add time to write ts file

@@ -46,8 +46,12 @@ void SrsCodecSample::clear()
 
 	cts = 0;
 	frame_type = SrsCodecVideoAVCFrameReserved;
-	codec_id = SrsCodecVideoReserved;
 	avc_packet_type = SrsCodecVideoAVCTypeReserved;
+	
+	sound_rate = SrsCodecAudioSampleRateReserved;
+	sound_size = SrsCodecAudioSampleSizeReserved;
+	sound_type = SrsCodecAudioSoundTypeReserved;
+	aac_packet_type = SrsCodecAudioTypeReserved;
 }
 
 int SrsCodecSample::add_sample(char* bytes, int size)
@@ -79,9 +83,6 @@ SrsCodec::SrsCodec()
 	video_codec_id 			= 0;
 	audio_data_rate 		= 0;
 	audio_codec_id 			= 0;
-	sound_rate		 		= 0;
-	sound_size		 		= 0;
-	sound_type	 			= 0;
 	profile 				= 0;
 	level 					= 0;
 	avc_extra_size 			= 0;
@@ -124,12 +125,15 @@ int SrsCodec::audio_aac_demux(int8_t* data, int size, SrsCodecSample* sample)
 	
 	int8_t sound_format = stream->read_1bytes();
 	
-	sound_type = sound_format & 0x01;
-	sound_size = (sound_format >> 1) & 0x01;
-	sound_rate = (sound_format >> 2) & 0x01;
+	int8_t sound_type = sound_format & 0x01;
+	int8_t sound_size = (sound_format >> 1) & 0x01;
+	int8_t sound_rate = (sound_format >> 2) & 0x01;
 	sound_format = (sound_format >> 4) & 0x0f;
 	
 	audio_codec_id = sound_format;
+	sample->sound_type = (SrsCodecAudioSoundType)sound_type;
+	sample->sound_rate = (SrsCodecAudioSampleRate)sound_rate;
+	sample->sound_size = (SrsCodecAudioSampleSize)sound_size;
 	
 	// only support aac
 	if (audio_codec_id != SrsCodecAudioAAC) {
@@ -145,6 +149,7 @@ int SrsCodec::audio_aac_demux(int8_t* data, int size, SrsCodecSample* sample)
 	}
 	
 	int8_t aac_packet_type = stream->read_1bytes();
+	sample->aac_packet_type = (SrsCodecAudioType)aac_packet_type;
 	
 	if (aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
 		// AudioSpecificConfig
@@ -206,7 +211,6 @@ int SrsCodec::video_avc_demux(int8_t* data, int size, SrsCodecSample* sample)
 	frame_type = (frame_type >> 4) & 0x0f;
 	
 	sample->frame_type = (SrsCodecVideoAVCFrame)frame_type;
-	sample->codec_id = (SrsCodecVideo)codec_id;
 	
 	// only support h.264/avc
 	if (codec_id != SrsCodecVideoAVC) {

@@ -132,6 +132,7 @@ enum SrsCodecAudio
 // 	1 = AAC raw
 enum SrsCodecAudioType
 {
+	SrsCodecAudioTypeReserved						= -1,
 	SrsCodecAudioTypeSequenceHeader 				= 0,
 	SrsCodecAudioTypeRawData 						= 1,
 };
@@ -143,10 +144,12 @@ enum SrsCodecAudioType
 // 3 = 44 kHz = 44100 Hz
 enum SrsCodecAudioSampleRate
 {
-	SrsCodecAudioSampleRate5512 				= 0,
-	SrsCodecAudioSampleRate11025 				= 1,
-	SrsCodecAudioSampleRate22050 				= 2,
-	SrsCodecAudioSampleRate44100 				= 3,
+	SrsCodecAudioSampleRateReserved					= -1,
+	
+	SrsCodecAudioSampleRate5512 					= 0,
+	SrsCodecAudioSampleRate11025 					= 1,
+	SrsCodecAudioSampleRate22050 					= 2,
+	SrsCodecAudioSampleRate44100 					= 3,
 };
 
 // Size of each audio sample. This parameter only pertains to
@@ -156,8 +159,10 @@ enum SrsCodecAudioSampleRate
 // 1 = 16-bit samples
 enum SrsCodecAudioSampleSize
 {
-	SrsCodecAudioSampleSize8bit 				= 0,
-	SrsCodecAudioSampleSize16bit 				= 1,
+	SrsCodecAudioSampleSizeReserved					= -1,
+	
+	SrsCodecAudioSampleSize8bit 					= 0,
+	SrsCodecAudioSampleSize16bit 					= 1,
 };
 
 // Mono or stereo sound
@@ -165,8 +170,10 @@ enum SrsCodecAudioSampleSize
 // 1 = Stereo sound
 enum SrsCodecAudioSoundType
 {
-	SrsCodecAudioSoundTypeMono 					= 0,
-	SrsCodecAudioSoundTypeStereo 				= 1,
+	SrsCodecAudioSoundTypeReserved					= -1, 
+	
+	SrsCodecAudioSoundTypeMono 						= 0,
+	SrsCodecAudioSoundTypeStereo 					= 1,
 };
 
 /**
@@ -188,12 +195,17 @@ public:
 	SrsCodecBuffer buffers[SRS_MAX_CODEC_SAMPLE];
 public:
 	bool is_video;
+	// video specified
+	SrsCodecVideoAVCFrame frame_type;
+	SrsCodecVideoAVCType avc_packet_type;
 	// CompositionTime, video_file_format_spec_v10_1.pdf, page 78.
 	// cts = pts - dts, where dts = flvheader->timestamp.
 	int32_t cts;
-	SrsCodecVideoAVCFrame frame_type;
-	SrsCodecVideo codec_id;
-	SrsCodecVideoAVCType avc_packet_type;
+	// audio specified
+	SrsCodecAudioSampleRate sound_rate;
+	SrsCodecAudioSampleSize sound_size;
+	SrsCodecAudioSoundType sound_type;
+	SrsCodecAudioType aac_packet_type;
 public:
 	SrsCodecSample();
 	virtual ~SrsCodecSample();
@@ -230,12 +242,6 @@ public:
 	*/
 	// @see: SrsCodecAudioType
 	int			audio_codec_id;
-	// @see: SrsCodecAudioSampleRate
-	int			sound_rate;
-	// @see: SrsCodecAudioSampleSize
-	int			sound_size;
-	// @see: SrsCodecAudioSoundType
-	int			sound_type;
 	int			audio_data_rate; // in bps
 	// the avc extra data, the AVC sequence header,
 	// without the flv codec header,

@@ -139,41 +139,11 @@ int SrsHLS::on_meta_data(SrsOnMetaDataPacket* metadata)
 	if ((prop = obj->get_property("audiodatarate")) != NULL && prop->is_number()) {
 		codec->audio_data_rate = (int)(1000 * srs_amf0_convert<SrsAmf0Number>(prop)->value);
 	}
-	if ((prop = obj->get_property("audiosamplerate")) != NULL && prop->is_number()) {
-		int sound_rate = (int)srs_amf0_convert<SrsAmf0Number>(prop)->value;
-		if (sound_rate == 5512) {
-			codec->sound_rate = SrsCodecAudioSampleRate5512;
-		} else if (sound_rate == 11025) {
-			codec->sound_rate = SrsCodecAudioSampleRate11025;
-		} else if (sound_rate == 22050) {
-			codec->sound_rate = SrsCodecAudioSampleRate22050;
-		} else if (sound_rate == 44100) {
-			codec->sound_rate = SrsCodecAudioSampleRate44100;
-		} else {
-			ret = ERROR_HLS_METADATA;
-			srs_error("invalid sound_rate of metadata: %d, ret=%d", sound_rate, ret);
-			return ret;
-		}
-	}
-	if ((prop = obj->get_property("audiosamplesize")) != NULL && prop->is_number()) {
-		int sound_size = (int)srs_amf0_convert<SrsAmf0Number>(prop)->value;
-		if (sound_size == 16) {
-			codec->sound_size = SrsCodecAudioSampleSize16bit;
-		} else if (sound_size == 8) {
-			codec->sound_size = SrsCodecAudioSampleSize8bit;
-		} else {
-			ret = ERROR_HLS_METADATA;
-			srs_error("invalid sound_size of metadata: %d, ret=%d", sound_size, ret);
-			return ret;
-		}
-	}
-	if ((prop = obj->get_property("stereo")) != NULL && prop->is_number()) {
-		if (srs_amf0_convert<SrsAmf0Boolean>(prop)->value) {
-			codec->sound_type = SrsCodecAudioSoundTypeStereo;
-		} else {
-			codec->sound_type = SrsCodecAudioSoundTypeMono;
-		}
-	}
+	
+	// ignore the following, for each flv/rtmp packet contains them:
+	// audiosamplerate, sample->sound_rate
+	// audiosamplesize, sample->sound_size
+	// stereo, 			sample->sound_type
 	
 	return ret;
 }
@@ -196,7 +166,15 @@ int SrsHLS::on_audio(SrsCommonMessage* audio)
 		return ret;
 	}
 	
-	if ((ret = muxer->write(codec, sample)) != ERROR_SUCCESS) {
+	// ignore sequence header
+	if (sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
+		return ret;
+	}
+	
+	u_int32_t timestamp = audio->header.timestamp;
+	// TODO: correct the timestamp.
+	
+	if ((ret = muxer->write_audio(timestamp, codec, sample)) != ERROR_SUCCESS) {
 		return ret;
 	}
 	
@@ -221,7 +199,15 @@ int SrsHLS::on_video(SrsCommonMessage* video)
 		return ret;
 	}
 	
-	if ((ret = muxer->write(codec, sample)) != ERROR_SUCCESS) {
+	// ignore sequence header
+	if (sample->frame_type == SrsCodecVideoAVCFrameKeyFrame && sample->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
+		return ret;
+	}
+	
+	u_int32_t timestamp = video->header.timestamp;
+	// TODO: correct the timestamp.
+	
+	if ((ret = muxer->write_video(timestamp, codec, sample)) != ERROR_SUCCESS) {
 		return ret;
 	}
 	
@@ -229,7 +215,7 @@ int SrsHLS::on_video(SrsCommonMessage* video)
 }
 
 // @see: ngx_rtmp_mpegts_header
-static u_char mpegts_header[] = {
+u_int8_t mpegts_header[] = {
     /* TS */
     0x47, 0x40, 0x00, 0x10, 0x00,
     /* PSI */
@@ -329,9 +315,21 @@ int SrsTSMuxer::open(std::string _path)
 	return ret;
 }
 
-int SrsTSMuxer::write(SrsCodec* codec, SrsCodecSample* sample)
+int SrsTSMuxer::write_audio(u_int32_t time, SrsCodec* codec, SrsCodecSample* sample)
+{
+	int ret = ERROR_SUCCESS;
+	
+	static u_int8_t packet[188];
+	
+	return ret;
+}
+
+int SrsTSMuxer::write_video(u_int32_t time, SrsCodec* codec, SrsCodecSample* sample)
 {
 	int ret = ERROR_SUCCESS;
+	
+	static u_int8_t packet[188];
+	
 	return ret;
 }
 

@@ -66,7 +66,8 @@ public:
 	virtual ~SrsTSMuxer();
 public:
 	virtual int open(std::string _path);
-	virtual int write(SrsCodec* codec, SrsCodecSample* sample);
+	virtual int write_audio(u_int32_t time, SrsCodec* codec, SrsCodecSample* sample);
+	virtual int write_video(u_int32_t time, SrsCodec* codec, SrsCodecSample* sample);
 	virtual void close();
 };
 

@@ -199,7 +199,7 @@ struct SrsMessageHeader
 	* The 4 bytes are packed in the big-endian order.
 	* @remark, used as calc timestamp when decode and encode time.
 	*/
-	int32_t timestamp;
+	u_int32_t timestamp;
 	
 	SrsMessageHeader();
 	virtual ~SrsMessageHeader();

@@ -192,7 +192,7 @@ int SrsConsumer::jitter_correct(SrsSharedPtrMessage* msg, int audio_sample_rate,
 	* 3. last_pkt_correct_time: simply add the positive delta, 
 	* 	and enforce the time monotonically.
 	*/
-	int32_t time = msg->header.timestamp;
+	u_int32_t time = msg->header.timestamp;
 	int32_t delta = time - last_pkt_time;
 
 	// if jitter detected, reset the delta.
@@ -207,7 +207,7 @@ int SrsConsumer::jitter_correct(SrsSharedPtrMessage* msg, int audio_sample_rate,
 		}
 
 		// sometimes, the time is absolute time, so correct it again.
-		if (delta > CONST_MAX_JITTER_MS) {
+		if (delta < 0 || delta > CONST_MAX_JITTER_MS) {
 			delta = DEFAULT_FRAME_TIME_MS;
 		}
 		

@@ -46,8 +46,8 @@ class SrsHLS;
 class SrsConsumer
 {
 private:
-	int32_t last_pkt_time;
-	int32_t last_pkt_correct_time;
+	u_int32_t last_pkt_time;
+	u_int32_t last_pkt_correct_time;
 	SrsSource* source;
 	std::vector<SrsSharedPtrMessage*> msgs;
 	bool paused;