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Diffstat (limited to 'audio/alsaaudio.c')
-rw-r--r--audio/alsaaudio.c218
1 files changed, 129 insertions, 89 deletions
diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c
index f7748ca82f..3ab264e01b 100644
--- a/audio/alsaaudio.c
+++ b/audio/alsaaudio.c
@@ -31,15 +31,12 @@ 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 {
@@ -58,6 +55,7 @@ static struct {
int buffer_size_out_overriden;
int period_size_out_overriden;
+ int verbose;
} conf = {
#ifdef HIGH_LATENCY
.size_in_usec_in = 1,
@@ -73,8 +71,8 @@ static struct {
#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_in = DEFAULT_BUFFER_SIZE * 4,
+ .period_size_in = DEFAULT_PERIOD_SIZE * 4,
.buffer_size_out = DEFAULT_BUFFER_SIZE,
.period_size_out = DEFAULT_PERIOD_SIZE,
.buffer_size_in_overriden = 0,
@@ -82,7 +80,8 @@ static struct {
.period_size_in_overriden = 0,
.period_size_out_overriden = 0,
#endif
- .threshold = 0
+ .threshold = 0,
+ .verbose = 0
};
struct alsa_params_req {
@@ -97,7 +96,6 @@ struct alsa_params_obt {
int freq;
audfmt_e fmt;
int nchannels;
- int can_pause;
snd_pcm_uframes_t samples;
};
@@ -474,12 +472,6 @@ static int alsa_open (int in, struct alsa_params_req *req,
goto err;
}
- obt->can_pause = snd_pcm_hw_params_can_pause (hw_params);
- if (obt->can_pause < 0) {
- alsa_logerr (err, "Could 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;
@@ -527,6 +519,28 @@ static int alsa_recover (snd_pcm_t *handle)
return 0;
}
+static snd_pcm_sframes_t alsa_get_avail (snd_pcm_t *handle)
+{
+ snd_pcm_sframes_t avail;
+
+ avail = snd_pcm_avail_update (handle);
+ if (avail < 0) {
+ if (avail == -EPIPE) {
+ if (!alsa_recover (handle)) {
+ avail = snd_pcm_avail_update (handle);
+ }
+ }
+
+ if (avail < 0) {
+ alsa_logerr (avail,
+ "Could not obtain number of available frames\n");
+ return -1;
+ }
+ }
+
+ return avail;
+}
+
static int alsa_run_out (HWVoiceOut *hw)
{
ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;
@@ -541,57 +555,53 @@ static int alsa_run_out (HWVoiceOut *hw)
return 0;
}
- avail = snd_pcm_avail_update (alsa->handle);
+ avail = alsa_get_avail (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, "Could not get amount free space\n");
+ dolog ("Could not get number of available playback frames\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);
+ int len = 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);
+ hw->clip (dst, src, len);
- while (convert_samples) {
- written = snd_pcm_writei (alsa->handle, dst, convert_samples);
+ while (len) {
+ written = snd_pcm_writei (alsa->handle, dst, len);
- if (written < 0) {
+ if (written <= 0) {
switch (written) {
- case -EPIPE:
- if (!alsa_recover (alsa->handle)) {
- continue;
+ case 0:
+ if (conf.verbose) {
+ dolog ("Failed to write %d frames (wrote zero)\n", len);
}
- dolog ("Failed to write %d frames to %p, "
- "handle %p not prepared\n",
- convert_samples,
- dst,
- alsa->handle);
goto exit;
- case -EAGAIN:
+ case -EPIPE:
+ if (alsa_recover (alsa->handle)) {
+ alsa_logerr (written, "Failed to write %d frames\n",
+ len);
+ goto exit;
+ }
+ if (conf.verbose) {
+ dolog ("Recovering from playback xrun\n");
+ }
continue;
+ case -EAGAIN:
+ goto exit;
+
default:
alsa_logerr (written, "Failed to write %d frames to %p\n",
- convert_samples, dst);
+ len, dst);
goto exit;
}
}
@@ -599,7 +609,7 @@ static int alsa_run_out (HWVoiceOut *hw)
mixeng_clear (src, written);
rpos = (rpos + written) % hw->samples;
samples -= written;
- convert_samples -= written;
+ len -= written;
dst = advance (dst, written << hw->info.shift);
src += written;
}
@@ -659,7 +669,6 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
&obt_as,
audio_need_to_swap_endian (endianness)
);
- alsa->can_pause = obt.can_pause;
hw->samples = obt.samples;
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, obt.samples, 1 << hw->info.shift);
@@ -671,46 +680,46 @@ static int alsa_init_out (HWVoiceOut *hw, audsettings_t *as)
}
alsa->handle = handle;
- alsa->was_enabled = 0;
return 0;
}
-static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+static int alsa_voice_ctl (snd_pcm_t *handle, const char *typ, int pause)
{
int err;
+
+ if (pause) {
+ err = snd_pcm_drop (handle);
+ if (err < 0) {
+ alsa_logerr (err, "Could not stop %s", typ);
+ return -1;
+ }
+ }
+ else {
+ err = snd_pcm_prepare (handle);
+ if (err < 0) {
+ alsa_logerr (err, "Could not prepare handle for %s", typ);
+ return -1;
+ }
+ }
+
+ return 0;
+}
+
+static int alsa_ctl_out (HWVoiceOut *hw, int cmd, ...)
+{
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;
+ return alsa_voice_ctl (alsa->handle, "playback", 0);
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 alsa_voice_ctl (alsa->handle, "playback", 1);
}
- return 0;
+
+ return -1;
}
static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
@@ -749,7 +758,6 @@ static int alsa_init_in (HWVoiceIn *hw, audsettings_t *as)
&obt_as,
audio_need_to_swap_endian (endianness)
);
- alsa->can_pause = obt.can_pause;
hw->samples = obt.samples;
alsa->pcm_buf = audio_calloc (AUDIO_FUNC, hw->samples, 1 << hw->info.shift);
@@ -783,6 +791,7 @@ static int alsa_run_in (HWVoiceIn *hw)
int i;
int live = audio_pcm_hw_get_live_in (hw);
int dead = hw->samples - live;
+ int decr;
struct {
int add;
int len;
@@ -790,22 +799,36 @@ static int alsa_run_in (HWVoiceIn *hw)
{ hw->wpos, 0 },
{ 0, 0 }
};
-
+ snd_pcm_sframes_t avail;
snd_pcm_uframes_t read_samples = 0;
if (!dead) {
return 0;
}
- if (hw->wpos + dead > hw->samples) {
+ avail = alsa_get_avail (alsa->handle);
+ if (avail < 0) {
+ dolog ("Could not get number of captured frames\n");
+ return 0;
+ }
+
+ if (!avail && (snd_pcm_state (alsa->handle) == SND_PCM_STATE_PREPARED)) {
+ avail = hw->samples;
+ }
+
+ decr = audio_MIN (dead, avail);
+ if (!decr) {
+ return 0;
+ }
+
+ if (hw->wpos + decr > hw->samples) {
bufs[0].len = (hw->samples - hw->wpos);
- bufs[1].len = (dead - (hw->samples - hw->wpos));
+ bufs[1].len = (decr - (hw->samples - hw->wpos));
}
else {
- bufs[0].len = dead;
+ bufs[0].len = decr;
}
-
for (i = 0; i < 2; ++i) {
void *src;
st_sample_t *dst;
@@ -820,24 +843,27 @@ static int alsa_run_in (HWVoiceIn *hw)
while (len) {
nread = snd_pcm_readi (alsa->handle, src, len);
- if (nread < 0) {
+ if (nread <= 0) {
switch (nread) {
- case -EPIPE:
- if (!alsa_recover (alsa->handle)) {
- continue;
+ case 0:
+ if (conf.verbose) {
+ dolog ("Failed to read %ld frames (read zero)\n", len);
}
- dolog (
- "Failed to read %ld frames from %p, "
- "handle %p not prepared\n",
- len,
- src,
- alsa->handle
- );
goto exit;
- case -EAGAIN:
+ case -EPIPE:
+ if (alsa_recover (alsa->handle)) {
+ alsa_logerr (nread, "Failed to read %ld frames\n", len);
+ goto exit;
+ }
+ if (conf.verbose) {
+ dolog ("Recovering from capture xrun\n");
+ }
continue;
+ case -EAGAIN:
+ goto exit;
+
default:
alsa_logerr (
nread,
@@ -871,9 +897,19 @@ static int alsa_read (SWVoiceIn *sw, void *buf, int size)
static int alsa_ctl_in (HWVoiceIn *hw, int cmd, ...)
{
- (void) hw;
- (void) cmd;
- return 0;
+ ALSAVoiceIn *alsa = (ALSAVoiceIn *) hw;
+
+ switch (cmd) {
+ case VOICE_ENABLE:
+ ldebug ("enabling voice\n");
+ return alsa_voice_ctl (alsa->handle, "capture", 0);
+
+ case VOICE_DISABLE:
+ ldebug ("disabling voice\n");
+ return alsa_voice_ctl (alsa->handle, "capture", 1);
+ }
+
+ return -1;
}
static void *alsa_audio_init (void)
@@ -909,6 +945,10 @@ static struct audio_option alsa_options[] = {
{"ADC_DEV", AUD_OPT_STR, &conf.pcm_name_in,
"ADC device name", NULL, 0},
+
+ {"VERBOSE", AUD_OPT_BOOL, &conf.verbose,
+ "Behave in a more verbose way", NULL, 0},
+
{NULL, 0, NULL, NULL, NULL, 0}
};
generated by cgit v1.2.3 (git 2.26.2) at 2024-12-19 01:51:35 +0000