd4b733..51875d7741 100644
--- a/qemu-doc.texi
+++ b/qemu-doc.texi
@@ -181,8 +181,23 @@ The default is @code{en-us}.
@item -enable-audio
-The SB16 emulation is disabled by default as it may give problems with
-Windows. You can enable it manually with this option.
+Will enable audio and all the sound hardware QEMU was built with.
+
+@item -audio-help
+
+Will show the audio subsystem help: list of drivers, tunable
+parameters.
+
+@item -soundhw card1,card2,...
+
+Enable audio and selected sound hardware. Use ? to print all
+available sound hardware.
+
+@example
+qemu -soundhw sb16,adlib hda
+qemu -soundhw es1370 hda
+qemu -soundhw ?
+@end example
@item -localtime
Set the real time clock to local time (the default is to UTC
diff --git a/vl.c b/vl.c
index e7cd9668f5..1ca0c347a9 100644
--- a/vl.c
+++ b/vl.c
@@ -127,10 +127,13 @@ int nb_nics;
NetDriverState nd_table[MAX_NICS];
QEMUTimer *gui_timer;
int vm_running;
+#ifdef HAS_AUDIO
int audio_enabled = 0;
-int sb16_enabled = 1;
-int adlib_enabled = 1;
-int gus_enabled = 1;
+int sb16_enabled = 0;
+int adlib_enabled = 0;
+int gus_enabled = 0;
+int es1370_enabled = 0;
+#endif
int pci_enabled = 1;
int prep_enabled = 0;
int rtc_utc = 1;
@@ -930,7 +933,7 @@ static void init_timers(void)
#ifdef _WIN32
{
int count=0;
- timerID = timeSetEvent(10, // interval (ms)
+ timerID = timeSetEvent(1, // interval (ms)
0, // resolution
host_alarm_handler, // function
(DWORD)&count, // user parameter
@@ -2837,7 +2840,12 @@ void help(void)
#ifndef _WIN32
"-k language use keyboard layout (for example \"fr\" for French)\n"
#endif
+#ifdef HAS_AUDIO
"-enable-audio enable audio support\n"
+ "-audio-help print list of audio drivers and their options\n"
+ "-soundhw c1,... comma separated list of sound card names\n"
+ " use -soundhw ? to get the list of supported sound cards\n"
+#endif
"-localtime set the real time clock to local time [default=utc]\n"
"-full-screen start in full screen\n"
#ifdef TARGET_I386
@@ -2935,7 +2943,11 @@ enum {
QEMU_OPTION_snapshot,
QEMU_OPTION_m,
QEMU_OPTION_nographic,
+#ifdef HAS_AUDIO
QEMU_OPTION_enable_audio,
+ QEMU_OPTION_audio_help,
+ QEMU_OPTION_soundhw,
+#endif
QEMU_OPTION_nics,
QEMU_OPTION_macaddr,
@@ -2998,7 +3010,11 @@ const QEMUOption qemu_options[] = {
{ "m", HAS_ARG, QEMU_OPTION_m },
{ "nographic", 0, QEMU_OPTION_nographic },
{ "k", HAS_ARG, QEMU_OPTION_k },
+#ifdef HAS_AUDIO
{ "enable-audio", 0, QEMU_OPTION_enable_audio },
+ { "audio-help", 0, QEMU_OPTION_audio_help },
+ { "soundhw", HAS_ARG, QEMU_OPTION_soundhw },
+#endif
{ "nics", HAS_ARG, QEMU_OPTION_nics},
{ "macaddr", HAS_ARG, QEMU_OPTION_macaddr},
@@ -3117,6 +3133,78 @@ void register_machines(void)
#endif
}
+#ifdef HAS_AUDIO
+static void select_soundhw (const char *optarg)
+{
+ if (*optarg == '?') {
+ show_valid_cards:
+ printf ("Valid sound card names (comma separated):\n");
+ printf ("sb16 Creative Sound Blaster 16\n");
+#ifdef CONFIG_ADLIB
+#ifdef HAS_YMF262
+ printf ("adlib Ymaha YMF262 (OPL3)\n");
+#else
+ printf ("adlib Ymaha YM3812 (OPL2)\n");
+#endif
+#endif
+#ifdef CONFIG_GUS
+ printf ("gus Gravis Ultrasound GF1\n");
+#endif
+ printf ("es1370 ENSONIQ AudioPCI ES1370\n");
+ exit (*optarg != '?');
+ }
+ else {
+ struct {
+ char *name;
+ int *enabledp;
+ } soundhw_tab[] = {
+ { "sb16", &sb16_enabled },
+#ifdef CONFIG_ADLIB
+ { "adlib", &adlib_enabled },
+#endif
+#ifdef CONFIG_GUS
+ { "gus", &gus_enabled },
+#endif
+ { "es1370", &es1370_enabled },
+ };
+ size_t tablen, l, i;
+ const char *p;
+ char *e;
+ int bad_card = 0;
+
+ p = optarg;
+ tablen = sizeof (soundhw_tab) / sizeof (soundhw_tab[0]);
+
+ while (*p) {
+ e = strchr (p, ',');
+ l = !e ? strlen (p) : (size_t) (e - p);
+ for (i = 0; i < tablen; ++i) {
+ if (!strncmp (soundhw_tab[i].name, p, l)) {
+ audio_enabled = 1;
+ *soundhw_tab[i].enabledp = 1;
+ break;
+ }
+ }
+ if (i == tablen) {
+ if (l > 80) {
+ fprintf (stderr,
+ "Unknown sound card name (too big to show)\n");
+ }
+ else {
+ fprintf (stderr, "Unknown sound card name `%.*s'\n",
+ (int) l, p);
+ }
+ bad_card = 1;
+ }
+ p += l + (e != NULL);
+ }
+
+ if (bad_card)
+ goto show_valid_cards;
+ }
+}
+#endif
+
#define NET_IF_TUN 0
#define NET_IF_USER 1
#define NET_IF_DUMMY 2
@@ -3401,9 +3489,22 @@ int main(int argc, char **argv)
case QEMU_OPTION_dummy_net:
net_if_type = NET_IF_DUMMY;
break;
+#ifdef HAS_AUDIO
case QEMU_OPTION_enable_audio:
audio_enabled = 1;
+ sb16_enabled = 1;
+ adlib_enabled = 1;
+ gus_enabled = 1;
+ es1370_enabled = 1;
break;
+ case QEMU_OPTION_audio_help:
+ AUD_help ();
+ exit (0);
+ break;
+ case QEMU_OPTION_soundhw:
+ select_soundhw (optarg);
+ break;
+#endif
case QEMU_OPTION_h:
help();
break;
diff --git a/vl.h b/vl.h
index 31d05b6b58..543d57972a 100644
--- a/vl.h
+++ b/vl.h
@@ -124,6 +124,7 @@ extern int audio_enabled;
extern int sb16_enabled;
extern int adlib_enabled;
extern int gus_enabled;
+extern int es1370_enabled;
extern int ram_size;
extern int bios_size;
extern int rtc_utc;
@@ -628,6 +629,9 @@ void pci_piix3_ide_init(PCIBus *bus, BlockDriverState **hd_table);
int pmac_ide_init (BlockDriverState **hd_table,
SetIRQFunc *set_irq, void *irq_opaque, int irq);
+/* es1370.c */
+int es1370_init (PCIBus *bus);
+
/* sb16.c */
void SB16_init (void);
generated by cgit v1.2.3 (git 2.26.2) at 2024-11-25 20:21:24 +0000