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

src/audio.c

@@ -223,7 +223,10 @@ inline void syncAudioDevicesEnabledArrays() {
 		if(myAudioDevicesEnabled[i]) {
 			openAudioOutput(audioOutputArray[i], &audio_format);
 		}
-		else closeAudioOutput(audioOutputArray[i]);
+		else {
+			dropBufferedAudioOutput(audioOutputArray[i]);
+			closeAudioOutput(audioOutputArray[i]);
+		}
 	}
 }
 
@@ -312,6 +315,23 @@ int isAudioDeviceOpen() {
 	return audioOpened;
 }
 
+void dropBufferedAudio() {
+	int i;
+
+	if(0 != memcmp(pdAudioDevicesEnabled, myAudioDevicesEnabled,
+			AUDIO_MAX_DEVICES)) 
+	{
+		syncAudioDevicesEnabledArrays();
+	}
+
+	audioBufferPos = 0;
+
+	for(i = 0; i < audioOutputArraySize; i++) {
+		if(!myAudioDevicesEnabled[i]) continue;
+		dropBufferedAudioOutput(audioOutputArray[i]);
+	}
+}
+
 void closeAudioDevice() {
 	int i;
 

src/audio.h

@@ -57,6 +57,8 @@ int openAudioDevice(AudioFormat * audioFormat);
 
 int playAudio(char * playChunk,int size);
 
+void dropBufferedAudio();
+
 void closeAudioDevice();
 
 int isAudioDeviceOpen();

src/audioOutput.c

@@ -61,6 +61,7 @@ AudioOutput * newAudioOutput(ConfigParam * param) {
 		ret->finishDriverFunc = plugin->finishDriverFunc;
 		ret->openDeviceFunc = plugin->openDeviceFunc;
 		ret->playFunc = plugin->playFunc;
+		ret->dropBufferedAudioFunc = plugin->dropBufferedAudioFunc;
 		ret->closeDeviceFunc = plugin->closeDeviceFunc;
 		ret->sendMetdataFunc = plugin->sendMetdataFunc;
 		ret->open = 0;
@@ -165,6 +166,10 @@ int playAudioOutput(AudioOutput * audioOutput, char * playChunk, int size) {
 	return ret;
 }
 
+void dropBufferedAudioOutput(AudioOutput * audioOutput) {
+	if(audioOutput->open) audioOutput->dropBufferedAudioFunc(audioOutput);
+}
+
 void closeAudioOutput(AudioOutput * audioOutput) {
 	if(audioOutput->open) audioOutput->closeDeviceFunc(audioOutput);
 }

src/audioOutput.h

@@ -38,6 +38,8 @@ typedef int (* AudioOutputOpenDeviceFunc) (AudioOutput * audioOutput);
 typedef int (* AudioOutputPlayFunc) (AudioOutput * audioOutput,
                 char * playChunk, int size);
 
+typedef void (* AudioOutputDropBufferedAudioFunc) (AudioOutput * audioOutput);
+
 typedef void (* AudioOutputCloseDeviceFunc) (AudioOutput * audioOutput);
 
 typedef void (* AudioOutputSendMetadataFunc) (AudioOutput * audioOutput,
@@ -51,6 +53,7 @@ struct _AudioOutput {
         AudioOutputFinishDriverFunc finishDriverFunc;
         AudioOutputOpenDeviceFunc openDeviceFunc;
         AudioOutputPlayFunc playFunc;
+        AudioOutputDropBufferedAudioFunc dropBufferedAudioFunc;
         AudioOutputCloseDeviceFunc closeDeviceFunc;
 	AudioOutputSendMetadataFunc sendMetdataFunc;
 
@@ -71,6 +74,7 @@ typedef struct _AudioOutputPlugin {
         AudioOutputFinishDriverFunc finishDriverFunc;
         AudioOutputOpenDeviceFunc openDeviceFunc;
         AudioOutputPlayFunc playFunc;
+        AudioOutputDropBufferedAudioFunc dropBufferedAudioFunc;
         AudioOutputCloseDeviceFunc closeDeviceFunc;
 	AudioOutputSendMetadataFunc sendMetdataFunc;
 } AudioOutputPlugin;
@@ -84,6 +88,7 @@ void unloadAudioOutputPlugin(AudioOutputPlugin * audioOutputPlugin);
 AudioOutput * newAudioOutput(ConfigParam * param);
 int openAudioOutput(AudioOutput * audioOutput, AudioFormat * audioFormat);
 int playAudioOutput(AudioOutput * audioOutput, char * playChunk, int size);
+void dropBufferedAudioOutput(AudioOutput * audioOutput);
 void closeAudioOutput(AudioOutput * audioOutput);
 void finishAudioOutput(AudioOutput * audioOutput);
 int keepAudioOutputAlive(AudioOutput * audioOutput, int ms);

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);
+	err = snd_pcm_nonblock(ad->pcm_handle, 0);
-	if(err < 0) goto error;*/
+	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_drop(ad->pcm_handle);
-		snd_pcm_drain(ad->pcm_handle);
-		ad->pcm_handle = NULL;
-	}
-
-	audioOutput->open = 0;
 }
 
 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 */
 };

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
 };
 

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;
+	}
 
-	i = audioFormat->channels;	
+	if(ioctl(od->fd, SNDCTL_DSP_SETFMT, &od->bitFormat)) {
-	if(ioctl(od->fd, SNDCTL_DSP_CHANNELS, &i)) goto fail;
+		ERROR("Error setting bitformat on OSS device \"%s\": %s\n", 
+				od->device, 
+				strerror(errno));
+		goto fail;
+	}
 
-	i = audioFormat->sampleRate;
+	if(ioctl(od->fd, SNDCTL_DSP_CHANNELS, &od->channels)) {
-	if(ioctl(od->fd, SNDCTL_DSP_SPEED, &i)) goto fail;
+		ERROR("OSS device \"%s\" does not support %i channels: %s\n", 
+				od->device,
+				od->channels,
+				strerror(errno));
+		goto fail;
+	}
 
-	i = audioFormat->bits;
+	if(ioctl(od->fd, SNDCTL_DSP_SPEED, &od->sampleRate)) {
-	if(ioctl(od->fd, SNDCTL_DSP_SAMPLESIZE, &i)) goto fail;
+		ERROR("OSS device \"%s\" does not support %i Hz audio: %s\n", 
+				od->device,
+				od->sampleRate,
+				strerror(errno));
+		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 */
 };
 

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;
 

src/decode.c

@@ -251,6 +251,7 @@ int decodeSeek(PlayerControl * pc, DecoderControl * dc, OutputBuffer * cb,
                 } \
 	} \
 	if(pc->stop) { \
+		dropBufferedAudio(); \
 		quitDecode(pc,dc); \
 		return; \
 	}