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for bug #293, support http aac stream
This commit is contained in:
parent
7674524c31
commit
3244c083c7
4 changed files with 148 additions and 10 deletions
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@ -214,9 +214,10 @@ int SrsAacStreamEncoder::write_audio(int64_t timestamp, char* data, int size)
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return enc->write_audio(timestamp, data, size);
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}
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int SrsAacStreamEncoder::write_video(int64_t timestamp, char* data, int size)
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int SrsAacStreamEncoder::write_video(int64_t /*timestamp*/, char* /*data*/, int /*size*/)
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{
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return enc->write_video(timestamp, data, size);
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// aac ignore any flv video.
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return ERROR_SUCCESS;
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}
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int SrsAacStreamEncoder::write_metadata(int64_t /*timestamp*/, char* /*data*/, int /*size*/)
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@ -318,6 +319,9 @@ int SrsLiveStream::serve_http(ISrsGoHttpResponseWriter* w, SrsHttpMessage* r)
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if (serve_flv_streaming) {
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enc = new SrsFlvStreamEncoder();
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}
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if (serve_aac_streaming) {
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enc = new SrsAacStreamEncoder();
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}
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SrsAutoFree(ISrsStreamEncoder, enc);
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if ((ret = enc->initialize(&writer)) != ERROR_SUCCESS) {
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@ -36,6 +36,7 @@ using namespace std;
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#include <srs_kernel_error.hpp>
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#include <srs_kernel_stream.hpp>
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#include <srs_kernel_file.hpp>
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#include <srs_kernel_codec.hpp>
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#define SRS_FLV_TAG_HEADER_SIZE 11
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#define SRS_FLV_PREVIOUS_TAG_SIZE 4
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@ -43,6 +44,7 @@ using namespace std;
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SrsAacEncoder::SrsAacEncoder()
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{
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_fs = NULL;
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got_sequence_header = false;
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tag_stream = new SrsStream();
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}
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@ -76,16 +78,143 @@ int SrsAacEncoder::write_audio(int64_t timestamp, char* data, int size)
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timestamp &= 0x7fffffff;
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SrsStream* stream = tag_stream;
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if ((ret = stream->initialize(data, size)) != ERROR_SUCCESS) {
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return ret;
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}
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int SrsAacEncoder::write_video(int64_t timestamp, char* data, int size)
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{
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int ret = ERROR_SUCCESS;
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// audio decode
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if (!stream->require(1)) {
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ret = ERROR_AAC_DECODE_ERROR;
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srs_error("aac decode audio sound_format failed. ret=%d", ret);
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return ret;
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}
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srs_assert(data);
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// @see: E.4.2 Audio Tags, video_file_format_spec_v10_1.pdf, page 76
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int8_t sound_format = stream->read_1bytes();
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timestamp &= 0x7fffffff;
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// @see: SrsAvcAacCodec::audio_aac_demux
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//int8_t sound_type = sound_format & 0x01;
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//int8_t sound_size = (sound_format >> 1) & 0x01;
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//int8_t sound_rate = (sound_format >> 2) & 0x03;
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sound_format = (sound_format >> 4) & 0x0f;
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if ((SrsCodecAudio)sound_format != SrsCodecAudioAAC) {
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ret = ERROR_AAC_DECODE_ERROR;
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srs_error("aac required, format=%d. ret=%d", sound_format, ret);
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return ret;
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}
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if (!stream->require(1)) {
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ret = ERROR_AAC_DECODE_ERROR;
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srs_error("aac decode aac_packet_type failed. ret=%d", ret);
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return ret;
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}
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SrsCodecAudioType aac_packet_type = (SrsCodecAudioType)stream->read_1bytes();
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if (aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
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// AudioSpecificConfig
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// 1.6.2.1 AudioSpecificConfig, in aac-mp4a-format-ISO_IEC_14496-3+2001.pdf, page 33.
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//
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// only need to decode the first 2bytes:
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// audioObjectType, aac_profile, 5bits.
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// samplingFrequencyIndex, aac_sample_rate, 4bits.
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// channelConfiguration, aac_channels, 4bits
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if (!stream->require(2)) {
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ret = ERROR_AAC_DECODE_ERROR;
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srs_error("aac decode sequence header failed. ret=%d", ret);
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return ret;
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}
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aac_profile = stream->read_1bytes();
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aac_sample_rate = stream->read_1bytes();
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aac_channels = (aac_sample_rate >> 3) & 0x0f;
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aac_sample_rate = ((aac_profile << 1) & 0x0e) | ((aac_sample_rate >> 7) & 0x01);
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aac_profile = (aac_profile >> 3) & 0x1f;
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got_sequence_header = true;
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return ret;
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}
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if (!got_sequence_header) {
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ret = ERROR_AAC_DECODE_ERROR;
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srs_error("aac no sequence header. ret=%d", ret);
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return ret;
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}
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// the left is the aac raw frame data.
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int16_t aac_frame_length = stream->size() - stream->pos();
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// write the ADTS header.
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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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char aac_fixed_header[7];
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if(true) {
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char* pp = aac_fixed_header;
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// Syncword 12 bslbf
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*pp++ = 0xff;
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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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*pp++ = 0xf1;
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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 = aac_profile - 1;
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*pp++ = ((fh_Profile_ObjectType << 6) & 0xc0) | ((aac_sample_rate << 2) & 0x3c) | ((aac_channels >> 2) & 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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// 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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*pp++ = ((aac_channels << 6) & 0xc0) | ((aac_frame_length >> 11) & 0x03);
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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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*pp++ = aac_frame_length >> 3;
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// adts_buffer_fullness 11 bslbf
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*pp++ = (aac_frame_length << 5) & 0xe0;
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// no_raw_data_blocks_in_frame 2 uimsbf
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*pp++ = 0xfc;
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}
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// write 7bytes fixed header.
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if ((ret = _fs->write(aac_fixed_header, 7, NULL)) != ERROR_SUCCESS) {
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return ret;
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}
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// write aac frame body.
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if ((ret = _fs->write(data + stream->pos(), aac_frame_length, NULL)) != ERROR_SUCCESS) {
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return ret;
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}
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return ret;
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}
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@ -42,6 +42,11 @@ class SrsAacEncoder
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{
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private:
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SrsFileWriter* _fs;
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private:
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int8_t aac_profile;
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int8_t aac_sample_rate;
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int8_t aac_channels;
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bool got_sequence_header;
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private:
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SrsStream* tag_stream;
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public:
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@ -60,7 +65,6 @@ public:
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* @remark assert data is not NULL.
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*/
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virtual int write_audio(int64_t timestamp, char* data, int size);
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virtual int write_video(int64_t timestamp, char* data, int size);
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};
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#endif
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@ -204,6 +204,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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#define ERROR_AAC_DATA_INVALID 3048
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#define ERROR_HTTP_STATUS_INVLIAD 3049
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#define ERROR_KERNEL_AAC_STREAM_CLOSED 3050
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#define ERROR_AAC_DECODE_ERROR 3051
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///////////////////////////////////////////////////////
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// HTTP protocol error.
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