diff options
Diffstat (limited to 'audio')
| -rw-r--r-- | audio/alsaaudio.c | 88 | ||||
| -rw-r--r-- | audio/audio.c | 5 | ||||
| -rw-r--r-- | audio/audio_int.h | 1 | ||||
| -rw-r--r-- | audio/audio_pt_int.c | 149 | ||||
| -rw-r--r-- | audio/audio_pt_int.h | 22 | ||||
| -rw-r--r-- | audio/dsound_template.h | 9 | ||||
| -rw-r--r-- | audio/dsoundaudio.c | 20 | ||||
| -rw-r--r-- | audio/esdaudio.c | 591 | ||||
| -rw-r--r-- | audio/ossaudio.c | 2 | ||||
| -rw-r--r-- | audio/wavaudio.c | 2 |
10 files changed, 837 insertions, 52 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 77a08a1c58..43cfa258d7 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -86,9 +86,9 @@ static struct { }; struct alsa_params_req { - unsigned int freq; - audfmt_e fmt; - unsigned int nchannels; + int freq; + snd_pcm_format_t fmt; + int nchannels; unsigned int buffer_size; unsigned int period_size; }; @@ -96,6 +96,7 @@ struct alsa_params_req { struct alsa_params_obt { int freq; audfmt_e fmt; + int endianness; int nchannels; snd_pcm_uframes_t samples; }; @@ -143,7 +144,7 @@ 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) +static snd_pcm_format_t aud_to_alsafmt (audfmt_e fmt) { switch (fmt) { case AUD_FMT_S8: @@ -173,7 +174,8 @@ static int aud_to_alsafmt (audfmt_e fmt) } } -static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) +static int alsa_to_audfmt (snd_pcm_format_t alsafmt, audfmt_e *fmt, + int *endianness) { switch (alsafmt) { case SND_PCM_FORMAT_S8: @@ -234,7 +236,6 @@ static int alsa_to_audfmt (int alsafmt, audfmt_e *fmt, int *endianness) return 0; } -#if defined DEBUG_MISMATCHES || defined DEBUG static void alsa_dump_info (struct alsa_params_req *req, struct alsa_params_obt *obt) { @@ -248,7 +249,6 @@ static void alsa_dump_info (struct alsa_params_req *req, req->buffer_size, req->period_size); dolog ("obtained: samples %ld\n", obt->samples); } -#endif static void alsa_set_threshold (snd_pcm_t *handle, snd_pcm_uframes_t threshold) { @@ -291,6 +291,7 @@ static int alsa_open (int in, struct alsa_params_req *req, unsigned int period_size, buffer_size; snd_pcm_uframes_t obt_buffer_size; const char *typ = in ? "ADC" : "DAC"; + snd_pcm_format_t obtfmt; freq = req->freq; period_size = req->period_size; @@ -327,9 +328,8 @@ static int alsa_open (int in, struct alsa_params_req *req, } err = snd_pcm_hw_params_set_format (handle, hw_params, req->fmt); - if (err < 0) { + if (err < 0 && conf.verbose) { 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); @@ -494,6 +494,17 @@ static int alsa_open (int in, struct alsa_params_req *req, goto err; } + err = snd_pcm_hw_params_get_format (hw_params, &obtfmt); + if (err < 0) { + alsa_logerr2 (err, typ, "Failed to get format\n"); + goto err; + } + + if (alsa_to_audfmt (obtfmt, &obt->fmt, &obt->endianness)) { + dolog ("Invalid format was returned %d\n", obtfmt); + goto err; + } + err = snd_pcm_prepare (handle); if (err < 0) { alsa_logerr2 (err, typ, "Could not prepare handle %p\n", handle); @@ -504,28 +515,41 @@ static int alsa_open (int in, struct alsa_params_req *req, 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); + bytes_per_sec = freq << (nchannels == 2); + + switch (obt->fmt) { + case AUD_FMT_S8: + case AUD_FMT_U8: + break; + + case AUD_FMT_S16: + case AUD_FMT_U16: + bytes_per_sec <<= 1; + break; + + case AUD_FMT_S32: + case AUD_FMT_U32: + bytes_per_sec <<= 2; + break; + } threshold = (conf.threshold * bytes_per_sec) / 1000; alsa_set_threshold (handle, threshold); } - obt->fmt = req->fmt; obt->nchannels = nchannels; obt->freq = freq; obt->samples = obt_buffer_size; + *handlep = handle; -#if defined DEBUG_MISMATCHES || defined DEBUG - if (obt->fmt != req->fmt || - obt->nchannels != req->nchannels || - obt->freq != req->freq) { - dolog ("Audio paramters mismatch for %s\n", typ); + if (conf.verbose && + (obt->fmt != req->fmt || + obt->nchannels != req->nchannels || + obt->freq != req->freq)) { + dolog ("Audio paramters for %s\n", typ); alsa_dump_info (req, obt); } -#endif #ifdef DEBUG alsa_dump_info (req, obt); @@ -665,9 +689,6 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) 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; audsettings_t obt_as; @@ -681,16 +702,10 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as) return -1; } - err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); - if (err) { - alsa_anal_close (&handle); - return -1; - } - obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; - obt_as.fmt = effective_fmt; - obt_as.endianness = endianness; + obt_as.fmt = obt.fmt; + obt_as.endianness = obt.endianness; audio_pcm_init_info (&hw->info, &obt_as); hw->samples = obt.samples; @@ -751,9 +766,6 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) 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; audsettings_t obt_as; @@ -767,16 +779,10 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as) return -1; } - err = alsa_to_audfmt (obt.fmt, &effective_fmt, &endianness); - if (err) { - alsa_anal_close (&handle); - return -1; - } - obt_as.freq = obt.freq; obt_as.nchannels = obt.nchannels; - obt_as.fmt = effective_fmt; - obt_as.endianness = endianness; + obt_as.fmt = obt.fmt; + obt_as.endianness = obt.endianness; audio_pcm_init_info (&hw->info, &obt_as); hw->samples = obt.samples; diff --git a/audio/audio.c b/audio/audio.c index 5e9d88bb1a..2bea53efa8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -56,6 +56,9 @@ static struct audio_driver *drvtab[] = { #ifdef CONFIG_SDL &sdl_audio_driver, #endif +#ifdef CONFIG_ESD + &esd_audio_driver, +#endif &no_audio_driver, &wav_audio_driver }; @@ -414,7 +417,7 @@ static void audio_print_options (const char *prefix, { audfmt_e *fmtp = opt->valp; printf ( - "format, %s = %s, (one of: U8 S8 U16 S16)\n", + "format, %s = %s, (one of: U8 S8 U16 S16 U32 S32)\n", state, audio_audfmt_to_string (*fmtp) ); diff --git a/audio/audio_int.h b/audio/audio_int.h index cd22a30880..f969602016 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -202,6 +202,7 @@ 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 struct audio_driver esd_audio_driver; extern volume_t nominal_volume; void audio_pcm_init_info (struct audio_pcm_info *info, audsettings_t *as); diff --git a/audio/audio_pt_int.c b/audio/audio_pt_int.c new file mode 100644 index 0000000000..e889a983b1 --- /dev/null +++ b/audio/audio_pt_int.c @@ -0,0 +1,149 @@ +#include "qemu-common.h" +#include "audio.h" + +#define AUDIO_CAP "audio-pt" + +#include "audio_int.h" +#include "audio_pt_int.h" + +static void logerr (struct audio_pt *pt, int err, const char *fmt, ...) +{ + va_list ap; + + va_start (ap, fmt); + AUD_vlog (pt->drv, fmt, ap); + va_end (ap); + + AUD_log (NULL, "\n"); + AUD_log (pt->drv, "Reason: %s\n", strerror (err)); +} + +int audio_pt_init (struct audio_pt *p, void *(*func) (void *), + void *opaque, const char *drv, const char *cap) +{ + int err, err2; + const char *efunc; + + p->drv = drv; + + err = pthread_mutex_init (&p->mutex, NULL); + if (err) { + efunc = "pthread_mutex_init"; + goto err0; + } + + err = pthread_cond_init (&p->cond, NULL); + if (err) { + efunc = "pthread_cond_init"; + goto err1; + } + + err = pthread_create (&p->thread, NULL, func, opaque); + if (err) { + efunc = "pthread_create"; + goto err2; + } + + return 0; + + err2: + err2 = pthread_cond_destroy (&p->cond); + if (err2) { + logerr (p, err2, "%s(%s): pthread_cond_destroy failed", cap, AUDIO_FUNC); + } + + err1: + err2 = pthread_mutex_destroy (&p->mutex); + if (err2) { + logerr (p, err2, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_FUNC); + } + + err0: + logerr (p, err, "%s(%s): %s failed", cap, AUDIO_FUNC, efunc); + return -1; +} + +int audio_pt_fini (struct audio_pt *p, const char *cap) +{ + int err, ret = 0; + + err = pthread_cond_destroy (&p->cond); + if (err) { + logerr (p, err, "%s(%s): pthread_cond_destroy failed", cap, AUDIO_FUNC); + ret = -1; + } + + err = pthread_mutex_destroy (&p->mutex); + if (err) { + logerr (p, err, "%s(%s): pthread_mutex_destroy failed", cap, AUDIO_FUNC); + ret = -1; + } + return ret; +} + +int audio_pt_lock (struct audio_pt *p, const char *cap) +{ + int err; + + err = pthread_mutex_lock (&p->mutex); + if (err) { + logerr (p, err, "%s(%s): pthread_mutex_lock failed", cap, AUDIO_FUNC); + return -1; + } + return 0; +} + +int audio_pt_unlock (struct audio_pt *p, const char *cap) +{ + int err; + + err = pthread_mutex_unlock (&p->mutex); + if (err) { + logerr (p, err, "%s(%s): pthread_mutex_unlock failed", cap, AUDIO_FUNC); + return -1; + } + return 0; +} + +int audio_pt_wait (struct audio_pt *p, const char *cap) +{ + int err; + + err = pthread_cond_wait (&p->cond, &p->mutex); + if (err) { + logerr (p, err, "%s(%s): pthread_cond_wait failed", cap, AUDIO_FUNC); + return -1; + } + return 0; +} + +int audio_pt_unlock_and_signal (struct audio_pt *p, const char *cap) +{ + int err; + + err = pthread_mutex_unlock (&p->mutex); + if (err) { + logerr (p, err, "%s(%s): pthread_mutex_unlock failed", cap, AUDIO_FUNC); + return -1; + } + err = pthread_cond_signal (&p->cond); + if (err) { + logerr (p, err, "%s(%s): pthread_cond_signal failed", cap, AUDIO_FUNC); + return -1; + } + return 0; +} + +int audio_pt_join (struct audio_pt *p, void **arg, const char *cap) +{ + int err; + void *ret; + + err = pthread_join (p->thread, &ret); + if (err) { + logerr (p, err, "%s(%s): pthread_join failed", cap, AUDIO_FUNC); + return -1; + } + *arg = ret; + return 0; +} diff --git a/audio/audio_pt_int.h b/audio/audio_pt_int.h new file mode 100644 index 0000000000..0dfff76aa3 --- /dev/null +++ b/audio/audio_pt_int.h @@ -0,0 +1,22 @@ +#ifndef QEMU_AUDIO_PT_INT_H +#define QEMU_AUDIO_PT_INT_H + +#include <pthread.h> + +struct audio_pt { + const char *drv; + pthread_t thread; + pthread_cond_t cond; + pthread_mutex_t mutex; +}; + +int audio_pt_init (struct audio_pt *, void *(*) (void *), void *, + const char *, const char *); +int audio_pt_fini (struct audio_pt *, const char *); +int audio_pt_lock (struct audio_pt *, const char *); +int audio_pt_unlock (struct audio_pt *, const char *); +int audio_pt_wait (struct audio_pt *, const char *); +int audio_pt_unlock_and_signal (struct audio_pt *, const char *); +int audio_pt_join (struct audio_pt *, void **, const char *); + +#endif /* audio_pt_int.h */ diff --git a/audio/dsound_template.h b/audio/dsound_template.h index 0896b04a03..d517c90ce2 100644 --- a/audio/dsound_template.h +++ b/audio/dsound_template.h @@ -23,16 +23,20 @@ */ #ifdef DSBTYPE_IN #define NAME "capture buffer" +#define NAME2 "DirectSoundCapture" #define TYPE in #define IFACE IDirectSoundCaptureBuffer #define BUFPTR LPDIRECTSOUNDCAPTUREBUFFER #define FIELD dsound_capture_buffer +#define FIELD2 dsound_capture #else #define NAME "playback buffer" +#define NAME2 "DirectSound" #define TYPE out #define IFACE IDirectSoundBuffer #define BUFPTR LPDIRECTSOUNDBUFFER #define FIELD dsound_buffer +#define FIELD2 dsound #endif static int glue (dsound_unlock_, TYPE) ( @@ -192,6 +196,11 @@ static int dsound_init_out (HWVoiceOut *hw, audsettings_t *as) DSBCAPS bc; #endif + if (!s->FIELD2) { + dsound_logerr ("Attempt to initialize voice without " NAME2 " object"); + return -1; + } + err = waveformat_from_audio_settings (&wfx, as); if (err) { return -1; diff --git a/audio/dsoundaudio.c b/audio/dsoundaudio.c index 41ddf4a18c..cba8c80030 100644 --- a/audio/dsoundaudio.c +++ b/audio/dsoundaudio.c @@ -320,23 +320,22 @@ static int waveformat_from_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as) switch (as->fmt) { case AUD_FMT_S8: - wfx->wBitsPerSample = 8; - break; - case AUD_FMT_U8: wfx->wBitsPerSample = 8; break; case AUD_FMT_S16: + case AUD_FMT_U16: wfx->wBitsPerSample = 16; wfx->nAvgBytesPerSec <<= 1; wfx->nBlockAlign <<= 1; break; - case AUD_FMT_U16: - wfx->wBitsPerSample = 16; - wfx->nAvgBytesPerSec <<= 1; - wfx->nBlockAlign <<= 1; + case AUD_FMT_S32: + case AUD_FMT_U32: + wfx->wBitsPerSample = 32; + wfx->nAvgBytesPerSec <<= 2; + wfx->nBlockAlign <<= 2; break; default: @@ -387,8 +386,13 @@ static int waveformat_to_audio_settings (WAVEFORMATEX *wfx, audsettings_t *as) as->fmt = AUD_FMT_S16; break; + case 32: + as->fmt = AUD_FMT_S32; + break; + default: - dolog ("Invalid wave format, bits per sample is not 8 or 16, but %d\n", + dolog ("Invalid wave format, bits per sample is not " + "8, 16 or 32, but %d\n", wfx->wBitsPerSample); return -1; } diff --git a/audio/esdaudio.c b/audio/esdaudio.c new file mode 100644 index 0000000000..fa42348849 --- /dev/null +++ b/audio/esdaudio.c @@ -0,0 +1,591 @@ +/* + * QEMU ESD audio driver + * + * Copyright (c) 2006 Frederick Reeve (brushed up by 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 <esd.h> +#include "qemu-common.h" +#include "audio.h" +#include <signal.h> + +#define AUDIO_CAP "esd" +#include "audio_int.h" +#include "audio_pt_int.h" + +typedef struct { + HWVoiceOut hw; + int done; + int live; + int decr; + int rpos; + void *pcm_buf; + int fd; + struct audio_pt pt; +} ESDVoiceOut; + +typedef struct { + HWVoiceIn hw; + int done; + int dead; + int incr; + int wpos; + void *pcm_buf; + int fd; + struct audio_pt pt; +} ESDVoiceIn; + +static struct { + int samples; + int divisor; + char *dac_host; + char *adc_host; +} conf = { + 1024, + 2, + NULL, + NULL +}; + +static void GCC_FMT_ATTR (2, 3) qesd_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", strerror (err)); +} + +/* playback */ +static void *qesd_thread_out (void *arg) +{ + ESDVoiceOut *esd = arg; + HWVoiceOut *hw = &esd->hw; + int threshold; + + threshold = conf.divisor ? hw->samples / conf.divisor : 0; + + for (;;) { + int decr, to_mix, rpos; + + for (;;) { + if (esd->done) { + goto exit; + } + + if (esd->live > threshold) { + break; + } + + if (audio_pt_wait (&esd->pt, AUDIO_FUNC)) { + goto exit; + } + } + + decr = to_mix = esd->live; + rpos = hw->rpos; + + if (audio_pt_unlock (&esd->pt, AUDIO_FUNC)) { + return NULL; + } + + while (to_mix) { + ssize_t written; + int chunk = audio_MIN (to_mix, hw->samples - rpos); + st_sample_t *src = hw->mix_buf + rpos; + + hw->clip (esd->pcm_buf, src, chunk); + + again: + written = write (esd->fd, esd->pcm_buf, chunk << hw->info.shift); + if (written == -1) { + if (errno == EINTR || errno == EAGAIN) { + goto again; + } + qesd_logerr (errno, "write failed\n"); + return NULL; + } + + if (written != chunk << hw->info.shift) { + int wsamples = written >> hw->info.shift; + int wbytes = wsamples << hw->info.shift; + if (wbytes != written) { + dolog ("warning: Misaligned write %d (requested %d), " + "alignment %d\n", + wbytes, written, hw->info.align + 1); + } + to_mix -= wsamples; + rpos = (rpos + wsamples) % hw->samples; + break; + } + + rpos = (rpos + chunk) % hw->samples; + to_mix -= chunk; + } + + if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) { + return NULL; + } + + esd->rpos = rpos; + esd->live -= decr; + esd->decr += decr; + } + + exit: + audio_pt_unlock (&esd->pt, AUDIO_FUNC); + return NULL; +} + +static int qesd_run_out (HWVoiceOut *hw) +{ + int live, decr; + ESDVoiceOut *esd = (ESDVoiceOut *) hw; + + if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) { + return 0; + } + + live = audio_pcm_hw_get_live_out (hw); + decr = audio_MIN (live, esd->decr); + esd->decr -= decr; + esd->live = live - decr; + hw->rpos = esd->rpos; + if (esd->live > 0) { + audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC); + } + else { + audio_pt_unlock (&esd->pt, AUDIO_FUNC); + } + return decr; +} + +static int qesd_write (SWVoiceOut *sw, void *buf, int len) +{ + return audio_pcm_sw_write (sw, buf, len); +} + +static int qesd_init_out (HWVoiceOut *hw, audsettings_t *as) +{ + ESDVoiceOut *esd = (ESDVoiceOut *) hw; + audsettings_t obt_as = *as; + int esdfmt = ESD_STREAM | ESD_PLAY; + int err; + sigset_t set, old_set; + + sigfillset (&set); + + esdfmt |= (as->nchannels == 2) ? ESD_STEREO : ESD_MONO; + switch (as->fmt) { + case AUD_FMT_S8: + case AUD_FMT_U8: + esdfmt |= ESD_BITS8; + obt_as.fmt = AUD_FMT_U8; + break; + + case AUD_FMT_S32: + case AUD_FMT_U32: + dolog ("Will use 16 instead of 32 bit samples\n"); + + case AUD_FMT_S16: + case AUD_FMT_U16: + deffmt: + esdfmt |= ESD_BITS16; + obt_as.fmt = AUD_FMT_S16; + break; + + default: + dolog ("Internal logic error: Bad audio format %d\n", as->fmt); +#ifdef DEBUG_FMOD + abort (); +#endif + goto deffmt; + + } + obt_as.endianness = 0; + + audio_pcm_init_info (&hw->info, &obt_as); + + hw->samples = conf.samples; + esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); + if (!esd->pcm_buf) { + dolog ("Could not allocate buffer (%d bytes)\n", + hw->samples << hw->info.shift); + return -1; + } + + esd->fd = -1; + err = pthread_sigmask (SIG_BLOCK, &set, &old_set); + if (err) { + qesd_logerr (err, "pthread_sigmask failed\n"); + goto fail1; + } + + esd->fd = esd_play_stream (esdfmt, as->freq, conf.dac_host, NULL); + if (esd->fd < 0) { + qesd_logerr (errno, "esd_play_stream failed\n"); + goto fail2; + } + + if (audio_pt_init (&esd->pt, qesd_thread_out, esd, AUDIO_CAP, AUDIO_FUNC)) { + goto fail3; + } + + err = pthread_sigmask (SIG_SETMASK, &old_set, NULL); + if (err) { + qesd_logerr (err, "pthread_sigmask(restore) failed\n"); + } + + return 0; + + fail3: + if (close (esd->fd)) { + qesd_logerr (errno, "%s: close on esd socket(%d) failed\n", + AUDIO_FUNC, esd->fd); + } + esd->fd = -1; + + fail2: + err = pthread_sigmask (SIG_SETMASK, &old_set, NULL); + if (err) { + qesd_logerr (err, "pthread_sigmask(restore) failed\n"); + } + + fail1: + qemu_free (esd->pcm_buf); + esd->pcm_buf = NULL; + return -1; +} + +static void qesd_fini_out (HWVoiceOut *hw) +{ + void *ret; + ESDVoiceOut *esd = (ESDVoiceOut *) hw; + + audio_pt_lock (&esd->pt, AUDIO_FUNC); + esd->done = 1; + audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC); + audio_pt_join (&esd->pt, &ret, AUDIO_FUNC); + + if (esd->fd >= 0) { + if (close (esd->fd)) { + qesd_logerr (errno, "failed to close esd socket\n"); + } + esd->fd = -1; + } + + audio_pt_fini (&esd->pt, AUDIO_FUNC); + + qemu_free (esd->pcm_buf); + esd->pcm_buf = NULL; +} + +static int qesd_ctl_out (HWVoiceOut *hw, int cmd, ...) +{ + (void) hw; + (void) cmd; + return 0; +} + +/* capture */ +static void *qesd_thread_in (void *arg) +{ + ESDVoiceIn *esd = arg; + HWVoiceIn *hw = &esd->hw; + int threshold; + + threshold = conf.divisor ? hw->samples / conf.divisor : 0; + + for (;;) { + int incr, to_grab, wpos; + + for (;;) { + if (esd->done) { + goto exit; + } + + if (esd->dead > threshold) { + break; + } + + if (audio_pt_wait (&esd->pt, AUDIO_FUNC)) { + goto exit; + } + } + + incr = to_grab = esd->dead; + wpos = hw->wpos; + + if (audio_pt_unlock (&esd->pt, AUDIO_FUNC)) { + return NULL; + } + + while (to_grab) { + ssize_t nread; + int chunk = audio_MIN (to_grab, hw->samples - wpos); + void *buf = advance (esd->pcm_buf, wpos); + + again: + nread = read (esd->fd, buf, chunk << hw->info.shift); + if (nread == -1) { + if (errno == EINTR || errno == EAGAIN) { + goto again; + } + qesd_logerr (errno, "read failed\n"); + return NULL; + } + + if (nread != chunk << hw->info.shift) { + int rsamples = nread >> hw->info.shift; + int rbytes = rsamples << hw->info.shift; + if (rbytes != nread) { + dolog ("warning: Misaligned write %d (requested %d), " + "alignment %d\n", + rbytes, nread, hw->info.align + 1); + } + to_grab -= rsamples; + wpos = (wpos + rsamples) % hw->samples; + break; + } + + hw->conv (hw->conv_buf + wpos, buf, nread >> hw->info.shift, + &nominal_volume); + wpos = (wpos + chunk) % hw->samples; + to_grab -= chunk; + } + + if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) { + return NULL; + } + + esd->wpos = wpos; + esd->dead -= incr; + esd->incr += incr; + } + + exit: + audio_pt_unlock (&esd->pt, AUDIO_FUNC); + return NULL; +} + +static int qesd_run_in (HWVoiceIn *hw) +{ + int live, incr, dead; + ESDVoiceIn *esd = (ESDVoiceIn *) hw; + + if (audio_pt_lock (&esd->pt, AUDIO_FUNC)) { + return 0; + } + + live = audio_pcm_hw_get_live_in (hw); + dead = hw->samples - live; + incr = audio_MIN (dead, esd->incr); + esd->incr -= incr; + esd->dead = dead - incr; + hw->wpos = esd->wpos; + if (esd->dead > 0) { + audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC); + } + else { + audio_pt_unlock (&esd->pt, AUDIO_FUNC); + } + return incr; +} + +static int qesd_read (SWVoiceIn *sw, void *buf, int len) +{ + return audio_pcm_sw_read (sw, buf, len); +} + +static int qesd_init_in (HWVoiceIn *hw, audsettings_t *as) +{ + ESDVoiceIn *esd = (ESDVoiceIn *) hw; + audsettings_t obt_as = *as; + int esdfmt = ESD_STREAM | ESD_RECORD; + int err; + sigset_t set, old_set; + + sigfillset (&set); + + esdfmt |= (as->nchannels == 2) ? ESD_STEREO : ESD_MONO; + switch (as->fmt) { + case AUD_FMT_S8: + case AUD_FMT_U8: + esdfmt |= ESD_BITS8; + obt_as.fmt = AUD_FMT_U8; + break; + + case AUD_FMT_S16: + case AUD_FMT_U16: + esdfmt |= ESD_BITS16; + obt_as.fmt = AUD_FMT_S16; + break; + + case AUD_FMT_S32: + case AUD_FMT_U32: + dolog ("Will use 16 instead of 32 bit samples\n"); + esdfmt |= ESD_BITS16; + obt_as.fmt = AUD_FMT_S16; + break; + } + obt_as.endianness = 0; + + audio_pcm_init_info (&hw->info, &obt_as); + + hw->samples = conf.samples; + esd->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift); + if (!esd->pcm_buf) { + dolog ("Could not allocate buffer (%d bytes)\n", + hw->samples << hw->info.shift); + return -1; + } + + esd->fd = -1; + + err = pthread_sigmask (SIG_BLOCK, &set, &old_set); + if (err) { + qesd_logerr (err, "pthread_sigmask failed\n"); + goto fail1; + } + + esd->fd = esd_record_stream (esdfmt, as->freq, conf.adc_host, NULL); + if (esd->fd < 0) { + qesd_logerr (errno, "esd_record_stream failed\n"); + goto fail2; + } + + if (audio_pt_init (&esd->pt, qesd_thread_in, esd, AUDIO_CAP, AUDIO_FUNC)) { + goto fail3; + } + + err = pthread_sigmask (SIG_SETMASK, &old_set, NULL); + if (err) { + qesd_logerr (err, "pthread_sigmask(restore) failed\n"); + } + + return 0; + + fail3: + if (close (esd->fd)) { + qesd_logerr (errno, "%s: close on esd socket(%d) failed\n", + AUDIO_FUNC, esd->fd); + } + esd->fd = -1; + + fail2: + err = pthread_sigmask (SIG_SETMASK, &old_set, NULL); + if (err) { + qesd_logerr (err, "pthread_sigmask(restore) failed\n"); + } + + fail1: + qemu_free (esd->pcm_buf); + esd->pcm_buf = NULL; + return -1; +} + +static void qesd_fini_in (HWVoiceIn *hw) +{ + void *ret; + ESDVoiceIn *esd = (ESDVoiceIn *) hw; + + audio_pt_lock (&esd->pt, AUDIO_FUNC); + esd->done = 1; + audio_pt_unlock_and_signal (&esd->pt, AUDIO_FUNC); + audio_pt_join (&esd->pt, &ret, AUDIO_FUNC); + + if (esd->fd >= 0) { + if (close (esd->fd)) { + qesd_logerr (errno, "failed to close esd socket\n"); + } + esd->fd = -1; + } + + audio_pt_fini (&esd->pt, AUDIO_FUNC); + + qemu_free (esd->pcm_buf); + esd->pcm_buf = NULL; +} + +static int qesd_ctl_in (HWVoiceIn *hw, int cmd, ...) +{ + (void) hw; + (void) cmd; + return 0; +} + +/* common */ +static void *qesd_audio_init (void) +{ + return &conf; +} + +static void qesd_audio_fini (void *opaque) +{ + (void) opaque; + ldebug ("esd_fini"); +} + +struct audio_option qesd_options[] = { + {"SAMPLES", AUD_OPT_INT, &conf.samples, + "buffer size in samples", NULL, 0}, + + {"DIVISOR", AUD_OPT_INT, &conf.divisor, + "threshold divisor", NULL, 0}, + + {"DAC_HOST", AUD_OPT_STR, &conf.dac_host, + "playback host", NULL, 0}, + + {"ADC_HOST", AUD_OPT_STR, &conf.adc_host, + "capture host", NULL, 0}, + + {NULL, 0, NULL, NULL, NULL, 0} +}; + +struct audio_pcm_ops qesd_pcm_ops = { + qesd_init_out, + qesd_fini_out, + qesd_run_out, + qesd_write, + qesd_ctl_out, + + qesd_init_in, + qesd_fini_in, + qesd_run_in, + qesd_read, + qesd_ctl_in, +}; + +struct audio_driver esd_audio_driver = { + INIT_FIELD (name = ) "esd", + INIT_FIELD (descr = ) + "http://en.wikipedia.org/wiki/Esound", + INIT_FIELD (options = ) qesd_options, + INIT_FIELD (init = ) qesd_audio_init, + INIT_FIELD (fini = ) qesd_audio_fini, + INIT_FIELD (pcm_ops = ) &qesd_pcm_ops, + INIT_FIELD (can_be_default = ) 0, + INIT_FIELD (max_voices_out = ) INT_MAX, + INIT_FIELD (max_voices_in = ) INT_MAX, + INIT_FIELD (voice_size_out = ) sizeof (ESDVoiceOut), + INIT_FIELD (voice_size_in = ) sizeof (ESDVoiceIn) +}; diff --git a/audio/ossaudio.c b/audio/ossaudio.c index 5a91556cc0..2a300c21de 100644 --- a/audio/ossaudio.c +++ b/audio/ossaudio.c @@ -150,7 +150,7 @@ static int oss_to_audfmt (int ossfmt, audfmt_e *fmt, int *endianness) { switch (ossfmt) { case AFMT_S8: - *endianness =0; + *endianness = 0; *fmt = AUD_FMT_S8; break; diff --git a/audio/wavaudio.c b/audio/wavaudio.c index 11ca86ddaf..ea7c6a8c36 100644 --- a/audio/wavaudio.c +++ b/audio/wavaudio.c @@ -44,7 +44,7 @@ static struct { 44100, 2, AUD_FMT_S16, - AUDIO_HOST_ENDIANNESS + 0 }, "qemu.wav" }; |