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Commit 8004ae34 authored by Warren Dukes's avatar Warren Dukes
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more alsa work

git-svn-id: https://svn.musicpd.org/mpd/trunk@3019 09075e82-0dd4-0310-85a5-a0d7c8717e4f
parent b94fa9c9
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Showing with 57 additions and 41 deletions
src/audioOutputs/audioOutput_alsa.c
View file @ 8004ae34
...@@ -43,19 +43,21 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer, ...@@ -43,19 +43,21 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer,
typedef struct _AlsaData { typedef struct _AlsaData {
char * device; char * device;
snd_pcm_t * pcm_handle; snd_pcm_t * pcmHandle;
int mmap;
alsa_writei_t * writei; alsa_writei_t * writei;
int sampleSize; int sampleSize;
int useMmap;
int canPause;
int canResume;
} AlsaData; } AlsaData;
static AlsaData * newAlsaData() { static AlsaData * newAlsaData() {
AlsaData * ret = malloc(sizeof(AlsaData)); AlsaData * ret = malloc(sizeof(AlsaData));
ret->device = NULL; ret->device = NULL;
ret->pcm_handle = NULL; ret->pcmHandle = NULL;
ret->writei = snd_pcm_writei; ret->writei = snd_pcm_writei;
ret->mmap = 0; ret->useMmap = 0;
return ret; return ret;
} }
...@@ -116,43 +118,43 @@ static int alsa_openDevice(AudioOutput * audioOutput) ...@@ -116,43 +118,43 @@ static int alsa_openDevice(AudioOutput * audioOutput)
return -1; return -1;
} }
err = snd_pcm_open(&ad->pcm_handle, ad->device, err = snd_pcm_open(&ad->pcmHandle, ad->device,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if(err < 0) { if(err < 0) {
ad->pcm_handle = NULL; ad->pcmHandle = NULL;
goto error; goto error;
} }
err = snd_pcm_nonblock(ad->pcm_handle, 0); err = snd_pcm_nonblock(ad->pcmHandle, 0);
if(err < 0) goto error; if(err < 0) goto error;
/* configure HW params */ /* configure HW params */
snd_pcm_hw_params_alloca(&hwparams); snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(ad->pcm_handle, hwparams); err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
if(err < 0) goto error; if(err < 0) goto error;
if(ad->mmap) { if(ad->useMmap) {
err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams, err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
SND_PCM_ACCESS_MMAP_INTERLEAVED); SND_PCM_ACCESS_MMAP_INTERLEAVED);
if(err < 0) { if(err < 0) {
ERROR("Cannot set mmap'ed mode on alsa device \"%s\": " ERROR("Cannot set mmap'ed mode on alsa device \"%s\": "
" %s\n", ad->device, " %s\n", ad->device,
snd_strerror(-err)); snd_strerror(-err));
ERROR("Falling back to direct write mode\n"); ERROR("Falling back to direct write mode\n");
ad->mmap = 0; ad->useMmap = 0;
} }
else ad->writei = snd_pcm_mmap_writei; else ad->writei = snd_pcm_mmap_writei;
} }
if(!ad->mmap) { if(!ad->useMmap) {
err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams, err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED); SND_PCM_ACCESS_RW_INTERLEAVED);
if(err < 0) goto error; if(err < 0) goto error;
ad->writei = snd_pcm_writei; ad->writei = snd_pcm_writei;
} }
err = snd_pcm_hw_params_set_format(ad->pcm_handle, hwparams, bitformat); err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
if(err < 0) { if(err < 0) {
ERROR("Alsa device \"%s\" does not support %i bit audio: " ERROR("Alsa device \"%s\" does not support %i bit audio: "
"%s\n", ad->device, (int)bitformat, "%s\n", ad->device, (int)bitformat,
...@@ -160,7 +162,7 @@ static int alsa_openDevice(AudioOutput * audioOutput) ...@@ -160,7 +162,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
goto fail; goto fail;
} }
err = snd_pcm_hw_params_set_channels(ad->pcm_handle, hwparams, err = snd_pcm_hw_params_set_channels(ad->pcmHandle, hwparams,
audioFormat->channels); audioFormat->channels);
if(err < 0) { if(err < 0) {
ERROR("Alsa device \"%s\" does not support %i channels: " ERROR("Alsa device \"%s\" does not support %i channels: "
...@@ -169,7 +171,7 @@ static int alsa_openDevice(AudioOutput * audioOutput) ...@@ -169,7 +171,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
goto fail; goto fail;
} }
err = snd_pcm_hw_params_set_rate_near(ad->pcm_handle, hwparams, err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
&sampleRate, 0); &sampleRate, 0);
if(err < 0 || sampleRate == 0) { if(err < 0 || sampleRate == 0) {
ERROR("Alsa device \"%s\" does not support %i Hz audio\n", ERROR("Alsa device \"%s\" does not support %i Hz audio\n",
...@@ -177,15 +179,15 @@ static int alsa_openDevice(AudioOutput * audioOutput) ...@@ -177,15 +179,15 @@ static int alsa_openDevice(AudioOutput * audioOutput)
goto fail; goto fail;
} }
err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm_handle, hwparams, err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
&alsa_buffer_time, 0); &alsa_buffer_time, 0);
if(err < 0) goto error; if(err < 0) goto error;
err = snd_pcm_hw_params_set_period_time_near(ad->pcm_handle, hwparams, err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
&alsa_period_time, 0); &alsa_period_time, 0);
if(err < 0) goto error; if(err < 0) goto error;
err = snd_pcm_hw_params(ad->pcm_handle, hwparams); err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
if(err < 0) goto error; if(err < 0) goto error;
err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size); err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
...@@ -194,15 +196,18 @@ static int alsa_openDevice(AudioOutput * audioOutput) ...@@ -194,15 +196,18 @@ static int alsa_openDevice(AudioOutput * audioOutput)
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0); err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0);
if(err < 0) goto error; if(err < 0) goto error;
ad->canPause = snd_pcm_hw_params_can_pause(hwparams);
ad->canResume = snd_pcm_hw_params_can_resume(hwparams);
/* configure SW params */ /* configure SW params */
snd_pcm_sw_params_alloca(&swparams); snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(ad->pcm_handle, swparams); snd_pcm_sw_params_current(ad->pcmHandle, swparams);
err = snd_pcm_sw_params_set_start_threshold(ad->pcm_handle, swparams, err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
alsa_buffer_size - alsa_period_size); alsa_buffer_size - alsa_period_size);
if(err < 0) goto error; if(err < 0) goto error;
err = snd_pcm_sw_params(ad->pcm_handle, swparams); err = snd_pcm_sw_params(ad->pcmHandle, swparams);
if(err < 0) goto error; if(err < 0) goto error;
ad->sampleSize = (audioFormat->bits/8)*audioFormat->channels; ad->sampleSize = (audioFormat->bits/8)*audioFormat->channels;
...@@ -215,8 +220,8 @@ error: ...@@ -215,8 +220,8 @@ error:
ERROR("Error opening alsa device \"%s\": %s\n", ad->device, ERROR("Error opening alsa device \"%s\": %s\n", ad->device,
snd_strerror(-err)); snd_strerror(-err));
fail: fail:
if(ad->pcm_handle) snd_pcm_close(ad->pcm_handle); if(ad->pcmHandle) snd_pcm_close(ad->pcmHandle);
ad->pcm_handle = NULL; ad->pcmHandle = NULL;
audioOutput->open = 0; audioOutput->open = 0;
return -1; return -1;
} }
...@@ -224,8 +229,8 @@ fail: ...@@ -224,8 +229,8 @@ fail:
static void alsa_dropBufferedAudio(AudioOutput * audioOutput) { static void alsa_dropBufferedAudio(AudioOutput * audioOutput) {
AlsaData * ad = audioOutput->data; AlsaData * ad = audioOutput->data;
snd_pcm_drop(ad->pcm_handle); snd_pcm_drop(ad->pcmHandle);
snd_pcm_prepare(ad->pcm_handle); snd_pcm_prepare(ad->pcmHandle);
} }
inline static int alsa_errorRecovery(AlsaData * ad, int err) { inline static int alsa_errorRecovery(AlsaData * ad, int err) {
...@@ -236,12 +241,19 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) { ...@@ -236,12 +241,19 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) {
DEBUG("alsa device \"%s\" was suspended\n", ad->device); DEBUG("alsa device \"%s\" was suspended\n", ad->device);
} }
switch(snd_pcm_state(ad->pcm_handle)) { switch(snd_pcm_state(ad->pcmHandle)) {
case SND_PCM_STATE_PAUSED:
err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
break;
case SND_PCM_STATE_SUSPENDED:
err = ad->canResume ?
snd_pcm_resume(ad->pcmHandle) :
snd_pcm_prepare(ad->pcmHandle);
break;
case SND_PCM_STATE_SETUP: case SND_PCM_STATE_SETUP:
case SND_PCM_STATE_XRUN: case SND_PCM_STATE_XRUN:
err = snd_pcm_prepare(ad->pcm_handle); err = snd_pcm_prepare(ad->pcmHandle);
if(err < 0) return -1; break;
return 0;
default: default:
/* unknown state, do nothing */ /* unknown state, do nothing */
break; break;
...@@ -253,10 +265,10 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) { ...@@ -253,10 +265,10 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) {
static void alsa_closeDevice(AudioOutput * audioOutput) { static void alsa_closeDevice(AudioOutput * audioOutput) {
AlsaData * ad = audioOutput->data; AlsaData * ad = audioOutput->data;
if(ad->pcm_handle) { if(ad->pcmHandle) {
snd_pcm_drain(ad->pcm_handle); snd_pcm_drain(ad->pcmHandle);
snd_pcm_close(ad->pcm_handle); snd_pcm_close(ad->pcmHandle);
ad->pcm_handle = NULL; ad->pcmHandle = NULL;
} }
audioOutput->open = 0; audioOutput->open = 0;
...@@ -271,17 +283,21 @@ static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk, ...@@ -271,17 +283,21 @@ static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk,
size /= ad->sampleSize; size /= ad->sampleSize;
while (size > 0) { while (size > 0) {
ret = ad->writei(ad->pcm_handle, playChunk, size); ret = ad->writei(ad->pcmHandle, playChunk, size);
if(ret == -EAGAIN) continue; if(ret == -EAGAIN) continue;
if(ret < 0 && alsa_errorRecovery(ad, ret) < 0) { if(ret < 0) {
ERROR("closing alsa device \"%s\" due to write error:" if( alsa_errorRecovery(ad, ret) < 0) {
" %s\n", ad->device, ERROR("closing alsa device \"%s\" due to write "
snd_strerror(-errno)); "error: %s\n", ad->device,
alsa_closeDevice(audioOutput); snd_strerror(-errno));
return -1; alsa_closeDevice(audioOutput);
return -1;
}
continue;
} }
playChunk += ret * ad->sampleSize; playChunk += ret * ad->sampleSize;
size -= ret; size -= ret;
} }
......
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