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fix #250, support push MPEGTS over UDP to SRS. 2.0.111
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winlin
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4246be92c9
commit
16afe7ddbb
7 changed files with 508 additions and 252 deletions
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@ -81,9 +81,10 @@ struct Context
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SimpleSocketStream* skt;
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int stream_id;
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// for h264 raw stream,
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// @see: https://github.com/winlinvip/simple-rtmp-server/issues/66#issuecomment-62240521
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// the remux raw codec.
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SrsRawH264Stream avc_raw;
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SrsRawAacStream aac_raw;
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// for h264 raw stream,
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// @see: https://github.com/winlinvip/simple-rtmp-server/issues/66#issuecomment-62240521
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SrsStream h264_raw_stream;
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@ -1073,32 +1074,15 @@ srs_bool srs_rtmp_is_onMetaData(char type, char* data, int size)
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* directly write a audio frame.
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*/
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int __srs_write_audio_raw_frame(Context* context,
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char sound_format, char sound_rate, char sound_size, char sound_type,
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char aac_packet_type, char* frame, int frame_size, u_int32_t timestamp
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char* frame, int frame_size, SrsRawAacStreamCodec* codec, u_int32_t timestamp
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) {
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// for audio frame, there is 1 or 2 bytes header:
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// 1bytes, SoundFormat|SoundRate|SoundSize|SoundType
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// 1bytes, AACPacketType for SoundFormat == 10, 0 is sequence header.
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int size = frame_size + 1;
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if (sound_format == SrsCodecAudioAAC) {
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size += 1;
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int ret = ERROR_SUCCESS;
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char* data = NULL;
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int size = 0;
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if ((ret = context->aac_raw.mux_aac2flv(frame, frame_size, codec, timestamp, &data, &size)) != ERROR_SUCCESS) {
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return ret;
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}
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char* data = new char[size];
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char* p = data;
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u_int8_t audio_header = sound_type & 0x01;
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audio_header |= (sound_size << 1) & 0x02;
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audio_header |= (sound_rate << 2) & 0x0c;
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audio_header |= (sound_format << 4) & 0xf0;
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*p++ = audio_header;
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if (sound_format == SrsCodecAudioAAC) {
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*p++ = aac_packet_type;
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}
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memcpy(p, frame, frame_size);
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return srs_rtmp_write_packet(context, SRS_RTMP_TYPE_AUDIO, timestamp, data, size);
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}
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@ -1106,73 +1090,28 @@ int __srs_write_audio_raw_frame(Context* context,
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/**
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* write aac frame in adts.
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*/
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int __srs_write_aac_adts_frame(Context* context,
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char sound_format, char sound_rate, char sound_size, char sound_type,
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char aac_profile, char aac_samplerate, char aac_channel,
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char* frame, int frame_size, u_int32_t timestamp
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int __srs_write_aac_adts_frame(Context* context,
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SrsRawAacStreamCodec* codec, char* frame, int frame_size, u_int32_t timestamp
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) {
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int ret = ERROR_SUCCESS;
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// override the aac samplerate by user specified.
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64146899
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switch (sound_rate) {
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case SrsCodecAudioSampleRate11025:
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aac_samplerate = 0x0a; break;
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case SrsCodecAudioSampleRate22050:
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aac_samplerate = 0x07; break;
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case SrsCodecAudioSampleRate44100:
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aac_samplerate = 0x04; break;
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default:
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break;
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}
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// send out aac sequence header if not sent.
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if (context->aac_specific_config.empty()) {
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char ch = 0;
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// @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf
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// AudioSpecificConfig (), page 33
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// 1.6.2.1 AudioSpecificConfig
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// audioObjectType; 5 bslbf
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ch = (aac_profile << 3) & 0xf8;
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// 3bits left.
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// samplingFrequencyIndex; 4 bslbf
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ch |= (aac_samplerate >> 1) & 0x07;
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context->aac_specific_config += ch;
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ch = (aac_samplerate << 7) & 0x80;
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if (aac_samplerate == 0x0f) {
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return ERROR_AAC_DATA_INVALID;
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std::string sh;
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if ((ret = context->aac_raw.mux_sequence_header(codec, sh)) != ERROR_SUCCESS) {
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return ret;
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}
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// 7bits left.
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// channelConfiguration; 4 bslbf
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ch |= (aac_channel << 3) & 0x78;
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// 3bits left.
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// only support aac profile 1-4.
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if (aac_profile < 1 || aac_profile > 4) {
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return ERROR_AAC_DATA_INVALID;
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}
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// GASpecificConfig(), page 451
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// 4.4.1 Decoder configuration (GASpecificConfig)
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// frameLengthFlag; 1 bslbf
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// dependsOnCoreCoder; 1 bslbf
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// extensionFlag; 1 bslbf
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context->aac_specific_config += ch;
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char* sh = (char*)context->aac_specific_config.data();
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int nb_sh = (int)context->aac_specific_config.length();
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if ((ret = __srs_write_audio_raw_frame(context,
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sound_format, sound_rate, sound_size, sound_type,
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0, sh, nb_sh, timestamp)) != ERROR_SUCCESS
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) {
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context->aac_specific_config = sh;
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codec->aac_packet_type = 0;
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if ((ret = __srs_write_audio_raw_frame(context, (char*)sh.data(), (int)sh.length(), codec, timestamp)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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return __srs_write_audio_raw_frame(context,
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sound_format, sound_rate, sound_size, sound_type,
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1, frame, frame_size, timestamp);
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codec->aac_packet_type = 1;
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return __srs_write_audio_raw_frame(context, frame, frame_size, codec, timestamp);
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}
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/**
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@ -1180,126 +1119,32 @@ int __srs_write_aac_adts_frame(Context* context,
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*/
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int __srs_write_aac_adts_frames(Context* context,
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char sound_format, char sound_rate, char sound_size, char sound_type,
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char* frame, int frame_size, u_int32_t timestamp
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char* frames, int frames_size, u_int32_t timestamp
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) {
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int ret = ERROR_SUCCESS;
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SrsStream* stream = &context->aac_raw_stream;
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if ((ret = stream->initialize(frame, frame_size)) != ERROR_SUCCESS) {
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if ((ret = stream->initialize(frames, frames_size)) != ERROR_SUCCESS) {
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return ret;
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}
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while (!stream->empty()) {
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int adts_header_start = stream->pos();
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// decode the ADTS.
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// @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 75,
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// 1.A.2.2 Audio_Data_Transport_Stream frame, ADTS
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64145885
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// byte_alignment()
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// adts_fixed_header:
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// 12bits syncword,
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// 16bits left.
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// adts_variable_header:
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// 28bits
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// 12+16+28=56bits
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// adts_error_check:
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// 16bits if protection_absent
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// 56+16=72bits
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// if protection_absent:
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// require(7bytes)=56bits
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// else
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// require(9bytes)=72bits
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if (!stream->require(7)) {
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return ERROR_AAC_ADTS_HEADER;
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}
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// for aac, the frame must be ADTS format.
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if (!srs_aac_startswith_adts(stream)) {
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return ERROR_AAC_REQUIRED_ADTS;
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}
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// Syncword 12 bslbf
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stream->read_1bytes();
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// 4bits left.
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// adts_fixed_header(), 1.A.2.2.1 Fixed Header of ADTS
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// ID 1 bslbf
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// Layer 2 uimsbf
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// protection_absent 1 bslbf
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int8_t fh0 = (stream->read_1bytes() & 0x0f);
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/*int8_t fh_id = (fh0 >> 3) & 0x01;*/
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/*int8_t fh_layer = (fh0 >> 1) & 0x03;*/
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int8_t fh_protection_absent = fh0 & 0x01;
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int16_t fh1 = stream->read_2bytes();
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// Profile_ObjectType 2 uimsbf
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// sampling_frequency_index 4 uimsbf
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// private_bit 1 bslbf
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// channel_configuration 3 uimsbf
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// original/copy 1 bslbf
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// home 1 bslbf
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int8_t fh_Profile_ObjectType = (fh1 >> 14) & 0x03;
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int8_t fh_sampling_frequency_index = (fh1 >> 10) & 0x0f;
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/*int8_t fh_private_bit = (fh1 >> 9) & 0x01;*/
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int8_t fh_channel_configuration = (fh1 >> 6) & 0x07;
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/*int8_t fh_original = (fh1 >> 5) & 0x01;*/
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/*int8_t fh_home = (fh1 >> 4) & 0x01;*/
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// @remark, Emphasis is removed,
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// @see https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-64154736
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//int8_t fh_Emphasis = (fh1 >> 2) & 0x03;
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// 4bits left.
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// adts_variable_header(), 1.A.2.2.2 Variable Header of ADTS
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// copyright_identification_bit 1 bslbf
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// copyright_identification_start 1 bslbf
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/*int8_t fh_copyright_identification_bit = (fh1 >> 3) & 0x01;*/
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/*int8_t fh_copyright_identification_start = (fh1 >> 2) & 0x01;*/
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// aac_frame_length 13 bslbf: Length of the frame including headers and error_check in bytes.
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// use the left 2bits as the 13 and 12 bit,
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// the aac_frame_length is 13bits, so we move 13-2=11.
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int16_t fh_aac_frame_length = (fh1 << 11) & 0x0800;
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int32_t fh2 = stream->read_3bytes();
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// aac_frame_length 13 bslbf: consume the first 13-2=11bits
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// the fh2 is 24bits, so we move right 24-11=13.
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fh_aac_frame_length |= (fh2 >> 13) & 0x07ff;
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// adts_buffer_fullness 11 bslbf
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/*int16_t fh_adts_buffer_fullness = (fh2 >> 2) & 0x7ff;*/
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// no_raw_data_blocks_in_frame 2 uimsbf
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/*int16_t fh_no_raw_data_blocks_in_frame = fh2 & 0x03;*/
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// adts_error_check(), 1.A.2.2.3 Error detection
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if (!fh_protection_absent) {
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if (!stream->require(2)) {
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return ERROR_AAC_ADTS_HEADER;
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}
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// crc_check 16 Rpchof
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/*int16_t crc_check = */stream->read_2bytes();
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}
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// TODO: check the fh_sampling_frequency_index
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// TODO: check the fh_channel_configuration
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// raw_data_blocks
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int adts_header_size = stream->pos() - adts_header_start;
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int raw_data_size = fh_aac_frame_length - adts_header_size;
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if (!stream->require(raw_data_size)) {
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return ERROR_AAC_ADTS_HEADER;
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}
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// the profile = object_id + 1
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// @see aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 78,
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// Table 1. A.9 – MPEG-2 Audio profiles and MPEG-4 Audio object types
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char aac_profile = fh_Profile_ObjectType + 1;
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char* raw_data = stream->data() + stream->pos();
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if ((ret = __srs_write_aac_adts_frame(context,
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sound_format, sound_rate, sound_size, sound_type,
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aac_profile, fh_sampling_frequency_index, fh_channel_configuration,
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raw_data, raw_data_size, timestamp)) != ERROR_SUCCESS
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) {
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char* frame = NULL;
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int frame_size = 0;
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SrsRawAacStreamCodec codec;
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if ((ret = context->aac_raw.adts_demux(stream, &frame, &frame_size, codec)) != ERROR_SUCCESS) {
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return ret;
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}
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// override by user specified.
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codec.sound_format = sound_format;
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codec.sound_rate = sound_rate;
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codec.sound_size = sound_size;
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codec.sound_type = sound_type;
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if ((ret = __srs_write_aac_adts_frame(context, &codec, frame, frame_size, timestamp)) != ERROR_SUCCESS) {
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return ret;
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}
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stream->skip(raw_data_size);
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}
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return ret;
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@ -1328,10 +1173,16 @@ int srs_audio_write_raw_frame(srs_rtmp_t rtmp,
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sound_format, sound_rate, sound_size, sound_type,
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frame, frame_size, timestamp);
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} else {
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// use codec info for aac.
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SrsRawAacStreamCodec codec;
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codec.sound_format = sound_format;
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codec.sound_rate = sound_rate;
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codec.sound_size = sound_size;
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codec.sound_type = sound_type;
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codec.aac_packet_type = 0;
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// for other data, directly write frame.
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return __srs_write_audio_raw_frame(context,
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sound_format, sound_rate, sound_size, sound_type,
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0, frame, frame_size, timestamp);
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return __srs_write_audio_raw_frame(context, frame, frame_size, &codec, timestamp);
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}
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