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for #250, merge avc to codec. use queue to dequeue.
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winlin
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12 changed files with 959 additions and 928 deletions
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@ -30,6 +30,10 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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#include <srs_core.hpp>
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#include <string>
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class SrsStream;
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// AACPacketType IF SoundFormat == 10 UI8
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// The following values are defined:
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// 0 = AAC sequence header
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@ -212,4 +216,263 @@ public:
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static bool audio_is_aac(char* data, int size);
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};
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/**
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* the public data, event HLS disable, others can use it.
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*/
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/**
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* the flv sample rate map
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*/
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extern int flv_sample_rates[];
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/**
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* the aac sample rate map
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*/
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extern int aac_sample_rates[];
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#define __SRS_SRS_MAX_CODEC_SAMPLE 128
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#define __SRS_AAC_SAMPLE_RATE_UNSET 15
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// in ms, for HLS aac flush the audio
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#define SRS_CONF_DEFAULT_AAC_DELAY 100
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// max PES packets size to flush the video.
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#define SRS_AUTO_HLS_AUDIO_CACHE_SIZE 1024 * 1024
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/**
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* the FLV/RTMP supported audio sample size.
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* Size of each audio sample. This parameter only pertains to
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* uncompressed formats. Compressed formats always decode
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* to 16 bits internally.
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* 0 = 8-bit samples
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* 1 = 16-bit samples
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*/
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enum SrsCodecAudioSampleSize
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{
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// set to the max value to reserved, for array map.
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SrsCodecAudioSampleSizeReserved = 2,
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SrsCodecAudioSampleSize8bit = 0,
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SrsCodecAudioSampleSize16bit = 1,
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};
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/**
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* the FLV/RTMP supported audio sound type/channel.
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* Mono or stereo sound
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* 0 = Mono sound
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* 1 = Stereo sound
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*/
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enum SrsCodecAudioSoundType
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{
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// set to the max value to reserved, for array map.
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SrsCodecAudioSoundTypeReserved = 2,
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SrsCodecAudioSoundTypeMono = 0,
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SrsCodecAudioSoundTypeStereo = 1,
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};
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/**
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* the codec sample unit.
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* for h.264 video packet, a NALU is a sample unit.
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* for aac raw audio packet, a NALU is the entire aac raw data.
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* for sequence header, it's not a sample unit.
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*/
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class SrsCodecSampleUnit
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{
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public:
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/**
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* the sample bytes is directly ptr to packet bytes,
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* user should never use it when packet destroyed.
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*/
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int size;
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char* bytes;
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public:
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SrsCodecSampleUnit();
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virtual ~SrsCodecSampleUnit();
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};
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/**
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* the samples in the flv audio/video packet.
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* the sample used to analysis a video/audio packet,
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* split the h.264 NALUs to buffers, or aac raw data to a buffer,
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* and decode the video/audio specified infos.
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*
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* the sample unit:
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* a video packet codec in h.264 contains many NALUs, each is a sample unit.
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* a audio packet codec in aac is a sample unit.
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* @remark, the video/audio sequence header is not sample unit,
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* all sequence header stores as extra data,
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* @see SrsAvcAacCodec.avc_extra_data and SrsAvcAacCodec.aac_extra_data
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* @remark, user must clear all samples before decode a new video/audio packet.
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*/
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class SrsCodecSample
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{
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public:
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/**
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* each audio/video raw data packet will dumps to one or multiple buffers,
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* the buffers will write to hls and clear to reset.
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* generally, aac audio packet corresponding to one buffer,
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* where avc/h264 video packet may contains multiple buffer.
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*/
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int nb_sample_units;
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SrsCodecSampleUnit sample_units[__SRS_SRS_MAX_CODEC_SAMPLE];
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public:
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/**
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* whether the sample is video sample which demux from video packet.
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*/
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bool is_video;
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/**
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* CompositionTime, video_file_format_spec_v10_1.pdf, page 78.
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* cts = pts - dts, where dts = flvheader->timestamp.
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*/
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int32_t cts;
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public:
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// video specified
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SrsCodecVideoAVCFrame frame_type;
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SrsCodecVideoAVCType avc_packet_type;
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public:
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// audio specified
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SrsCodecAudio acodec;
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// audio aac specified.
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SrsCodecAudioSampleRate sound_rate;
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SrsCodecAudioSampleSize sound_size;
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SrsCodecAudioSoundType sound_type;
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SrsCodecAudioType aac_packet_type;
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public:
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SrsCodecSample();
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virtual ~SrsCodecSample();
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public:
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/**
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* clear all samples.
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* the sample units never copy the bytes, it directly use the ptr,
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* so when video/audio packet is destroyed, the sample must be clear.
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* in a word, user must clear sample before demux it.
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* @remark demux sample use SrsAvcAacCodec.audio_aac_demux or video_avc_demux.
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*/
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void clear();
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/**
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* add the a sample unit, it's a h.264 NALU or aac raw data.
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* the sample unit directly use the ptr of packet bytes,
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* so user must never use sample unit when packet is destroyed.
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* in a word, user must clear sample before demux it.
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*/
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int add_sample_unit(char* bytes, int size);
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};
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/**
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* the h264/avc and aac codec, for media stream.
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*
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* to demux the FLV/RTMP video/audio packet to sample,
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* add each NALUs of h.264 as a sample unit to sample,
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* while the entire aac raw data as a sample unit.
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*
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* for sequence header,
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* demux it and save it in the avc_extra_data and aac_extra_data,
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*
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* for the codec info, such as audio sample rate,
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* decode from FLV/RTMP header, then use codec info in sequence
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* header to override it.
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*/
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class SrsAvcAacCodec
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{
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private:
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SrsStream* stream;
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public:
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/**
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* metadata specified
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*/
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int duration;
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int width;
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int height;
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int frame_rate;
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// @see: SrsCodecVideo
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int video_codec_id;
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int video_data_rate; // in bps
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// @see: SrsCod ecAudioType
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int audio_codec_id;
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int audio_data_rate; // in bps
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public:
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/**
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* video specified
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*/
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// profile_idc, H.264-AVC-ISO_IEC_14496-10.pdf, page 45.
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u_int8_t avc_profile;
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// level_idc, H.264-AVC-ISO_IEC_14496-10.pdf, page 45.
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u_int8_t avc_level;
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// lengthSizeMinusOne, H.264-AVC-ISO_IEC_14496-15.pdf, page 16
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int8_t NAL_unit_length;
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u_int16_t sequenceParameterSetLength;
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char* sequenceParameterSetNALUnit;
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u_int16_t pictureParameterSetLength;
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char* pictureParameterSetNALUnit;
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public:
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/**
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* audio specified
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* audioObjectType, in 1.6.2.1 AudioSpecificConfig, page 33,
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* 1.5.1.1 Audio object type definition, page 23,
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* in aac-mp4a-format-ISO_IEC_14496-3+2001.pdf.
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*/
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u_int8_t aac_profile;
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/**
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* samplingFrequencyIndex
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*/
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u_int8_t aac_sample_rate;
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/**
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* channelConfiguration
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*/
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u_int8_t aac_channels;
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public:
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/**
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* the avc extra data, the AVC sequence header,
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* without the flv codec header,
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* @see: ffmpeg, AVCodecContext::extradata
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*/
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int avc_extra_size;
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char* avc_extra_data;
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/**
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* the aac extra data, the AAC sequence header,
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* without the flv codec header,
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* @see: ffmpeg, AVCodecContext::extradata
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*/
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int aac_extra_size;
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char* aac_extra_data;
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public:
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SrsAvcAacCodec();
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virtual ~SrsAvcAacCodec();
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// the following function used for hls to build the sample and codec.
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public:
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/**
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* demux the audio packet in aac codec.
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* the packet mux in FLV/RTMP format defined in flv specification.
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* demux the audio speicified data(sound_format, sound_size, ...) to sample.
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* demux the aac specified data(aac_profile, ...) to codec from sequence header.
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* demux the aac raw to sample units.
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*/
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virtual int audio_aac_demux(char* data, int size, SrsCodecSample* sample);
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virtual int audio_mp3_demux(char* data, int size, SrsCodecSample* sample);
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/**
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* demux the video packet in h.264 codec.
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* the packet mux in FLV/RTMP format defined in flv specification.
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* demux the video specified data(frame_type, codec_id, ...) to sample.
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* demux the h.264 sepcified data(avc_profile, ...) to codec from sequence header.
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* demux the h.264 NALUs to sampe units.
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*/
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virtual int video_avc_demux(char* data, int size, SrsCodecSample* sample);
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private:
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/**
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* when avc packet type is SrsCodecVideoAVCTypeSequenceHeader,
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* decode the sps and pps.
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*/
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virtual int avc_demux_sps_pps(SrsStream* stream);
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/**
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* demux the avc NALU in "AnnexB"
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* from H.264-AVC-ISO_IEC_14496-10.pdf, page 211.
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*/
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virtual int avc_demux_annexb_format(SrsStream* stream, SrsCodecSample* sample);
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/**
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* demux the avc NALU in "ISO Base Media File Format"
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* from H.264-AVC-ISO_IEC_14496-15.pdf, page 20
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*/
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virtual int avc_demux_ibmf_format(SrsStream* stream, SrsCodecSample* sample);
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};
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#endif
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