added DTS audio codec support

6 files changed, 153 insertions, 18 deletions

diff --git a/HISTORY b/HISTORY
index 9a1363a..dcba2c4 100644
--- a/HISTORY
+++ b/HISTORY
@@ -2,6 +2,7 @@ VDR Plugin 'rpihddevice' Revision History
 -----------------------------------------
 
 - new:
+  - added DTS audio codec support
   - set HDMI speaker layout to fix channel mapping for multi channel PCM output
   - support building against external ffmpeg/libav by setting EXT_LIBAV
   - support for >=ffmpeg-1.2 and >=libav-0.8 with resampling
diff --git a/audio.c b/audio.c
index 733fd6b..7ec524e 100644
--- a/audio.c
+++ b/audio.c
@@ -44,6 +44,7 @@ extern "C" {
 #  define AV_CODEC_ID_AC3  CODEC_ID_AC3
 #  define AV_CODEC_ID_EAC3 CODEC_ID_EAC3
 #  define AV_CODEC_ID_AAC  CODEC_ID_AAC
+#  define AV_CODEC_ID_DTS  CODEC_ID_DTS
 #endif
 
 #if LIBAVCODEC_VERSION_MAJOR < 54
@@ -119,6 +120,17 @@ public:
 		return m_samplingRate;
 	}
 
+	unsigned int GetFrameSize(void)
+	{
+		m_mutex->Lock();
+
+		if (!m_parsed)
+			Parse();
+
+		m_mutex->Unlock();
+		return m_packet.size;
+	}
+
 	uint64_t GetPts(void)
 	{
 		uint64_t pts = 0;
@@ -263,12 +275,12 @@ private:
 		unsigned int frameSize = 0;
 		unsigned int samplingRate = 0;
 
-		while (m_size - offset >= 3)
+		while (m_size - offset >= 4)
 		{
-			// 4 bytes 0xFFExxxxx MPEG audio
-			// 5 bytes 0x0B77xxxxxx AC-3 audio
-			// 6 bytes 0x0B77xxxxxxxx E-AC-3 audio
-			// 7/9 bytes 0xFFFxxxxxxxxxxx AAC audio
+			// 0xFFE...      MPEG audio
+			// 0x0B77...     (E)AC-3 audio
+			// 0xFFF...      AAC audio
+			// 0x7FFE8001... DTS audio
 			// PCM audio can't be found
 
 			const uint8_t *p = m_packet.data + offset;
@@ -294,6 +306,11 @@ private:
 					codec = cAudioCodec::eAAC;
 				break;
 
+			case cAudioCodec::eDTS:
+				if (DtsCheck(p, n, frameSize, channels, samplingRate))
+					codec = cAudioCodec::eDTS;
+				break;
+
 			default:
 				break;
 			}
@@ -302,7 +319,7 @@ private:
 			{
 				// if there is enough data in buffer, check if predicted next
 				// frame start is valid
-				if (n < frameSize + 3 ||
+				if (n < frameSize + 4 ||
 						FastCheck(p + frameSize) != cAudioCodec::eInvalid)
 				{
 					// if codec has been detected but buffer does not yet
@@ -364,19 +381,21 @@ private:
 	static const uint32_t Mpeg4SampleRateTable[16];
 	static const uint16_t Ac3SampleRateTable[4];
 	static const uint16_t Ac3FrameSizeTable[38][3];
+	static const uint32_t DtsSampleRateTable[16];
 
 	static cAudioCodec::eCodec FastCheck(const uint8_t *p)
 	{
 		return 	FastMpegCheck(p)  ? cAudioCodec::eMPG :
 				FastAc3Check (p)  ? cAudioCodec::eAC3 :
 				FastAdtsCheck(p)  ? cAudioCodec::eAAC :
+				FastDtsCheck (p)  ? cAudioCodec::eDTS :
 									cAudioCodec::eInvalid;
 	}
 
 	///
 	///	Fast check for MPEG audio.
 	///
-	///	4 bytes 0xFFExxxxx MPEG audio
+	///	0xFFE... MPEG audio
 	///
 	static bool FastMpegCheck(const uint8_t *p)
 	{
@@ -466,7 +485,7 @@ private:
 	///
 	///	Fast check for (E-)AC-3 audio.
 	///
-	///	5 bytes 0x0B77xxxxxx AC-3 audio
+	///	0x0B77... AC-3 audio
 	///
 	static bool FastAc3Check(const uint8_t *p)
 	{
@@ -578,7 +597,7 @@ private:
 	///
 	///	Fast check for AAC LATM audio.
 	///
-	///	3 bytes 0x56Exxx AAC LATM audio
+	///	0x56E... AAC LATM audio
 	///
 	static bool FastLatmCheck(const uint8_t *p)
 	{
@@ -618,7 +637,7 @@ private:
 	///
 	///	Fast check for ADTS Audio Data Transport Stream.
 	///
-	///	7/9 bytes 0xFFFxxxxxxxxxxx(xxxx)  ADTS audio
+	///	0xFFF...  ADTS audio
 	///
 	static bool FastAdtsCheck(const uint8_t *p)
 	{
@@ -681,6 +700,91 @@ private:
 
 	    return true;
 	}
+
+	///
+	///	Fast check for DTS Audio Data Transport Stream.
+	///
+	///	0x7FFE8001....  DTS audio
+	///
+	static bool FastDtsCheck(const uint8_t *p)
+	{
+		if (p[0] != 0x7F)			// 32bit sync
+			return false;
+		if (p[1] != 0xFE)
+			return false;
+		if (p[2] != 0x80)
+			return false;
+		if (p[3] != 0x01)
+			return false;
+		return true;
+	}
+
+	///
+	///	Check for DTS Audio Data Transport Stream.
+	///
+	///	0x7FFE8001 already checked.
+	///
+	///	AAAAAAAA AAAAAAAA AAAAAAAA AAAAAAAA BCCCCCDE EEEEEEFF FFFFFFFF FFFFGGGG
+	/// GGHHHHII IIIJKLMN OOOPQRRS TTTTTTTT TTTTTTTT UVVVVWWX XXYZaaaa
+	///
+	///	o A*32	sync word 0x7FFE8001
+	///	o B*1   frame type
+	///	o C*5   deficit sample count
+	///	o D*1   CRC present flag
+	///	o E*7   number of PCM sample blocks
+	///	o F*14  primary frame size
+	///	o G*6   audio channel arrangement
+	///	o H*4   core audio sampling frequency
+	///	o I*5   transmission bit rate
+	///	o J*1   embedded downmix enabled
+	///	o K*1   embedded dynamic range flag
+	///	o L*1   embedded time stamp flag
+	///	o M*1   auxiliary data flag
+	///	o N*1   HDCD
+	///	o O*3   extension audio descriptor flag
+	///	o P*1   extended coding flag
+	///	o Q*1   audio sync word insertion flag
+	///	o R*2   low frequency effects flag
+	///	o S*1   predictor history flag
+	///	o T*16  header CRC check (if CRC present flag set)
+	///	o U*1   multi rate interpolator switch
+	///	o V*4   encoder software revision
+	///	o W*2   copy history
+	///	o X*3   source PCM resolution
+	///	o Y*1   front sum/difference flag
+	///	o Z*1   surrounds sum/difference flag
+	///	o a*4   dialog normalization parameter
+	///
+	static bool DtsCheck(const uint8_t *p, unsigned int size,
+			unsigned int &frameSize, unsigned int &channels,
+			unsigned int &samplingRate)
+	{
+		frameSize = size;
+		if (size < 11)
+			return true;
+
+		frameSize = ((p[5] & 0x03) << 12) + (p[6] << 4) + ((p[7] & 0xF0) >> 4);
+		frameSize++;
+
+		samplingRate = DtsSampleRateTable[(p[8] & 0x3C) >> 2];
+
+		int amode = ((p[7] & 0x0F) << 2) + ((p[8] & 0xC0) >> 6);
+		channels =
+			amode == 0x00 ? 1 : 	// mono
+			amode == 0x02 ? 2 : 	// L, R
+			amode == 0x03 ? 2 : 	// (L + R), (L - R)
+			amode == 0x04 ? 2 : 	// LT, RT
+			amode == 0x05 ? 3 : 	// L, R, C
+			amode == 0x06 ? 3 : 	// L, R, S
+			amode == 0x08 ? 4 : 	// L, R, RL, RR
+			amode == 0x09 ? 5 : 0;	// L, C, R, RL, RR
+
+		if (!samplingRate || !channels)
+			return false;
+
+		if (p[10] & 0x06) channels++;
+		return true;
+	}
 };
 
 ///
@@ -741,6 +845,14 @@ const uint16_t cRpiAudioDecoder::cParser::Ac3FrameSizeTable[38][3] =
 	{1152, 1254, 1728}, {1280, 1393, 1920}, {1280, 1394, 1920},
 };
 
+///
+///	DTS sample rate table.
+///
+const uint32_t cRpiAudioDecoder::cParser::DtsSampleRateTable[16] =
+	{ 0,  8000, 16000, 32000, 64000,
+	  0, 11025, 22050, 44100, 88200,
+	  0, 12000, 24000, 48000, 96000, 0 };
+
 /* ------------------------------------------------------------------------- */
 
 #define AV_CH_LAYOUT(ch) ( \
@@ -799,6 +911,7 @@ int cRpiAudioDecoder::Init(void)
 	m_codecs[cAudioCodec::eAC3 ].codec = avcodec_find_decoder(AV_CODEC_ID_AC3);
 	m_codecs[cAudioCodec::eEAC3].codec = avcodec_find_decoder(AV_CODEC_ID_EAC3);
 	m_codecs[cAudioCodec::eAAC ].codec = avcodec_find_decoder(AV_CODEC_ID_AAC);
+	m_codecs[cAudioCodec::eDTS ].codec = avcodec_find_decoder(AV_CODEC_ID_DTS);
 
 	for (int i = 0; i < cAudioCodec::eNumCodecs; i++)
 	{
@@ -934,7 +1047,8 @@ void cRpiAudioDecoder::Action(void)
 			samplingRate = m_parser->GetSamplingRate();
 
 			outputChannels = channels;
-			SetCodec(codec, outputChannels, samplingRate);
+			SetCodec(codec, outputChannels, samplingRate,
+					m_parser->GetFrameSize());
 
 			av_frame_unref(frame);
 			m_setupChanged = false;
@@ -1127,7 +1241,7 @@ void cRpiAudioDecoder::Action(void)
 	DLOG("cAudioDecoder() thread ended");
 }
 
-void cRpiAudioDecoder::SetCodec(cAudioCodec::eCodec codec, unsigned int &channels, unsigned int samplingRate)
+void cRpiAudioDecoder::SetCodec(cAudioCodec::eCodec codec, unsigned int &channels, unsigned int samplingRate, unsigned int frameSize)
 {
 	m_omx->StopAudio();
 
@@ -1168,7 +1282,7 @@ void cRpiAudioDecoder::SetCodec(cAudioCodec::eCodec codec, unsigned int &channel
 #if FF_API_REQUEST_CHANNELS
 		m_codecs[codec].context->request_channels = channels;
 #endif
-		m_omx->SetupAudioRender(outputFormat, channels,	outputPort, samplingRate);
+		m_omx->SetupAudioRender(outputFormat, channels,	outputPort, samplingRate, frameSize);
 		ILOG("set %s audio output format to %dch %s, %d.%dkHz%s",
 				cRpiAudioPort::Str(outputPort), channels, cAudioCodec::Str(outputFormat),
 				samplingRate / 1000, (samplingRate % 1000) / 100,
diff --git a/audio.h b/audio.h
index d0da5db..ac7b838 100644
--- a/audio.h
+++ b/audio.h
@@ -33,7 +33,7 @@ protected:
 
 	virtual void Action(void);
 	void SetCodec(cAudioCodec::eCodec codec, unsigned int &channels,
-			unsigned int samplingRate);
+			unsigned int samplingRate, unsigned int frameSize = 0);
 
 	static void OnAudioSetupChanged(void *data)
 		{ (static_cast <cRpiAudioDecoder*> (data))->HandleAudioSetupChanged(); }
diff --git a/omx.c b/omx.c
index 0e68a30..b5be01c 100644
--- a/omx.c
+++ b/omx.c
@@ -944,8 +944,9 @@ void cOmx::SetVideoDecoderExtraBuffers(int extraBuffers)
 }
 
 int cOmx::SetupAudioRender(cAudioCodec::eCodec outputFormat, int channels,
-		cRpiAudioPort::ePort audioPort, int samplingRate)
+		cRpiAudioPort::ePort audioPort, int samplingRate, int frameSize)
 {
+	ELOG("%d", frameSize);
 	OMX_AUDIO_PARAM_PORTFORMATTYPE format;
 	OMX_INIT_STRUCT(format);
 	format.nPortIndex = 100;
@@ -959,7 +960,7 @@ int cOmx::SetupAudioRender(cAudioCodec::eCodec outputFormat, int channels,
 		outputFormat == cAudioCodec::eAC3  ? OMX_AUDIO_CodingDDP :
 		outputFormat == cAudioCodec::eEAC3 ? OMX_AUDIO_CodingDDP :
 		outputFormat == cAudioCodec::eAAC  ? OMX_AUDIO_CodingAAC :
-		outputFormat == cAudioCodec::eAAC ? OMX_AUDIO_CodingAAC :
+		outputFormat == cAudioCodec::eDTS  ? OMX_AUDIO_CodingDTS :
 				OMX_AUDIO_CodingAutoDetect;
 
 	if (OMX_SetParameter(ILC_GET_HANDLE(m_comp[eAudioRender]),
@@ -1009,6 +1010,22 @@ int cOmx::SetupAudioRender(cAudioCodec::eCodec outputFormat, int channels,
 			ELOG("failed to set audio render aac parameters!");
 		break;
 
+	case cAudioCodec::eDTS:
+		OMX_AUDIO_PARAM_DTSTYPE dts;
+		OMX_INIT_STRUCT(dts);
+		dts.nPortIndex = 100;
+		dts.nChannels = channels;
+		dts.nSampleRate = samplingRate;
+		dts.nDtsType = 1;
+		dts.nFormat = 3; /* 16bit, LE */
+		dts.nDtsFrameSizeBytes = frameSize;
+		OMX_AUDIO_CHANNEL_MAPPING(dts, channels);
+
+		if (OMX_SetParameter(ILC_GET_HANDLE(m_comp[eAudioRender]),
+				OMX_IndexParamAudioDts, &dts) != OMX_ErrorNone)
+			ELOG("failed to set audio render dts parameters!");
+		break;
+
 	case cAudioCodec::ePCM:
 		OMX_AUDIO_PARAM_PCMMODETYPE pcm;
 		OMX_INIT_STRUCT(pcm);
diff --git a/omx.h b/omx.h
index ea97561..cdd08ef 100644
--- a/omx.h
+++ b/omx.h
@@ -85,7 +85,8 @@ public:
 	int SetVideoCodec(cVideoCodec::eCodec codec,
 			eDataUnitType dataUnit = eArbitraryStreamSection);
 	int SetupAudioRender(cAudioCodec::eCodec outputFormat,
-			int channels, cRpiAudioPort::ePort audioPort, int samplingRate = 0);
+			int channels, cRpiAudioPort::ePort audioPort,
+			int samplingRate = 0, int frameSize = 0);
 	void GetVideoFormat(int &width, int &height, int &frameRate,
 			bool &interlaced);
 
diff --git a/tools.h b/tools.h
index fdfa528..39890ea 100644
--- a/tools.h
+++ b/tools.h
@@ -96,6 +96,7 @@ public:
 		eAC3,
 		eEAC3,
 		eAAC,
+		eDTS,
 		eNumCodecs,
 		eInvalid
 	};
@@ -105,7 +106,8 @@ public:
 				(codec == eMPG)  ? "MPEG"  :
 				(codec == eAC3)  ? "AC3"   :
 				(codec == eEAC3) ? "E-AC3" :
-				(codec == eAAC)  ? "AAC"   : "unknown";
+				(codec == eAAC)  ? "AAC"   :
+				(codec == eDTS)  ? "DTS"   : "unknown";
 	}
 };