Use different alsa device for AC3/pass-through.

ChangeLog

@@ -1,5 +1,10 @@
+User Morone
+Data: Sun Jan 22 16:43:23 CET 2012
+
+    Use different alsa devices for AC3/pass-through and pcm.
+
 User johns
-Data:
+Data: Sun Jan 22 11:12:57 CET 2012
 
     Add dummy player and control for suspend mode.
     Buffertime compile time configurable in ms.

README.txt

@@ -79,6 +79,8 @@ Setup:	environment
     only if alsa is configured
 	ALSA_DEVICE=default
 		alsa PCM device name
+	ALSA_AC3_DEVICE=
+		alsa AC3/pass-though device name
 	ALSA_MIXER=default
 		alsa control device name
 	ALSA_MIXER_CHANNEL=PCM
@@ -128,6 +130,8 @@ Setup: /etc/vdr/setup.conf
 	softhddevice.AudioPassthrough = 0
 	0 = none, 1 = AC-3
 
+	for AC-3 the pass-through device is used.
+
 	softhddevice.AutoCrop.Interval = 0
 	0 disables auto-crop
 	n each 'n' frames auto-crop is checked.

audio.c

@@ -112,7 +112,7 @@ typedef struct _audio_module_
     int (*FreeBytes) (void);		///< number of bytes free in buffer
      uint64_t(*GetDelay) (void);	///< get current audio delay
     void (*SetVolume) (int);		///< set output volume
-    int (*Setup) (int *, int *);	///< setup channels, samplerate
+    int (*Setup) (int *, int *, int);	///< setup channels, samplerate
     void (*Init) (void);		///< initialize audio output module
     void (*Exit) (void);		///< cleanup audio output module
 } AudioModule;
@@ -128,6 +128,7 @@ static const char *AudioModuleName;	///< which audio module to use
     /// Selected audio module.
 static const AudioModule *AudioUsedModule = &NoopModule;
 static const char *AudioPCMDevice;	///< alsa/OSS PCM device name
+static const char *AudioAC3Device;	///< alsa/OSS AC3 device name
 static const char *AudioMixerDevice;	///< alsa/OSS mixer device name
 static const char *AudioMixerChannel;	///< alsa/OSS mixer channel name
 static volatile char AudioRunning;	///< thread running / stopped
@@ -696,18 +697,24 @@ static void AlsaThreadFlushBuffers(void)
 
 /**
 **	Open alsa pcm device.
+**
+**	@param use_ac3	use ac3/pass-through device
 */
-static snd_pcm_t *AlsaOpenPCM(void)
+static snd_pcm_t *AlsaOpenPCM(int use_ac3)
 {
     const char *device;
     snd_pcm_t *handle;
     int err;
 
-    if (!(device = AudioPCMDevice)) {
+    // &&|| hell
-	if (!(device = getenv("ALSA_DEVICE"))) {
+    if (!(use_ac3 && ((device = AudioAC3Device)
-	    device = "default";
+		|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
-	}
+	&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
+	device = "default";
     }
+    Debug(3, "audio/alsa: &&|| hell '%s'\n", device);
+
+    // open none blocking; if device is already used, we don't want wait
     if ((err =
 	    snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK,
 		SND_PCM_NONBLOCK)) < 0) {
@@ -734,7 +741,7 @@ static void AlsaInitPCM(void)
     int err;
     snd_pcm_uframes_t buffer_size;
 
-    if (!(handle = AlsaOpenPCM())) {
+    if (!(handle = AlsaOpenPCM(0))) {
 	return;
     }
 
@@ -878,6 +885,7 @@ static uint64_t AlsaGetDelay(void)
 **
 **	@param freq	sample frequency
 **	@param channels	number of channels
+**	@param use_ac3	use ac3/pass-through device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
@@ -885,7 +893,7 @@ static uint64_t AlsaGetDelay(void)
 **
 **	@todo audio changes must be queued and done when the buffer is empty
 */
-static int AlsaSetup(int *freq, int *channels)
+static int AlsaSetup(int *freq, int *channels, int use_ac3)
 {
     snd_pcm_uframes_t buffer_size;
     snd_pcm_uframes_t period_size;
@@ -904,7 +912,7 @@ static int AlsaSetup(int *freq, int *channels)
 	handle = AlsaPCMHandle;
 	AlsaPCMHandle = NULL;
 	snd_pcm_close(handle);
-	if (!(handle = AlsaOpenPCM())) {
+	if (!(handle = AlsaOpenPCM(use_ac3))) {
 	    return -1;
 	}
 	AlsaPCMHandle = handle;
@@ -1584,6 +1592,7 @@ static uint64_t OssGetDelay(void)
 **
 **	@param freq	sample frequency
 **	@param channels	number of channels
+**	@param use_ac3	use ac3/pass-through device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
@@ -1591,7 +1600,8 @@ static uint64_t OssGetDelay(void)
 **
 **	@todo audio changes must be queued and done when the buffer is empty
 */
-static int OssSetup(int *freq, int *channels)
+static int OssSetup(int *freq, int *channels, __attribute__ ((unused))
+    int use_ac3)
 {
     int ret;
     int tmp;
@@ -1690,6 +1700,8 @@ static int OssSetup(int *freq, int *channels)
 
 /**
 **	Initialize OSS audio output module.
+**
+**	@param use_ac3	use ac3/pass-through device
 */
 static void OssInit(void)
 {
@@ -1790,7 +1802,8 @@ static void NoopSetVolume( __attribute__ ((unused))
 */
 static int NoopSetup( __attribute__ ((unused))
     int *channels, __attribute__ ((unused))
-    int *freq)
+    int *freq, __attribute__ ((unused))
+    int use_ac3)
 {
     return -1;
 }
@@ -2040,6 +2053,7 @@ void AudioSetVolume(int volume)
 **
 **	@param freq	sample frequency
 **	@param channels	number of channels
+**	@param use_ac3	use ac3/pass-through device
 **
 **	@retval 0	everything ok
 **	@retval 1	didn't support frequency/channels combination
@@ -2047,9 +2061,10 @@ void AudioSetVolume(int volume)
 **
 **	@todo audio changes must be queued and done when the buffer is empty
 */
-int AudioSetup(int *freq, int *channels)
+int AudioSetup(int *freq, int *channels, int use_ac3)
 {
-    Debug(3, "audio: channels %d frequency %d hz\n", *channels, *freq);
+    Debug(3, "audio: channels %d frequency %d hz %s\n", *channels, *freq,
+	use_ac3 ? "ac3" : "pcm");
 
     // invalid parameter
     if (!freq || !channels || !*freq || !*channels) {
@@ -2059,9 +2074,9 @@ int AudioSetup(int *freq, int *channels)
     }
 #ifdef USE_AUDIORING
     // FIXME: need to store possible combination and report this
-    return AudioRingAdd(*freq, *channels);
+    return AudioRingAdd(*freq, *channels, use_ac3);
 #endif
-    return AudioUsedModule->Setup(freq, channels);
+    return AudioUsedModule->Setup(freq, channels, use_ac3);
 }
 
 /**
@@ -2073,15 +2088,37 @@ int AudioSetup(int *freq, int *channels)
 */
 void AudioSetDevice(const char *device)
 {
-    AudioModuleName = "alsa";		// detect alsa/OSS
+    if (!AudioModuleName) {
-    if (!device[0]) {
+	AudioModuleName = "alsa";	// detect alsa/OSS
-	AudioModuleName = "noop";
+	if (!device[0]) {
-    } else if (device[0] == '/') {
+	    AudioModuleName = "noop";
-	AudioModuleName = "oss";
+	} else if (device[0] == '/') {
+	    AudioModuleName = "oss";
+	}
     }
     AudioPCMDevice = device;
 }
 
+/**
+**	Set pass-through audio device.
+**
+**	@param device	name of pass-through device (fe. "hw:0,1")
+**
+**	@note this is currently usable with alsa only.
+*/
+void AudioSetDeviceAC3(const char *device)
+{
+    if (!AudioModuleName) {
+	AudioModuleName = "alsa";	// detect alsa/OSS
+	if (!device[0]) {
+	    AudioModuleName = "noop";
+	} else if (device[0] == '/') {
+	    AudioModuleName = "oss";
+	}
+    }
+    AudioAC3Device = device;
+}
+
 /**
 **	Initialize audio output module.
 **
@@ -2125,7 +2162,7 @@ void AudioInit(void)
     AudioUsedModule->Init();
     freq = 48000;
     chan = 2;
-    if (AudioSetup(&freq, &chan)) {	// set default parameters
+    if (AudioSetup(&freq, &chan, 0)) {	// set default parameters
 	Error(_("audio: can't do initial setup\n"));
     }
 #ifdef USE_AUDIO_THREAD

audio.h

@@ -37,12 +37,13 @@ extern uint64_t AudioGetDelay(void);	///< get current audio delay
 extern void AudioSetClock(int64_t);	///< set audio clock base
 extern int64_t AudioGetClock();		///< get current audio clock
 extern void AudioSetVolume(int);	///< set volume
-extern int AudioSetup(int *, int *);	///< setup audio output
+extern int AudioSetup(int *, int *, int);	///< setup audio output
 
 //extern void AudioPlay(void);		///< play audio
 //extern void AudioPause(void);		///< pause audio
 
 extern void AudioSetDevice(const char *);	///< set PCM audio device
+extern void AudioSetDeviceAC3(const char *);	///< set Passthrough device
 extern void AudioInit(void);		///< setup audio module
 extern void AudioExit(void);		///< cleanup and exit audio module
 

codec.c

@@ -320,20 +320,30 @@ static void Codec_draw_horiz_band(AVCodecContext * video_ctx,
 **
 **	@param hw_decoder	video hardware decoder
 **
-**	@returns private decoder pointer for audio/video decoder.
+**	@returns private decoder pointer for video decoder.
 */
 VideoDecoder *CodecVideoNewDecoder(VideoHwDecoder * hw_decoder)
 {
     VideoDecoder *decoder;
 
     if (!(decoder = calloc(1, sizeof(*decoder)))) {
-	Fatal(_("codec: Can't allocate vodeo decoder\n"));
+	Fatal(_("codec: can't allocate vodeo decoder\n"));
     }
     decoder->HwDecoder = hw_decoder;
 
     return decoder;
 }
 
+/**
+**	Deallocate a video decoder context.
+**
+**	@param decoder	private video decoder
+*/
+void CodecVideoDelDecoder(VideoDecoder * decoder)
+{
+    free(decoder);
+}
+
 /**
 **	Open video decoder.
 **
@@ -613,21 +623,29 @@ static char CodecPassthroughAC3;	///< pass ac3 through
 /**
 **	Allocate a new audio decoder context.
 **
-**	@param hw_decoder	video hardware decoder
+**	@returns private decoder pointer for audio decoder.
-**
-**	@returns private decoder pointer for audio/video decoder.
 */
 AudioDecoder *CodecAudioNewDecoder(void)
 {
     AudioDecoder *audio_decoder;
 
     if (!(audio_decoder = calloc(1, sizeof(*audio_decoder)))) {
-	Fatal(_("codec: Can't allocate audio decoder\n"));
+	Fatal(_("codec: can't allocate audio decoder\n"));
     }
 
     return audio_decoder;
 }
 
+/**
+**	Deallocate an audio decoder context.
+**
+**	@param decoder	private audio decoder
+*/
+void CodecAudioDelDecoder(AudioDecoder * decoder)
+{
+    free(decoder);
+}
+
 /**
 **	Open audio decoder.
 **
@@ -794,6 +812,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	    if (audio_decoder->SampleRate != audio_ctx->sample_rate
 		|| audio_decoder->Channels != audio_ctx->channels) {
 		int err;
+		int isAC3;
 
 		if (audio_decoder->ReSample) {
 		    audio_resample_close(audio_decoder->ReSample);
@@ -807,16 +826,18 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 		// SPDIF/HDMI passthrough
 		if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
 		    audio_decoder->HwChannels = 2;
+		    isAC3 = 1;
 		} else
 #endif
 		{
 		    audio_decoder->HwChannels = audio_ctx->channels;
+		    isAC3 = 0;
 		}
 
 		// channels not support?
 		if ((err =
 			AudioSetup(&audio_decoder->HwSampleRate,
-			    &audio_decoder->HwChannels))) {
+			    &audio_decoder->HwChannels, isAC3))) {
 		    Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
 			audio_ctx->sample_rate, audio_ctx->channels,
 			audio_decoder->HwSampleRate,

softhddev.c

@@ -989,6 +989,7 @@ static char ConfigStartX11Server;	///< flag start the x11 server
 const char *CommandLineHelp(void)
 {
     return "  -a device\taudio device (fe. alsa: hw:0,0 oss: /dev/dsp)\n"
+	"  -p device\taudio device (alsa only) for pass-through (hw:0,1)\n"
 	"  -d display\tdisplay of x11 server (fe. :0.0)\n"
 	"  -f\t\tstart with fullscreen window (only with window manager)\n"
 	"  -g geometry\tx11 window geometry wxh+x+y\n"
@@ -1007,10 +1008,13 @@ int ProcessArgs(int argc, char *const argv[])
     //	Parse arguments.
     //
     for (;;) {
-	switch (getopt(argc, argv, "-a:d:fg:x")) {
+	switch (getopt(argc, argv, "-a:p:d:fg:x")) {
 	    case 'a':			// audio device
 		AudioSetDevice(optarg);
 		continue;
+	    case 'p':			// pass-through audio device
+		AudioSetDeviceAC3(optarg);
+		continue;
 	    case 'd':			// x11 display name
 		X11DisplayName = optarg;
 		continue;
@@ -1154,12 +1158,12 @@ void SoftHdDeviceExit(void)
 
     if (MyVideoDecoder) {
 	CodecVideoClose(MyVideoDecoder);
-	// FIXME: CodecDelVideoDecoder(MyVideoDecoder);
+	CodecVideoDelDecoder(MyVideoDecoder);
 	MyVideoDecoder = NULL;
     }
     if (MyAudioDecoder) {
 	CodecAudioClose(MyAudioDecoder);
-	// FIXME: CodecDelAudioDecoder(MyAudioDecoder);
+	CodecAudioDelDecoder(MyAudioDecoder);
 	MyAudioDecoder = NULL;
     }