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authorWarren Dukes <warren.dukes@gmail.com>2005-03-05 20:51:36 +0000
committerWarren Dukes <warren.dukes@gmail.com>2005-03-05 20:51:36 +0000
commit8004ae341f59dfe8c43bfa53d2b961bc98d1c673 (patch)
treebffbc387ea7e5346e106b62a413f45ec13d4b903 /src
parentb94fa9c9492ad83dfd53a4f505a78fd21f715b40 (diff)
downloadmpd-8004ae341f59dfe8c43bfa53d2b961bc98d1c673.tar.gz
mpd-8004ae341f59dfe8c43bfa53d2b961bc98d1c673.tar.xz
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more alsa work
git-svn-id: https://svn.musicpd.org/mpd/trunk@3019 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src')
-rw-r--r--src/audioOutputs/audioOutput_alsa.c98
1 files changed, 57 insertions, 41 deletions
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
index 6b50c05fc..2c3328d83 100644
--- a/src/audioOutputs/audioOutput_alsa.c
+++ b/src/audioOutputs/audioOutput_alsa.c
@@ -43,19 +43,21 @@ typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer,
typedef struct _AlsaData {
char * device;
- snd_pcm_t * pcm_handle;
- int mmap;
+ snd_pcm_t * pcmHandle;
alsa_writei_t * writei;
int sampleSize;
+ int useMmap;
+ int canPause;
+ int canResume;
} AlsaData;
static AlsaData * newAlsaData() {
AlsaData * ret = malloc(sizeof(AlsaData));
ret->device = NULL;
- ret->pcm_handle = NULL;
+ ret->pcmHandle = NULL;
ret->writei = snd_pcm_writei;
- ret->mmap = 0;
+ ret->useMmap = 0;
return ret;
}
@@ -116,43 +118,43 @@ static int alsa_openDevice(AudioOutput * audioOutput)
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);
if(err < 0) {
- ad->pcm_handle = NULL;
+ ad->pcmHandle = NULL;
goto error;
}
- err = snd_pcm_nonblock(ad->pcm_handle, 0);
+ err = snd_pcm_nonblock(ad->pcmHandle, 0);
if(err < 0) goto error;
/* configure HW params */
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(ad->mmap) {
- err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
+ if(ad->useMmap) {
+ err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
SND_PCM_ACCESS_MMAP_INTERLEAVED);
if(err < 0) {
ERROR("Cannot set mmap'ed mode on alsa device \"%s\": "
" %s\n", ad->device,
snd_strerror(-err));
ERROR("Falling back to direct write mode\n");
- ad->mmap = 0;
+ ad->useMmap = 0;
}
else ad->writei = snd_pcm_mmap_writei;
}
- if(!ad->mmap) {
- err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
+ if(!ad->useMmap) {
+ err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if(err < 0) goto error;
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) {
ERROR("Alsa device \"%s\" does not support %i bit audio: "
"%s\n", ad->device, (int)bitformat,
@@ -160,7 +162,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
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);
if(err < 0) {
ERROR("Alsa device \"%s\" does not support %i channels: "
@@ -169,7 +171,7 @@ static int alsa_openDevice(AudioOutput * audioOutput)
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);
if(err < 0 || sampleRate == 0) {
ERROR("Alsa device \"%s\" does not support %i Hz audio\n",
@@ -177,15 +179,15 @@ static int alsa_openDevice(AudioOutput * audioOutput)
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);
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);
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;
err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
@@ -194,15 +196,18 @@ static int alsa_openDevice(AudioOutput * audioOutput)
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0);
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 */
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);
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;
ad->sampleSize = (audioFormat->bits/8)*audioFormat->channels;
@@ -215,8 +220,8 @@ error:
ERROR("Error opening alsa device \"%s\": %s\n", ad->device,
snd_strerror(-err));
fail:
- if(ad->pcm_handle) snd_pcm_close(ad->pcm_handle);
- ad->pcm_handle = NULL;
+ if(ad->pcmHandle) snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
audioOutput->open = 0;
return -1;
}
@@ -224,8 +229,8 @@ fail:
static void alsa_dropBufferedAudio(AudioOutput * audioOutput) {
AlsaData * ad = audioOutput->data;
- snd_pcm_drop(ad->pcm_handle);
- snd_pcm_prepare(ad->pcm_handle);
+ snd_pcm_drop(ad->pcmHandle);
+ snd_pcm_prepare(ad->pcmHandle);
}
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);
}
- 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_XRUN:
- err = snd_pcm_prepare(ad->pcm_handle);
- if(err < 0) return -1;
- return 0;
+ err = snd_pcm_prepare(ad->pcmHandle);
+ break;
default:
/* unknown state, do nothing */
break;
@@ -253,10 +265,10 @@ inline static int alsa_errorRecovery(AlsaData * ad, int err) {
static void alsa_closeDevice(AudioOutput * audioOutput) {
AlsaData * ad = audioOutput->data;
- if(ad->pcm_handle) {
- snd_pcm_drain(ad->pcm_handle);
- snd_pcm_close(ad->pcm_handle);
- ad->pcm_handle = NULL;
+ if(ad->pcmHandle) {
+ snd_pcm_drain(ad->pcmHandle);
+ snd_pcm_close(ad->pcmHandle);
+ ad->pcmHandle = NULL;
}
audioOutput->open = 0;
@@ -271,17 +283,21 @@ static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk,
size /= ad->sampleSize;
while (size > 0) {
- ret = ad->writei(ad->pcm_handle, playChunk, size);
+ ret = ad->writei(ad->pcmHandle, playChunk, size);
if(ret == -EAGAIN) continue;
- if(ret < 0 && alsa_errorRecovery(ad, ret) < 0) {
- ERROR("closing alsa device \"%s\" due to write error:"
- " %s\n", ad->device,
- snd_strerror(-errno));
- alsa_closeDevice(audioOutput);
- return -1;
+ if(ret < 0) {
+ if( alsa_errorRecovery(ad, ret) < 0) {
+ ERROR("closing alsa device \"%s\" due to write "
+ "error: %s\n", ad->device,
+ snd_strerror(-errno));
+ alsa_closeDevice(audioOutput);
+ return -1;
+ }
+ continue;
}
+
playChunk += ret * ad->sampleSize;
size -= ret;
}