New audio PES handling.

New easier and more flexible audio PES packet parser, which includes own
codec parser.
Removed av_parser use.
Reduced audio buffer time, faster channel switch.
New audio transport stream parser (not enabled as default).

ChangeLog

@@ -1,6 +1,7 @@
 User johns
 Date:
 
+    New audio PES packet parser.
     Fix bug: Grabbing a JPG image fails while suspended.
     Add support for hot keys.
     Add support to use characters input in edit mode.

Makefile

@@ -19,8 +19,9 @@ GIT_REV = $(shell git describe --always 2>/dev/null)
 ### Configuration (edit this for your needs)
 
 CONFIG := #-DDEBUG
-CONFIG += -DAV_INFO
+CONFIG += -DAV_INFO			# debug a/v sync
-#CONFIG += -DHAVE_PTHREAD_NAME
+#CONFIG += -DHAVE_PTHREAD_NAME		# supports new pthread_setname_np
+#CONFIG += -DUSE_TS_AUDIO		# build new ts audio parser
 CONFIG += $(shell pkg-config --exists vdpau && echo "-DUSE_VDPAU")
 CONFIG += $(shell pkg-config --exists libva && echo "-DUSE_VAAPI")
 CONFIG += $(shell pkg-config --exists alsa && echo "-DUSE_ALSA")
@@ -66,7 +67,7 @@ DEFINES += $(CONFIG) -D_GNU_SOURCE -DPLUGIN_NAME_I18N='"$(PLUGIN)"' \
 	$(if $(GIT_REV), -DGIT_REV='"$(GIT_REV)"')
 
 _CFLAGS = $(DEFINES) $(INCLUDES) \
-	$(shell pkg-config --cflags libavcodec libavformat) \
+	$(shell pkg-config --cflags libavcodec) \
 	`pkg-config --cflags x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
 		xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
 	`pkg-config --cflags gl glu` \
@@ -82,7 +83,7 @@ override CXXFLAGS += $(_CFLAGS)
 override CFLAGS	  += $(_CFLAGS)
 
 LIBS += -lrt \
-	$(shell pkg-config --libs libavcodec libavformat) \
+	$(shell pkg-config --libs libavcodec) \
 	`pkg-config --libs x11 x11-xcb xcb xcb-xv xcb-shm xcb-dpms xcb-atom\
 		xcb-screensaver xcb-randr xcb-glx xcb-icccm xcb-keysyms`\
 	`pkg-config --libs gl glu` \

Todo

@@ -26,6 +26,8 @@ missing:
     Option deinterlace off / deinterlace force!
     ColorSpace aren't configurable with the gui.
     Inverse telecine isn't configurable with the gui.
+    Replay of old vdr 1.6 recordings.
+    svdrp support for hot-keys.
 
 crash:
     AudioPlayHandlerThread -> pthread_cond_wait
@@ -99,8 +101,11 @@ HDMI/SPDIF Passthrough:
     only AC-3 written
 
 playback of recording
-    pause is not reset, when replay exit
+    pause is not reset, when replay exit (fixed?)
-    replay/pause need 100% cpu
+    replay/pause need 100% cpu (fixed?)
+
+plugins:
+    mp3 plugin needs 100% cpu (OSD updates?)
 
 setup:
     Setup of decoder type.

audio.c

@@ -141,7 +141,7 @@ static unsigned AudioSampleRate;	///< audio sample rate in hz
 static unsigned AudioChannels;		///< number of audio channels
 static const int AudioBytesProSample = 2;	///< number of bytes per sample
 static int64_t AudioPTS;		///< audio pts clock
-static const int AudioBufferTime = 350;	///< audio buffer time in ms
+static int AudioBufferTime = 216;	///< audio buffer time in ms
 
 #ifdef USE_AUDIO_THREAD
 static pthread_t AudioThread;		///< audio play thread
@@ -151,7 +151,8 @@ static pthread_cond_t AudioStartCond;	///< condition variable
 static const int AudioThread;		///< dummy audio thread
 #endif
 
-extern int VideoAudioDelay;		/// import audio/video delay
+extern int VideoAudioDelay;		///< import audio/video delay
+extern int VideoGetBuffers(void);	///< Get number of input buffers.
 
 #ifdef USE_AUDIORING
 
@@ -297,7 +298,17 @@ static int AlsaAddToRingbuffer(const void *samples, int count)
     }
 
     if (!AudioRunning) {
-	if (AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
+	Debug(3, "audio/alsa: start %zd ms %d\n",
+	    (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample),
+	    VideoGetBuffers());
+	// forced start
+	if (AlsaStartThreshold * 2 < RingBufferUsedBytes(AlsaRingBuffer)) {
+	    return 1;
+	}
+	// enough video + audio buffered
+	if (VideoGetBuffers() > 1
+	    && AlsaStartThreshold < RingBufferUsedBytes(AlsaRingBuffer)) {
 	    // restart play-back
 	    return 1;
 	}
@@ -654,7 +665,8 @@ static void AlsaThread(void)
 	    }
 	    state = snd_pcm_state(AlsaPCMHandle);
 	    if (state != SND_PCM_STATE_RUNNING) {
-		Debug(3, "audio/alsa: stopping play\n");
+		Debug(3, "audio/alsa: stopping play '%s'\n",
+		    snd_pcm_state_name(state));
 		break;
 	    }
 	    pthread_yield();
@@ -720,12 +732,11 @@ static snd_pcm_t *AlsaOpenPCM(int use_ac3)
 
     // &&|| hell
     if (!(use_ac3 && ((device = AudioAC3Device)
-		|| (device = getenv("ALSA_AC3_DEVICE"))
+		|| (device = getenv("ALSA_AC3_DEVICE"))))
-		|| (device = getenv("ALSA_PASSTHROUGH_DEVICE"))))
 	&& !(device = AudioPCMDevice) && !(device = getenv("ALSA_DEVICE"))) {
 	device = "default";
     }
-    Debug(3, "audio/alsa: &&|| hell '%s'\n", device);
+    Info(_("audio/alsa: using device '%s'\n"), device);
 
     // open none blocking; if device is already used, we don't want wait
     if ((err =
@@ -752,7 +763,8 @@ static void AlsaInitPCM(void)
     snd_pcm_t *handle;
     snd_pcm_hw_params_t *hw_params;
     int err;
-    snd_pcm_uframes_t buffer_size;
+
+    //snd_pcm_uframes_t buffer_size;
 
     if (!(handle = AlsaOpenPCM(0))) {
 	return;
@@ -767,8 +779,9 @@ static void AlsaInitPCM(void)
     }
     AlsaCanPause = snd_pcm_hw_params_can_pause(hw_params);
     Info(_("audio/alsa: supports pause: %s\n"), AlsaCanPause ? "yes" : "no");
-    snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
+    // needs audio setup
-    Info(_("audio/alsa: max buffer size %lu\n"), buffer_size);
+    //snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
+    //Info(_("audio/alsa: max buffer size %lu\n"), buffer_size);
 
     AlsaPCMHandle = handle;
 }
@@ -941,7 +954,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
 	    snd_pcm_set_params(AlsaPCMHandle, SND_PCM_FORMAT_S16,
 		AlsaUseMmap ? SND_PCM_ACCESS_MMAP_INTERLEAVED :
 		SND_PCM_ACCESS_RW_INTERLEAVED, *channels, *freq, 1,
-		125 * 1000))) {
+		96 * 1000))) {
 	Error(_("audio/alsa: set params error: %s\n"), snd_strerror(err));
 
 	/*
@@ -1053,7 +1066,7 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
     // FIXME: use hw_params for buffer_size period_size
 #endif
 
-#if 1
+#if 0
     if (0) {				// no underruns allowed, play silence
 	snd_pcm_sw_params_t *sw_params;
 	snd_pcm_uframes_t boundary;
@@ -1091,16 +1104,21 @@ static int AlsaSetup(int *freq, int *channels, int use_ac3)
     // update buffer
 
     snd_pcm_get_params(AlsaPCMHandle, &buffer_size, &period_size);
-    Info(_("audio/alsa: buffer size %lu, period size %lu\n"), buffer_size,
+    Info(_("audio/alsa: buffer size %lu %lums, period size %lu %lums\n"),
-	period_size);
+	buffer_size, snd_pcm_frames_to_bytes(AlsaPCMHandle,
+	    buffer_size) * 1000 / (AudioSampleRate * AudioChannels *
+	    AudioBytesProSample), period_size,
+	snd_pcm_frames_to_bytes(AlsaPCMHandle,
+	    period_size) * 1000 / (AudioSampleRate * AudioChannels *
+	    AudioBytesProSample));
     Debug(3, "audio/alsa: state %s\n",
 	snd_pcm_state_name(snd_pcm_state(AlsaPCMHandle)));
 
     AlsaStartThreshold = snd_pcm_frames_to_bytes(AlsaPCMHandle, period_size);
     // buffer time/delay in ms
     delay = AudioBufferTime;
-    if (VideoAudioDelay > -100) {
+    if (VideoAudioDelay > 0) {
-	delay += 100 + VideoAudioDelay / 90;
+	delay += VideoAudioDelay / 90;
     }
     if (AlsaStartThreshold <
 	(*freq * *channels * AudioBytesProSample * delay) / 1000U) {
@@ -1284,7 +1302,17 @@ static int OssAddToRingbuffer(const void *samples, int count)
     }
 
     if (!AudioRunning) {
-	if (OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
+	Debug(3, "audio/oss: start %zd ms %d\n",
+	    (RingBufferUsedBytes(OssRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample),
+	    VideoGetBuffers());
+	// forced start
+	if (OssStartThreshold * 2 < RingBufferUsedBytes(OssRingBuffer)) {
+	    return 1;
+	}
+	// enough video + audio buffered
+	if (VideoGetBuffers() > 1
+	    && OssStartThreshold < RingBufferUsedBytes(OssRingBuffer)) {
 	    // restart play-back
 	    return 1;
 	}
@@ -1522,7 +1550,7 @@ static int OssOpenPCM(int use_ac3)
 	&& !(device = AudioPCMDevice) && !(device = getenv("OSS_AUDIODEV"))) {
 	device = "/dev/dsp";
     }
-    Debug(3, "audio/oss: &&|| hell '%s'\n", device);
+    Info(_("audio/oss: using device '%s'\n"), device);
 
     if ((fildes = open(device, O_WRONLY)) < 0) {
 	Error(_("audio/oss: can't open dsp device '%s': %s\n"), device,
@@ -1774,8 +1802,8 @@ static int OssSetup(int *freq, int *channels, int use_ac3)
 	OssStartThreshold = bi.bytes + tmp;
 	// buffer time/delay in ms
 	delay = AudioBufferTime;
-	if (VideoAudioDelay > -100) {
+	if (VideoAudioDelay > 0) {
-	    delay += 100 + VideoAudioDelay / 90;
+	    delay += VideoAudioDelay / 90;
 	}
 	if (OssStartThreshold <
 	    (*freq * *channels * AudioBytesProSample * delay) / 1000U) {
@@ -1964,6 +1992,9 @@ static void *AudioPlayHandlerThread(void *dummy)
 	    pthread_cond_wait(&AudioStartCond, &AudioMutex);
 	    // cond_wait can return, without signal!
 	} while (!AudioRunning);
+	Debug(3, "audio/alsa: ----> %zd ms\n",
+	    (RingBufferUsedBytes(AlsaRingBuffer) * 1000)
+	    / (AudioSampleRate * AudioChannels * AudioBytesProSample));
 	pthread_mutex_unlock(&AudioMutex);
 
 #ifdef USE_AUDIORING
@@ -2064,6 +2095,21 @@ static const AudioModule *AudioModules[] = {
 */
 void AudioEnqueue(const void *samples, int count)
 {
+    if (0) {
+	static uint32_t last;
+	static uint32_t tick;
+	static uint32_t max = 110;
+	uint64_t delay;
+
+	delay = AudioGetDelay();
+	tick = GetMsTicks();
+	if ((last && tick - last > max) || delay < 80 * 90) {
+
+	    //max = tick - last;
+	    Debug(3, "audio: packet delta %d %lu\n", tick - last, delay / 90);
+	}
+	last = tick;
+    }
     AudioUsedModule->Enqueue(samples, count);
 }
 
@@ -2183,7 +2229,7 @@ int AudioSetup(int *freq, int *channels, int use_ac3)
 void AudioPlay(void)
 {
     if (!AudioPaused) {
-	Warning("audio: not paused, check the code\n");
+	Debug(3, "audio: not paused, check the code\n");
 	return;
     }
     Debug(3, "audio: resumed\n");
@@ -2197,13 +2243,24 @@ void AudioPlay(void)
 void AudioPause(void)
 {
     if (AudioPaused) {
-	Warning("audio: already paused, check the code\n");
+	Debug(3, "audio: already paused, check the code\n");
 	return;
     }
     Debug(3, "audio: paused\n");
     AudioPaused = 1;
 }
 
+/**
+**	Set audio buffer time.
+*/
+void AudioSetBufferTime(int delay)
+{
+    if (!delay) {
+	delay = 216;
+    }
+    AudioBufferTime = delay;
+}
+
 /**
 **	Set pcm audio device.
 **

audio.h

@@ -42,6 +42,8 @@ extern int AudioSetup(int *, int *, int);	///< setup audio output
 extern void AudioPlay(void);		///< play audio
 extern void AudioPause(void);		///< pause audio
 
+extern void AudioSetBufferTime(int);	///< set audio buffer time
+
 extern void AudioSetDevice(const char *);	///< set PCM audio device
 extern void AudioSetDeviceAC3(const char *);	///< set pass-through device
 extern void AudioSetChannel(const char *);	///< set mixer channel

codec.c

@@ -33,7 +33,7 @@
     /**
     **	use av_parser to support insane dvb audio streams.
     */
-#define USE_AVPARSER
+#define noUSE_AVPARSER
 
     /// compile with passthrough support (experimental)
 #define USE_PASSTHROUGH
@@ -603,8 +603,10 @@ struct _audio_decoder_
     AVCodec *AudioCodec;		///< audio codec
     AVCodecContext *AudioCtx;		///< audio codec context
 
+#ifdef USE_AVPARSER
     /// audio parser to support insane dvb streaks
     AVCodecParserContext *AudioParser;
+#endif
     int PassthroughAC3;			///< current ac-3 pass-through
     int SampleRate;			///< current stream sample rate
     int Channels;			///< current stream channels
@@ -697,10 +699,12 @@ void CodecAudioOpen(AudioDecoder * audio_decoder, const char *name,
 	// we do not send complete frames
 	audio_decoder->AudioCtx->flags |= CODEC_FLAG_TRUNCATED;
     }
+#ifdef USE_AVPARSER
     if (!(audio_decoder->AudioParser =
 	    av_parser_init(audio_decoder->AudioCtx->codec_id))) {
 	Fatal(_("codec: can't init audio parser\n"));
     }
+#endif
     audio_decoder->SampleRate = 0;
     audio_decoder->Channels = 0;
     audio_decoder->HwSampleRate = 0;
@@ -719,10 +723,12 @@ void CodecAudioClose(AudioDecoder * audio_decoder)
 	audio_resample_close(audio_decoder->ReSample);
 	audio_decoder->ReSample = NULL;
     }
+#ifdef USE_AVPARSER
     if (audio_decoder->AudioParser) {
 	av_parser_close(audio_decoder->AudioParser);
 	audio_decoder->AudioParser = NULL;
     }
+#endif
     if (audio_decoder->AudioCtx) {
 	pthread_mutex_lock(&CodecLockMutex);
 	avcodec_close(audio_decoder->AudioCtx);
@@ -808,7 +814,7 @@ static void CodecReorderAudioFrame(int16_t * buf, int size, int channels)
 **	@param audio_decoder	audio decoder data
 **	@param avpkt		audio packet
 */
-void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
+void CodecAudioDecodeOld(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 {
     int16_t buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
 	FF_INPUT_BUFFER_PADDING_SIZE] __attribute__ ((aligned(16)));
@@ -844,8 +850,8 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 	av_init_packet(dpkt);
 	n = av_parser_parse2(audio_decoder->AudioParser, audio_ctx,
 	    &dpkt->data, &dpkt->size, spkt->data + index, spkt->size - index,
-	    !index ? (uint64_t) spkt->pts : AV_NOPTS_VALUE,
+	    !index ? spkt->pts : (int64_t) AV_NOPTS_VALUE,
-	    !index ? (uint64_t) spkt->dts : AV_NOPTS_VALUE, -1);
+	    !index ? spkt->dts : (int64_t) AV_NOPTS_VALUE, -1);
 
 	// FIXME: make this a function for both #ifdef cases
 	if (dpkt->size) {
@@ -871,7 +877,7 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 #else
 #endif
 	    // Update audio clock
-	    if ((uint64_t) dpkt->pts != AV_NOPTS_VALUE) {
+	    if (dpkt->pts != (int64_t) AV_NOPTS_VALUE) {
 		AudioSetClock(dpkt->pts);
 	    }
 	    // FIXME: must first play remainings bytes, than change and play new.
@@ -1059,6 +1065,8 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 
 #else
 
+#endif
+
 /**
 **	Decode an audio packet.
 **
@@ -1074,62 +1082,206 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
     AVCodecContext *audio_ctx;
     int index;
 
-//#define spkt avpkt
-#if 1
-    AVPacket spkt[1];
-
-    // av_new_packet reserves FF_INPUT_BUFFER_PADDING_SIZE and clears it
-    if (av_new_packet(spkt, avpkt->size)) {
-	Error(_("codec: out of memory\n"));
-	return;
-    }
-    memcpy(spkt->data, avpkt->data, avpkt->size);
-    spkt->pts = avpkt->pts;
-    spkt->dts = avpkt->dts;
-#endif
     audio_ctx = audio_decoder->AudioCtx;
     index = 0;
-    while (spkt->size > index) {
+    while (avpkt->size > index) {
-	int n;
+	int l;
 	int buf_sz;
-	AVPacket dpkt[1];
-
-	av_init_packet(dpkt);
-	dpkt->data = spkt->data + index;
-	dpkt->size = spkt->size - index;
 
 	buf_sz = sizeof(buf);
-	n = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, dpkt);
+	l = avcodec_decode_audio3(audio_ctx, buf, &buf_sz, (AVPacket *)avpkt);
-	if (n < 0) {			// no audio frame could be decompressed
+	if (l == AVERROR(EAGAIN)) {
+	    Error(_("codec: latm\n"));
+	    break;
+	}
+	if (l < 0) {			// no audio frame could be decompressed
 	    Error(_("codec: error audio data at %d\n"), index);
 	    break;
 	}
-#ifdef DEBUG
-	Debug(4, "codec/audio: -> %d\n", buf_sz);
-	if ((unsigned)buf_sz > sizeof(buf)) {
-	    abort();
-	}
-#endif
 #ifdef notyetFF_API_OLD_DECODE_AUDIO
 	// FIXME: ffmpeg git comeing
 	int got_frame;
 
-	avcodec_decode_audio4(audio_ctx, frame, &got_frame, dpkt);
+	avcodec_decode_audio4(audio_ctx, frame, &got_frame, avpkt);
 #else
 #endif
-	// FIXME: see above, old code removed
+	// Update audio clock
+	if (avpkt->pts != (int64_t) AV_NOPTS_VALUE) {
+	    AudioSetClock(avpkt->pts);
+	}
+	// FIXME: must first play remainings bytes, than change and play new.
+	if (audio_decoder->PassthroughAC3 != CodecPassthroughAC3
+	    || audio_decoder->SampleRate != audio_ctx->sample_rate
+	    || audio_decoder->Channels != audio_ctx->channels) {
+	    int err;
+	    int isAC3;
 
-	index += n;
+	    audio_decoder->PassthroughAC3 = CodecPassthroughAC3;
+	    // FIXME: use swr_convert from swresample (only in ffmpeg!)
+	    // FIXME: tell ac3 decoder to use downmix
+	    if (audio_decoder->ReSample) {
+		audio_resample_close(audio_decoder->ReSample);
+		audio_decoder->ReSample = NULL;
+	    }
+
+	    audio_decoder->SampleRate = audio_ctx->sample_rate;
+	    audio_decoder->HwSampleRate = audio_ctx->sample_rate;
+	    audio_decoder->Channels = audio_ctx->channels;
+	    // SPDIF/HDMI passthrough
+	    if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
+		audio_decoder->HwChannels = 2;
+		isAC3 = 1;
+	    } else {
+		audio_decoder->HwChannels = audio_ctx->channels;
+		isAC3 = 0;
+	    }
+
+	    // channels not support?
+	    if ((err =
+		    AudioSetup(&audio_decoder->HwSampleRate,
+			&audio_decoder->HwChannels, isAC3))) {
+		Debug(3, "codec/audio: resample %dHz *%d -> %dHz *%d\n",
+		    audio_ctx->sample_rate, audio_ctx->channels,
+		    audio_decoder->HwSampleRate, audio_decoder->HwChannels);
+
+		if (err == 1) {
+		    audio_decoder->ReSample =
+			av_audio_resample_init(audio_decoder->HwChannels,
+			audio_ctx->channels, audio_decoder->HwSampleRate,
+			audio_ctx->sample_rate, audio_ctx->sample_fmt,
+			audio_ctx->sample_fmt, 16, 10, 0, 0.8);
+		    // libav-0.8_pre didn't support 6 -> 2 channels
+		    if (!audio_decoder->ReSample) {
+			Error(_("codec/audio: resample setup error\n"));
+			audio_decoder->HwChannels = 0;
+			audio_decoder->HwSampleRate = 0;
+		    }
+		} else {
+		    Debug(3, "codec/audio: audio setup error\n");
+		    // FIXME: handle errors
+		    audio_decoder->HwChannels = 0;
+		    audio_decoder->HwSampleRate = 0;
+		    break;
+		}
+	    }
+	}
+
+	if (audio_decoder->HwSampleRate && audio_decoder->HwChannels) {
+	    // need to resample audio
+	    if (audio_decoder->ReSample) {
+		int16_t outbuf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 4 +
+		    FF_INPUT_BUFFER_PADDING_SIZE]
+		    __attribute__ ((aligned(16)));
+		int outlen;
+
+		// FIXME: libav-0.7.2 crash here
+		outlen =
+		    audio_resample(audio_decoder->ReSample, outbuf, buf,
+		    buf_sz);
+#ifdef DEBUG
+		if (outlen != buf_sz) {
+		    Debug(3, "codec/audio: possible fixed ffmpeg\n");
+		}
+#endif
+		if (outlen) {
+		    // outlen seems to be wrong in ffmpeg-0.9
+		    outlen /= audio_decoder->Channels *
+			av_get_bytes_per_sample(audio_ctx->sample_fmt);
+		    outlen *=
+			audio_decoder->HwChannels *
+			av_get_bytes_per_sample(audio_ctx->sample_fmt);
+		    Debug(4, "codec/audio: %d -> %d\n", buf_sz, outlen);
+		    CodecReorderAudioFrame(outbuf, outlen,
+			audio_decoder->HwChannels);
+		    AudioEnqueue(outbuf, outlen);
+		}
+	    } else {
+#ifdef USE_PASSTHROUGH
+		// SPDIF/HDMI passthrough
+		if (CodecPassthroughAC3 && audio_ctx->codec_id == CODEC_ID_AC3) {
+		    // build SPDIF header and append A52 audio to it
+		    // avpkt is the original data
+		    buf_sz = 6144;
+		    if (buf_sz < avpkt->size + 8) {
+			Error(_
+			    ("codec/audio: decoded data smaller than encoded\n"));
+			break;
+		    }
+		    // copy original data for output
+		    // FIXME: not 100% sure, if endian is correct
+		    buf[0] = htole16(0xF872);	// iec 61937 sync word
+		    buf[1] = htole16(0x4E1F);
+		    buf[2] = htole16(0x01 | (avpkt->data[5] & 0x07) << 8);
+		    buf[3] = htole16(avpkt->size * 8);
+		    swab(avpkt->data, buf + 4, avpkt->size);
+		    memset(buf + 4 + avpkt->size / 2, 0,
+			buf_sz - 8 - avpkt->size);
+		}
+#if 0
+		//
+		//  old experimental code
+		//
+		if (1) {
+		    // FIXME: need to detect dts
+		    // copy original data for output
+		    // FIXME: buf is sint
+		    buf[0] = 0x72;
+		    buf[1] = 0xF8;
+		    buf[2] = 0x1F;
+		    buf[3] = 0x4E;
+		    buf[4] = 0x00;
+		    switch (avpkt->size) {
+			case 512:
+			    buf[5] = 0x0B;
+			    break;
+			case 1024:
+			    buf[5] = 0x0C;
+			    break;
+			case 2048:
+			    buf[5] = 0x0D;
+			    break;
+			default:
+			    Debug(3,
+				"codec/audio: dts sample burst not supported\n");
+			    buf[5] = 0x00;
+			    break;
+		    }
+		    buf[6] = (avpkt->size * 8);
+		    buf[7] = (avpkt->size * 8) >> 8;
+		    //buf[8] = 0x0B;
+		    //buf[9] = 0x77;
+		    //printf("%x %x\n", avpkt->data[0],avpkt->data[1]);
+		    // swab?
+		    memcpy(buf + 8, avpkt->data, avpkt->size);
+		    memset(buf + 8 + avpkt->size, 0, buf_sz - 8 - avpkt->size);
+		} else if (1) {
+		    // FIXME: need to detect mp2
+		    // FIXME: mp2 passthrough
+		    // see softhddev.c version/layer
+		    // 0x04 mpeg1 layer1
+		    // 0x05 mpeg1 layer23
+		    // 0x06 mpeg2 ext
+		    // 0x07 mpeg2.5 layer 1
+		    // 0x08 mpeg2.5 layer 2
+		    // 0x09 mpeg2.5 layer 3
+		}
+		// DTS HD?
+		// True HD?
+#endif
+#endif
+		CodecReorderAudioFrame(buf, buf_sz, audio_decoder->HwChannels);
+		AudioEnqueue(buf, buf_sz);
+	    }
+	}
+
+	if (avpkt->size > l) {
+	    Error(_("codec: error more than one frame data\n"));
+	}
+
+	index += l;
     }
-
-#if 1
-    // or av_free_packet, make no difference here
-    av_destruct_packet(spkt);
-#endif
 }
 
-#endif
-
 /**
 **	Flush the audio decoder.
 **
@@ -1137,7 +1289,17 @@ void CodecAudioDecode(AudioDecoder * audio_decoder, const AVPacket * avpkt)
 */
 void CodecAudioFlushBuffers(AudioDecoder * decoder)
 {
+#ifdef USE_AVPARSER
     // FIXME: reset audio parser
+    if (decoder->AudioParser) {
+	av_parser_close(decoder->AudioParser);
+	decoder->AudioParser = NULL;
+	if (!(decoder->AudioParser =
+		av_parser_init(decoder->AudioCtx->codec_id))) {
+	    Fatal(_("codec: can't init audio parser\n"));
+	}
+    }
+#endif
     avcodec_flush_buffers(decoder->AudioCtx);
 }
 

codec.h

@@ -67,6 +67,9 @@ extern void CodecAudioOpen(AudioDecoder *, const char *, int);
     /// Close audio codec.
 extern void CodecAudioClose(AudioDecoder *);
 
+    /// Decode an audio packet.
+extern void CodecAudioDecodeOld(AudioDecoder *, const AVPacket *);
+
     /// Decode an audio packet.
 extern void CodecAudioDecode(AudioDecoder *, const AVPacket *);
 

softhddev.h

@@ -37,6 +37,8 @@ extern "C"
 
     /// C plugin play audio packet
     extern int PlayAudio(const uint8_t *, int, uint8_t);
+    /// C plugin play TS audio packet
+    extern int PlayTsAudio(const uint8_t *, int);
     /// C plugin set audio volume
     extern void SetVolumeDevice(int);
 

softhddevice.cpp

@@ -835,9 +835,11 @@ class cSoftHdDevice:public cDevice
     virtual void GetVideoSize(int &, int &, double &);
     virtual void GetOsdSize(int &, int &, double &);
     virtual int PlayVideo(const uchar *, int);
-
+#ifndef xxUSE_TS_AUDIO
+    virtual int PlayAudio(const uchar *, int, uchar);
+#endif
     //virtual int PlayTsVideo(const uchar *, int);
-#ifndef USE_AUDIO_THREAD		// FIXME: testing none threaded
+#if !defined(USE_AUDIO_THREAD) || defined(USE_TS_AUDIO)
     virtual int PlayTsAudio(const uchar *, int);
 #endif
     virtual void SetAudioChannelDevice(int);
@@ -845,7 +847,6 @@ class cSoftHdDevice:public cDevice
     virtual void SetDigitalAudioDevice(bool);
     virtual void SetAudioTrackDevice(eTrackType);
     virtual void SetVolumeDevice(int);
-    virtual int PlayAudio(const uchar *, int, uchar);
 
 // Image Grab facilities
 
@@ -1066,8 +1067,8 @@ bool cSoftHdDevice::Flush(int timeout_ms)
 **	Sets the video display format to the given one (only useful if this
 **	device has an MPEG decoder).
 */
-void cSoftHdDevice::
+void cSoftHdDevice:: SetVideoDisplayFormat(eVideoDisplayFormat
-SetVideoDisplayFormat(eVideoDisplayFormat video_display_format)
+    video_display_format)
 {
     static int last = -1;
 
@@ -1124,8 +1125,14 @@ void cSoftHdDevice::GetOsdSize(int &width, int &height, double &pixel_aspect)
 
 // ----------------------------------------------------------------------------
 
+#ifndef xxUSE_TS_AUDIO
+
 /**
 **	Play a audio packet.
+**
+**	@param data	exactly one complete PES packet (which is incomplete)
+**	@param length	length of PES packet
+**	@param id	type of audio data this packet holds
 */
 int cSoftHdDevice::PlayAudio(const uchar * data, int length, uchar id)
 {
@@ -1134,6 +1141,8 @@ int cSoftHdDevice::PlayAudio(const uchar * data, int length, uchar id)
     return::PlayAudio(data, length, id);
 }
 
+#endif
+
 void cSoftHdDevice::SetAudioTrackDevice(
     __attribute__ ((unused)) eTrackType type)
 {
@@ -1173,6 +1182,9 @@ void cSoftHdDevice::SetVolumeDevice(int volume)
 
 /**
 **	Play a video packet.
+**
+**	@param data	exactly one complete PES packet (which is incomplete)
+**	@param length	length of PES packet
 */
 int cSoftHdDevice::PlayVideo(const uchar * data, int length)
 {

video.c

@@ -329,7 +329,6 @@ static xcb_atom_t NetWmState;		///< wm-state message atom
 static xcb_atom_t NetWmStateFullscreen;	///< fullscreen wm-state message atom
 
 extern uint32_t VideoSwitch;		///< ticks for channel switch
-extern atomic_t VideoPacketsFilled;	///< how many of the buffer is used
 
 #ifdef USE_VIDEO_THREAD
 
@@ -396,19 +395,19 @@ static void VideoSetPts(int64_t * pts_p, int interlaced, const AVFrame * frame)
     int64_t pts;
 
     // update video clock
-    if ((uint64_t) * pts_p != AV_NOPTS_VALUE) {
+    if (*pts_p != (int64_t) AV_NOPTS_VALUE) {
 	*pts_p += interlaced ? 40 * 90 : 20 * 90;
     }
     //pts = frame->best_effort_timestamp;
     pts = frame->pkt_pts;
-    if ((uint64_t) pts == AV_NOPTS_VALUE || !pts) {
+    if (pts == (int64_t) AV_NOPTS_VALUE || !pts) {
 	// libav: 0.8pre didn't set pts
 	pts = frame->pkt_dts;
     }
     // libav: sets only pkt_dts which can be 0
-    if (pts && (uint64_t) pts != AV_NOPTS_VALUE) {
+    if (pts && pts != (int64_t) AV_NOPTS_VALUE) {
 	// build a monotonic pts
-	if ((uint64_t) * pts_p != AV_NOPTS_VALUE) {
+	if (*pts_p != (int64_t) AV_NOPTS_VALUE) {
 	    int64_t delta;
 
 	    delta = pts - *pts_p;
@@ -4224,7 +4223,7 @@ static void VaapiAdvanceFrame(void)
 		Warning(_
 		    ("video: display buffer empty, duping frame (%d/%d) %d\n"),
 		    decoder->FramesDuped, decoder->FrameCounter,
-		    atomic_read(&VideoPacketsFilled));
+		    VideoGetBuffers());
 	    }
 	    last_warned_frame = decoder->FrameCounter;
 	    if (!(decoder->FramesDisplayed % 300)) {
@@ -4358,10 +4357,10 @@ static void VaapiSyncDisplayFrame(VaapiDecoder * decoder)
     // FIXME: audio not known assume 333ms delay
 
     if (decoder->DupNextFrame) {
-	++decoder->FramesDuped;
 	decoder->DupNextFrame--;
-    } else if ((uint64_t) audio_clock != AV_NOPTS_VALUE
+	++decoder->FramesDuped;
-	&& (uint64_t) video_clock != AV_NOPTS_VALUE) {
+    } else if (audio_clock != (int64_t) AV_NOPTS_VALUE
+	&& video_clock != (int64_t) AV_NOPTS_VALUE) {
 	// both clocks are known
 
 	if (abs(video_clock - audio_clock) > 5000 * 90) {
@@ -4377,6 +4376,12 @@ static void VaapiSyncDisplayFrame(VaapiDecoder * decoder)
 	    Debug(3, "video: speed up video\n");
 	    decoder->DropNextFrame = 1;
 	}
+    } else if (audio_clock == (int64_t) AV_NOPTS_VALUE
+	&& video_clock != (int64_t) AV_NOPTS_VALUE) {
+	if (VideoGetBuffers() < 4) {
+	    Debug(3, "video: initial slow down video\n");
+	    decoder->DupNextFrame++;
+	}
     }
 #if defined(DEBUG) || defined(AV_INFO)
     // debug audio/video sync
@@ -4386,8 +4391,7 @@ static void VaapiSyncDisplayFrame(VaapiDecoder * decoder)
 	    VideoTimeStampString(video_clock),
 	    abs((video_clock - audio_clock) / 90) <
 	    9999 ? ((video_clock - audio_clock) / 90) : 88888,
-	    AudioGetDelay() / 90, (int)VideoDeltaPTS / 90,
+	    AudioGetDelay() / 90, (int)VideoDeltaPTS / 90, VideoGetBuffers());
-	    atomic_read(&VideoPacketsFilled));
     }
 #endif
 }
@@ -4463,7 +4467,7 @@ static void VaapiSyncRenderFrame(VaapiDecoder * decoder,
 static int64_t VaapiGetClock(const VaapiDecoder * decoder)
 {
     // pts is the timestamp of the latest decoded frame
-    if (!decoder || (uint64_t) decoder->PTS == AV_NOPTS_VALUE) {
+    if (!decoder || decoder->PTS == (int64_t) AV_NOPTS_VALUE) {
 	return AV_NOPTS_VALUE;
     }
     // subtract buffered decoded frames
@@ -7235,7 +7239,7 @@ static void VdpauAdvanceFrame(void)
 		    Warning(_
 			("video: display buffer empty, duping frame (%d/%d) %d\n"),
 			decoder->FramesDuped, decoder->FrameCounter,
-			atomic_read(&VideoPacketsFilled));
+			VideoGetBuffers());
 		}
 		last_warned_frame = decoder->FrameCounter;
 		if (!(decoder->FramesDisplayed % 300)) {
@@ -7371,8 +7375,9 @@ static void VdpauSyncDisplayFrame(VdpauDecoder * decoder)
 
     if (decoder->DupNextFrame) {
 	decoder->DupNextFrame--;
-    } else if ((uint64_t) audio_clock != AV_NOPTS_VALUE
+	++decoder->FramesDuped;
-	&& (uint64_t) video_clock != AV_NOPTS_VALUE) {
+    } else if (audio_clock != (int64_t) AV_NOPTS_VALUE
+	&& video_clock != (int64_t) AV_NOPTS_VALUE) {
 	// both clocks are known
 
 	if (abs(video_clock - audio_clock) > 5000 * 90) {
@@ -7388,6 +7393,12 @@ static void VdpauSyncDisplayFrame(VdpauDecoder * decoder)
 	    Debug(3, "video: speed up video\n");
 	    decoder->DropNextFrame = 1;
 	}
+    } else if (audio_clock == (int64_t) AV_NOPTS_VALUE
+	&& video_clock != (int64_t) AV_NOPTS_VALUE) {
+	if (VideoGetBuffers() < 4) {
+	    Debug(3, "video: initial slow down video\n");
+	    decoder->DupNextFrame++;
+	}
     }
 #if defined(DEBUG) || defined(AV_INFO)
     // debug audio/video sync
@@ -7397,8 +7408,7 @@ static void VdpauSyncDisplayFrame(VdpauDecoder * decoder)
 	    VideoTimeStampString(video_clock),
 	    abs((video_clock - audio_clock) / 90) <
 	    9999 ? ((video_clock - audio_clock) / 90) : 88888,
-	    AudioGetDelay() / 90, (int)VideoDeltaPTS / 90,
+	    AudioGetDelay() / 90, (int)VideoDeltaPTS / 90, VideoGetBuffers());
-	    atomic_read(&VideoPacketsFilled));
     }
 #endif
 }
@@ -7480,7 +7490,7 @@ static void VdpauSyncRenderFrame(VdpauDecoder * decoder,
 static int64_t VdpauGetClock(const VdpauDecoder * decoder)
 {
     // pts is the timestamp of the latest decoded frame
-    if (!decoder || (uint64_t) decoder->PTS == AV_NOPTS_VALUE) {
+    if (!decoder || decoder->PTS == (int64_t) AV_NOPTS_VALUE) {
 	return AV_NOPTS_VALUE;
     }
     // subtract buffered decoded frames
@@ -9324,6 +9334,7 @@ void VideoExit(void)
     if (VideoUsedModule) {
 	VideoUsedModule->Exit();
     }
+    VideoUsedModule = NULL;		// FIXME: NoopModule;
 #ifdef USE_GLX
     if (GlxEnabled) {
 	GlxExit();

video.h

@@ -144,5 +144,6 @@ extern void VideoExit(void);		///< Cleanup and exit video module.
 
 extern void VideoFlushInput(void);	///< Flush video input buffers.
 extern int VideoDecode(void);		///< Decode video input buffers.
+extern int VideoGetBuffers(void);	///< Get number of input buffers.
 
 /// @}