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authorWarren Dukes <warren.dukes@gmail.com>2005-03-05 14:01:13 +0000
committerWarren Dukes <warren.dukes@gmail.com>2005-03-05 14:01:13 +0000
commit92653f847492ece39beb91d547434c9a7c5e6978 (patch)
treebd52e1367f684df9a736988acd8080f638408ee9 /src/audioOutputs
parent7808fea94abf1f0fef48c8891c028c83b2e0fde8 (diff)
downloadmpd-92653f847492ece39beb91d547434c9a7c5e6978.tar.gz
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implemented dropping of current buffered audio, works for oss, but there seems
to be a "blip" for alsa devices, needs more work git-svn-id: https://svn.musicpd.org/mpd/trunk@3011 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to '')
-rw-r--r--src/audioOutputs/audioOutput_alsa.c39
-rw-r--r--src/audioOutputs/audioOutput_ao.c6
-rw-r--r--src/audioOutputs/audioOutput_oss.c86
-rw-r--r--src/audioOutputs/audioOutput_shout.c4
4 files changed, 106 insertions, 29 deletions
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
index 97dad496b..9730ade36 100644
--- a/src/audioOutputs/audioOutput_alsa.c
+++ b/src/audioOutputs/audioOutput_alsa.c
@@ -117,14 +117,14 @@ static int alsa_openDevice(AudioOutput * audioOutput)
}
err = snd_pcm_open(&ad->pcm_handle, ad->device,
- SND_PCM_STREAM_PLAYBACK, 0);
+ SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if(err < 0) {
ad->pcm_handle = NULL;
goto error;
}
- /*err = snd_pcm_nonblock(ad->pcm_handle, 0);
- if(err < 0) goto error;*/
+ err = snd_pcm_nonblock(ad->pcm_handle, 0);
+ if(err < 0) goto error;
/* configure HW params */
snd_pcm_hw_params_alloca(&hwparams);
@@ -221,28 +221,46 @@ fail:
return -1;
}
-static void alsa_closeDevice(AudioOutput * audioOutput) {
+static void alsa_dropBufferedAudio(AudioOutput * audioOutput) {
AlsaData * ad = audioOutput->data;
- if(ad->pcm_handle) {
- snd_pcm_drain(ad->pcm_handle);
- ad->pcm_handle = NULL;
- }
-
- audioOutput->open = 0;
+ snd_pcm_drop(ad->pcm_handle);
}
inline static int alsa_errorRecovery(AlsaData * ad, int err) {
if(err == -EPIPE) {
DEBUG("Underrun on alsa device \"%s\"\n", ad->device);
+ }
+ else if(err == -ESTRPIPE) {
+ DEBUG("alsa device \"%s\" was suspended\n", ad->device);
+ }
+
+ switch(snd_pcm_state(ad->pcm_handle)) {
+ case SND_PCM_STATE_SETUP:
+ case SND_PCM_STATE_XRUN:
err = snd_pcm_prepare(ad->pcm_handle);
if(err < 0) return -1;
return 0;
+ default:
+ /* unknown state, do nothing */
+ break;
}
return 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;
+ }
+
+ audioOutput->open = 0;
+}
+
static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk,
int size)
{
@@ -277,6 +295,7 @@ AudioOutputPlugin alsaPlugin =
alsa_finishDriver,
alsa_openDevice,
alsa_playAudio,
+ alsa_dropBufferedAudio,
alsa_closeDevice,
NULL /* sendMetadataFunc */
};
diff --git a/src/audioOutputs/audioOutput_ao.c b/src/audioOutputs/audioOutput_ao.c
index ee3457197..380e6e9d5 100644
--- a/src/audioOutputs/audioOutput_ao.c
+++ b/src/audioOutputs/audioOutput_ao.c
@@ -171,6 +171,10 @@ static void audioOutputAo_finishDriver(AudioOutput * audioOutput) {
if(driverInitCount == 0) ao_shutdown();
}
+static void audioOutputAo_dropBufferedAudio(AudioOutput * audioOutput) {
+ // not supported by libao
+}
+
static void audioOutputAo_closeDevice(AudioOutput * audioOutput) {
AoData * ad = (AoData *) audioOutput->data;
@@ -237,6 +241,7 @@ AudioOutputPlugin aoPlugin =
audioOutputAo_finishDriver,
audioOutputAo_openDevice,
audioOutputAo_play,
+ audioOutputAo_dropBufferedAudio,
audioOutputAo_closeDevice,
NULL /* sendMetadataFunc */
};
@@ -253,6 +258,7 @@ AudioOutputPlugin aoPlugin =
NULL,
NULL,
NULL,
+ NULL,
NULL
};
diff --git a/src/audioOutputs/audioOutput_oss.c b/src/audioOutputs/audioOutput_oss.c
index 76de8de82..fce622c5d 100644
--- a/src/audioOutputs/audioOutput_oss.c
+++ b/src/audioOutputs/audioOutput_oss.c
@@ -48,6 +48,10 @@
typedef struct _OssData {
int fd;
char * device;
+ int channels;
+ int sampleRate;
+ int bitFormat;
+ int bits;
} OssData;
static OssData * newOssData() {
@@ -154,28 +158,45 @@ static void oss_finishDriver(AudioOutput * audioOutput) {
freeOssData(od);
}
-static int oss_openDevice(AudioOutput * audioOutput)
-{
+static int oss_open(AudioOutput * audioOutput) {
OssData * od = audioOutput->data;
- AudioFormat * audioFormat = &audioOutput->outAudioFormat;
-#ifdef WORDS_BIGENDIAN
- int i = AFMT_S16_BE;
-#else
- int i = AFMT_S16_LE;
-#endif
-
- if((od->fd = open(od->device, O_WRONLY)) < 0) goto fail;
- if(ioctl(od->fd, SNDCTL_DSP_SETFMT, &i)) goto fail;
+ if((od->fd = open(od->device, O_WRONLY)) < 0) {
+ ERROR("Error opening OSS device \"%s\": %s\n", od->device,
+ strerror(errno));
+ goto fail;
+ }
+
+ if(ioctl(od->fd, SNDCTL_DSP_SETFMT, &od->bitFormat)) {
+ ERROR("Error setting bitformat on OSS device \"%s\": %s\n",
+ od->device,
+ strerror(errno));
+ goto fail;
+ }
- i = audioFormat->channels;
- if(ioctl(od->fd, SNDCTL_DSP_CHANNELS, &i)) goto fail;
+ if(ioctl(od->fd, SNDCTL_DSP_CHANNELS, &od->channels)) {
+ ERROR("OSS device \"%s\" does not support %i channels: %s\n",
+ od->device,
+ od->channels,
+ strerror(errno));
+ goto fail;
+ }
- i = audioFormat->sampleRate;
- if(ioctl(od->fd, SNDCTL_DSP_SPEED, &i)) goto fail;
+ if(ioctl(od->fd, SNDCTL_DSP_SPEED, &od->sampleRate)) {
+ ERROR("OSS device \"%s\" does not support %i Hz audio: %s\n",
+ od->device,
+ od->sampleRate,
+ strerror(errno));
+ goto fail;
+ }
- i = audioFormat->bits;
- if(ioctl(od->fd, SNDCTL_DSP_SAMPLESIZE, &i)) goto fail;
+ if(ioctl(od->fd, SNDCTL_DSP_SAMPLESIZE, &od->bits)) {
+ ERROR("OSS device \"%s\" does not support %i bit audio: %s\n",
+ od->device,
+ od->bits,
+ strerror(errno));
+ goto fail;
+ }
audioOutput->open = 1;
@@ -184,11 +205,25 @@ static int oss_openDevice(AudioOutput * audioOutput)
fail:
if(od->fd >= 0) close(od->fd);
audioOutput->open = 0;
- ERROR("Error opening OSS device \"%s\": %s\n", od->device,
- strerror(errno));
return -1;
}
+static int oss_openDevice(AudioOutput * audioOutput)
+{
+ OssData * od = audioOutput->data;
+ AudioFormat * audioFormat = &audioOutput->outAudioFormat;
+#ifdef WORDS_BIGENDIAN
+ od->bitFormat = AFMT_S16_BE;
+#else
+ od->bitFormat = AFMT_S16_LE;
+#endif
+ od->channels = audioFormat->channels;
+ od->sampleRate = audioFormat->sampleRate;
+ od->bits = audioFormat->bits;
+
+ return oss_open(audioOutput);
+}
+
static void oss_closeDevice(AudioOutput * audioOutput) {
OssData * od = audioOutput->data;
@@ -200,6 +235,17 @@ static void oss_closeDevice(AudioOutput * audioOutput) {
audioOutput->open = 0;
}
+static void oss_dropBufferedAudio(AudioOutput * audioOutput) {
+ OssData * od = audioOutput->data;
+
+ if(od->fd >= 0) {
+ ioctl(od->fd, SNDCTL_DSP_RESET, 0);
+ oss_closeDevice(audioOutput);
+ }
+
+ /*oss_open(audioOutput);*/
+}
+
static int oss_playAudio(AudioOutput * audioOutput, char * playChunk,
int size)
{
@@ -227,6 +273,7 @@ AudioOutputPlugin ossPlugin =
oss_finishDriver,
oss_openDevice,
oss_playAudio,
+ oss_dropBufferedAudio,
oss_closeDevice,
NULL /* sendMetadataFunc */
};
@@ -241,6 +288,7 @@ AudioOutputPlugin ossPlugin =
NULL,
NULL,
NULL,
+ NULL,
NULL /* sendMetadataFunc */
};
diff --git a/src/audioOutputs/audioOutput_shout.c b/src/audioOutputs/audioOutput_shout.c
index 685b1cca2..1276e6b78 100644
--- a/src/audioOutputs/audioOutput_shout.c
+++ b/src/audioOutputs/audioOutput_shout.c
@@ -361,6 +361,10 @@ static void myShout_finishDriver(AudioOutput * audioOutput) {
if(shoutInitCount == 0) shout_shutdown();
}
+static void myShout_dropBufferedAudioDevice(AudioOutput * audioOutput) {
+ // needs to be implemented
+}
+
static void myShout_closeDevice(AudioOutput * audioOutput) {
ShoutData * sd = (ShoutData *)audioOutput->data;