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https://github.com/ZLMediaKit/ZLMediaKit.git
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精简aac相关代码
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@ -12,62 +12,55 @@
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namespace mediakit{
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void writeAdtsHeader(const AACFrame &hed, uint8_t *pcAdts) {
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pcAdts[0] = (hed.syncword >> 4 & 0xFF); //8bit
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pcAdts[1] = (hed.syncword << 4 & 0xF0); //4 bit
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pcAdts[1] |= (hed.id << 3 & 0x08); //1 bit
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pcAdts[1] |= (hed.layer << 1 & 0x06); //2bit
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pcAdts[1] |= (hed.protection_absent & 0x01); //1 bit
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class AdtsHeader{
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public:
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unsigned int syncword = 0; //12 bslbf 同步字The bit string ‘1111 1111 1111’,说明一个ADTS帧的开始
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unsigned int id; //1 bslbf MPEG 标示符, 设置为1
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unsigned int layer; //2 uimsbf Indicates which layer is used. Set to ‘00’
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unsigned int protection_absent; //1 bslbf 表示是否误码校验
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unsigned int profile; //2 uimsbf 表示使用哪个级别的AAC,如01 Low Complexity(LC)--- AACLC
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unsigned int sf_index; //4 uimsbf 表示使用的采样率下标
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unsigned int private_bit; //1 bslbf
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unsigned int channel_configuration; //3 uimsbf 表示声道数
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unsigned int original; //1 bslbf
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unsigned int home; //1 bslbf
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//下面的为改变的参数即每一帧都不同
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unsigned int copyright_identification_bit; //1 bslbf
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unsigned int copyright_identification_start; //1 bslbf
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unsigned int aac_frame_length; // 13 bslbf 一个ADTS帧的长度包括ADTS头和raw data block
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unsigned int adts_buffer_fullness; //11 bslbf 0x7FF 说明是码率可变的码流
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//no_raw_data_blocks_in_frame 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧.
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//所以说number_of_raw_data_blocks_in_frame == 0
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//表示说ADTS帧中有一个AAC数据块并不是说没有。(一个AAC原始帧包含一段时间内1024个采样及相关数据)
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unsigned int no_raw_data_blocks_in_frame; //2 uimsfb
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};
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pcAdts[2] = (hed.profile << 6 & 0xC0); // 2 bit
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pcAdts[2] |= (hed.sf_index << 2 & 0x3C); //4bit
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pcAdts[2] |= (hed.private_bit << 1 & 0x02); //1 bit
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pcAdts[2] |= (hed.channel_configuration >> 2 & 0x03); //1 bit
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pcAdts[3] = (hed.channel_configuration << 6 & 0xC0); // 2 bit
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pcAdts[3] |= (hed.original << 5 & 0x20); //1 bit
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pcAdts[3] |= (hed.home << 4 & 0x10); //1 bit
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pcAdts[3] |= (hed.copyright_identification_bit << 3 & 0x08); //1 bit
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pcAdts[3] |= (hed.copyright_identification_start << 2 & 0x04); //1 bit
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pcAdts[3] |= (hed.aac_frame_length >> 11 & 0x03); //2 bit
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pcAdts[4] = (hed.aac_frame_length >> 3 & 0xFF); //8 bit
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pcAdts[5] = (hed.aac_frame_length << 5 & 0xE0); //3 bit
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pcAdts[5] |= (hed.adts_buffer_fullness >> 6 & 0x1F); //5 bit
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pcAdts[6] = (hed.adts_buffer_fullness << 2 & 0xFC); //6 bit
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pcAdts[6] |= (hed.no_raw_data_blocks_in_frame & 0x03); //2 bit
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static void dumpAdtsHeader(const AdtsHeader &hed, uint8_t *out) {
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out[0] = (hed.syncword >> 4 & 0xFF); //8bit
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out[1] = (hed.syncword << 4 & 0xF0); //4 bit
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out[1] |= (hed.id << 3 & 0x08); //1 bit
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out[1] |= (hed.layer << 1 & 0x06); //2bit
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out[1] |= (hed.protection_absent & 0x01); //1 bit
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out[2] = (hed.profile << 6 & 0xC0); // 2 bit
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out[2] |= (hed.sf_index << 2 & 0x3C); //4bit
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out[2] |= (hed.private_bit << 1 & 0x02); //1 bit
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out[2] |= (hed.channel_configuration >> 2 & 0x03); //1 bit
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out[3] = (hed.channel_configuration << 6 & 0xC0); // 2 bit
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out[3] |= (hed.original << 5 & 0x20); //1 bit
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out[3] |= (hed.home << 4 & 0x10); //1 bit
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out[3] |= (hed.copyright_identification_bit << 3 & 0x08); //1 bit
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out[3] |= (hed.copyright_identification_start << 2 & 0x04); //1 bit
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out[3] |= (hed.aac_frame_length >> 11 & 0x03); //2 bit
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out[4] = (hed.aac_frame_length >> 3 & 0xFF); //8 bit
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out[5] = (hed.aac_frame_length << 5 & 0xE0); //3 bit
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out[5] |= (hed.adts_buffer_fullness >> 6 & 0x1F); //5 bit
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out[6] = (hed.adts_buffer_fullness << 2 & 0xFC); //6 bit
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out[6] |= (hed.no_raw_data_blocks_in_frame & 0x03); //2 bit
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}
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string makeAdtsConfig(const uint8_t *pcAdts){
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if (!(pcAdts[0] == 0xFF && (pcAdts[1] & 0xF0) == 0xF0)) {
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return "";
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}
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// Get and check the 'profile':
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unsigned char profile = (pcAdts[2] & 0xC0) >> 6; // 2 bits
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if (profile == 3) {
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return "";
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}
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// Get and check the 'sampling_frequency_index':
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unsigned char sampling_frequency_index = (pcAdts[2] & 0x3C) >> 2; // 4 bits
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if (samplingFrequencyTable[sampling_frequency_index] == 0) {
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return "";
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}
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// Get and check the 'channel_configuration':
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unsigned char channel_configuration = ((pcAdts[2] & 0x01) << 2)
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| ((pcAdts[3] & 0xC0) >> 6); // 3 bits
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unsigned char audioSpecificConfig[2];
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unsigned char const audioObjectType = profile + 1;
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audioSpecificConfig[0] = (audioObjectType << 3) | (sampling_frequency_index >> 1);
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audioSpecificConfig[1] = (sampling_frequency_index << 7) | (channel_configuration << 3);
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return string((char *)audioSpecificConfig,2);
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}
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void makeAdtsHeader(const string &strAudioCfg,AACFrame &adts) {
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uint8_t cfg1 = strAudioCfg[0];
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uint8_t cfg2 = strAudioCfg[1];
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static void parseAacConfig(const string &config, AdtsHeader &adts) {
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uint8_t cfg1 = config[0];
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uint8_t cfg2 = config[1];
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int audioObjectType;
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int sampling_frequency_index;
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@ -93,9 +86,44 @@ void makeAdtsHeader(const string &strAudioCfg,AACFrame &adts) {
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adts.adts_buffer_fullness = 2047;
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adts.no_raw_data_blocks_in_frame = 0;
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}
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void getAACInfo(const AACFrame &adts,int &iSampleRate,int &iChannel){
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iSampleRate = samplingFrequencyTable[adts.sf_index];
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iChannel = adts.channel_configuration;
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string makeAacConfig(const uint8_t *hex){
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if (!(hex[0] == 0xFF && (hex[1] & 0xF0) == 0xF0)) {
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return "";
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}
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// Get and check the 'profile':
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unsigned char profile = (hex[2] & 0xC0) >> 6; // 2 bits
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if (profile == 3) {
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return "";
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}
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// Get and check the 'sampling_frequency_index':
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unsigned char sampling_frequency_index = (hex[2] & 0x3C) >> 2; // 4 bits
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if (samplingFrequencyTable[sampling_frequency_index] == 0) {
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return "";
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}
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// Get and check the 'channel_configuration':
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unsigned char channel_configuration = ((hex[2] & 0x01) << 2) | ((hex[3] & 0xC0) >> 6); // 3 bits
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unsigned char audioSpecificConfig[2];
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unsigned char const audioObjectType = profile + 1;
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audioSpecificConfig[0] = (audioObjectType << 3) | (sampling_frequency_index >> 1);
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audioSpecificConfig[1] = (sampling_frequency_index << 7) | (channel_configuration << 3);
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return string((char *)audioSpecificConfig,2);
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}
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void dumpAacConfig(const string &config, int length, uint8_t *out){
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AdtsHeader header;
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parseAacConfig(config, header);
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header.aac_frame_length = length;
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dumpAdtsHeader(header, out);
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}
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void parseAacConfig(const string &config, int &samplerate, int &channels){
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AdtsHeader header;
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parseAacConfig(config, header);
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samplerate = samplingFrequencyTable[header.sf_index];
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channels = header.channel_configuration;
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}
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Sdp::Ptr AACTrack::getSdp() {
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@ -106,6 +134,4 @@ Sdp::Ptr AACTrack::getSdp() {
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return std::make_shared<AACSdp>(getAacCfg(),getAudioSampleRate(), getAudioChannel());
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}
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}//namespace mediakit
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}//namespace mediakit
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@ -13,71 +13,26 @@
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#include "Frame.h"
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#include "Track.h"
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#define ADTS_HEADER_LEN 7
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namespace mediakit{
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class AACFrame;
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unsigned const samplingFrequencyTable[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0 };
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void makeAdtsHeader(const string &strAudioCfg,AACFrame &adts);
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void writeAdtsHeader(const AACFrame &adts, uint8_t *pcAdts) ;
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string makeAdtsConfig(const uint8_t *pcAdts);
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void getAACInfo(const AACFrame &adts,int &iSampleRate,int &iChannel);
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string makeAacConfig(const uint8_t *hex);
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void dumpAacConfig(const string &config, int length, uint8_t *out);
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void parseAacConfig(const string &config, int &samplerate, int &channels);
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/**
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* aac帧,包含adts头
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*/
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class AACFrame : public Frame {
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class AACFrame : public FrameImp {
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public:
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typedef std::shared_ptr<AACFrame> Ptr;
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char *data() const override{
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return (char *)buffer;
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AACFrame(){
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_codecid = CodecAAC;
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}
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uint32_t size() const override {
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return aac_frame_length;
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}
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uint32_t dts() const override {
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return _dts;
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}
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uint32_t prefixSize() const override{
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return _prefix_size;
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}
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CodecId getCodecId() const override{
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return CodecAAC;
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}
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bool keyFrame() const override {
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return false;
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}
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bool configFrame() const override{
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return false;
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}
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public:
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unsigned int syncword = 0; //12 bslbf 同步字The bit string ‘1111 1111 1111’,说明一个ADTS帧的开始
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unsigned int id; //1 bslbf MPEG 标示符, 设置为1
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unsigned int layer; //2 uimsbf Indicates which layer is used. Set to ‘00’
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unsigned int protection_absent; //1 bslbf 表示是否误码校验
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unsigned int profile; //2 uimsbf 表示使用哪个级别的AAC,如01 Low Complexity(LC)--- AACLC
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unsigned int sf_index; //4 uimsbf 表示使用的采样率下标
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unsigned int private_bit; //1 bslbf
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unsigned int channel_configuration; //3 uimsbf 表示声道数
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unsigned int original; //1 bslbf
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unsigned int home; //1 bslbf
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//下面的为改变的参数即每一帧都不同
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unsigned int copyright_identification_bit; //1 bslbf
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unsigned int copyright_identification_start; //1 bslbf
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unsigned int aac_frame_length; // 13 bslbf 一个ADTS帧的长度包括ADTS头和raw data block
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unsigned int adts_buffer_fullness; //11 bslbf 0x7FF 说明是码率可变的码流
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//no_raw_data_blocks_in_frame 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧.
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//所以说number_of_raw_data_blocks_in_frame == 0
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//表示说ADTS帧中有一个AAC数据块并不是说没有。(一个AAC原始帧包含一段时间内1024个采样及相关数据)
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unsigned int no_raw_data_blocks_in_frame; //2 uimsfb
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unsigned char buffer[2 * 1024 + 7];
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uint32_t _dts;
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uint32_t _prefix_size = 7;
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};
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class AACFrameNoCacheAble : public FrameFromPtr {
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@ -138,7 +93,7 @@ public:
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if(adts_header_len < 7){
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throw std::invalid_argument("adts头必须不少于7个字节");
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}
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_cfg = makeAdtsConfig((uint8_t*)adts_header);
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_cfg = makeAacConfig((uint8_t *) adts_header);
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onReady();
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}
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@ -150,7 +105,7 @@ public:
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if(aac_frame_with_adts->getCodecId() != CodecAAC || aac_frame_with_adts->prefixSize() < 7){
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throw std::invalid_argument("必须输入带adts头的aac帧");
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}
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_cfg = makeAdtsConfig((uint8_t*)aac_frame_with_adts->data());
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_cfg = makeAacConfig((uint8_t *) aac_frame_with_adts->data());
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onReady();
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}
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@ -205,7 +160,7 @@ public:
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//未获取到aac_cfg信息
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if (frame->prefixSize() >= 7) {
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//7个字节的adts头
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_cfg = makeAdtsConfig((uint8_t *)(frame->data()));
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_cfg = makeAacConfig((uint8_t *) (frame->data()));
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onReady();
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} else {
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WarnL << "无法获取adts头!";
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@ -218,12 +173,10 @@ private:
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* 解析2个字节的aac配置
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*/
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void onReady(){
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if(_cfg.size() < 2){
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if (_cfg.size() < 2) {
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return;
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}
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AACFrame aacFrame;
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makeAdtsHeader(_cfg,aacFrame);
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getAACInfo(aacFrame,_sampleRate,_channel);
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parseAacConfig(_cfg, _sampleRate, _channel);
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}
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Track::Ptr clone() override {
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@ -20,8 +20,9 @@ AACRtmpDecoder::AACRtmpDecoder() {
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AACFrame::Ptr AACRtmpDecoder::obtainFrame() {
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//从缓存池重新申请对象,防止覆盖已经写入环形缓存的对象
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auto frame = ResourcePoolHelper<AACFrame>::obtainObj();
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frame->aac_frame_length = 7;
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frame->_prefix_size = 7;
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frame->_prefix_size = ADTS_HEADER_LEN;
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//预留7个字节的空位以便后续覆盖
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frame->_buffer.assign(ADTS_HEADER_LEN,(char)0);
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return frame;
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}
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@ -41,7 +42,7 @@ static string getAacCfg(const RtmpPacket &thiz) {
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return ret;
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}
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bool AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool key_pos) {
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bool AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool) {
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if (pkt->isCfgFrame()) {
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_aac_cfg = getAacCfg(*pkt);
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return false;
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@ -52,26 +53,18 @@ bool AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool key_pos) {
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return false;
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}
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void AACRtmpDecoder::onGetAAC(const char* pcData, int iLen, uint32_t ui32TimeStamp) {
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if(iLen + 7 > sizeof(_adts->buffer)){
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WarnL << "Illegal adts data, exceeding the length limit.";
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return;
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}
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//写adts结构头
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makeAdtsHeader(_aac_cfg,*_adts);
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//拷贝aac负载
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memcpy(_adts->buffer + 7, pcData, iLen);
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_adts->aac_frame_length = 7 + iLen;
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_adts->_dts = ui32TimeStamp;
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//adts结构头转成头7个字节
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writeAdtsHeader(*_adts, _adts->buffer);
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void AACRtmpDecoder::onGetAAC(const char* data, int len, uint32_t stamp) {
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_adts->_dts = stamp;
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//先追加数据
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_adts->_buffer.append(data, len);
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//覆盖adts头
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dumpAacConfig(_aac_cfg, _adts->size(), (uint8_t *) _adts->data());
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//写入环形缓存
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RtmpCodec::inputFrame(_adts);
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_adts = obtainFrame();
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}
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/////////////////////////////////////////////////////////////////////////////////////
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AACRtmpEncoder::AACRtmpEncoder(const Track::Ptr &track) {
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@ -93,7 +86,7 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
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if (_aac_cfg.empty()) {
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if (frame->prefixSize() >= 7) {
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//包含adts头,从adts头获取aac配置信息
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_aac_cfg = makeAdtsConfig((uint8_t *)(frame->data()));
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_aac_cfg = makeAacConfig((uint8_t *) (frame->data()));
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}
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makeConfigPacket();
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}
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@ -38,7 +38,7 @@ public:
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}
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protected:
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void onGetAAC(const char* pcData, int iLen, uint32_t ui32TimeStamp);
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void onGetAAC(const char* data, int len, uint32_t stamp);
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AACFrame::Ptr obtainFrame();
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protected:
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AACFrame::Ptr _adts;
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*/
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#include "AACRtp.h"
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#define ADTS_HEADER_LEN 7
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#define AAC_MAX_FRAME_SIZE (2 * 1024)
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namespace mediakit{
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@ -77,11 +77,9 @@ AACRtpDecoder::AACRtpDecoder() {
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AACFrame::Ptr AACRtpDecoder::obtainFrame() {
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//从缓存池重新申请对象,防止覆盖已经写入环形缓存的对象
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auto frame = ResourcePoolHelper<AACFrame>::obtainObj();
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frame->aac_frame_length = ADTS_HEADER_LEN;
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frame->_prefix_size = ADTS_HEADER_LEN;
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if(frame->syncword == 0 && !_aac_cfg.empty()) {
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makeAdtsHeader(_aac_cfg,*frame);
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}
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//预留7个字节的空位以便后续覆盖
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frame->_buffer.assign(ADTS_HEADER_LEN,(char)0);
|
||||
return frame;
|
||||
}
|
||||
|
||||
@ -96,19 +94,18 @@ bool AACRtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
|
||||
//忽略Au-Header区
|
||||
ptr += 2 + au_header_count * 2;
|
||||
|
||||
static const uint32_t max_size = sizeof(AACFrame::buffer) - ADTS_HEADER_LEN;
|
||||
static const uint32_t max_size = AAC_MAX_FRAME_SIZE - ADTS_HEADER_LEN;
|
||||
while (ptr < end) {
|
||||
auto size = (uint32_t) (end - ptr);
|
||||
if(size > max_size){
|
||||
size = max_size;
|
||||
}
|
||||
if (_adts->aac_frame_length + size > sizeof(AACFrame::buffer)) {
|
||||
if (_adts->size() + size > AAC_MAX_FRAME_SIZE) {
|
||||
//数据太多了,先清空
|
||||
flushData();
|
||||
}
|
||||
//追加aac数据
|
||||
memcpy(_adts->buffer + _adts->aac_frame_length, ptr, size);
|
||||
_adts->aac_frame_length += size;
|
||||
_adts->_buffer.append((char *)ptr, size);
|
||||
_adts->_dts = rtppack->timeStamp;
|
||||
ptr += size;
|
||||
}
|
||||
@ -120,13 +117,14 @@ bool AACRtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
|
||||
return false;
|
||||
}
|
||||
|
||||
|
||||
void AACRtpDecoder::flushData() {
|
||||
if(_adts->aac_frame_length == ADTS_HEADER_LEN){
|
||||
if (_adts->size() == ADTS_HEADER_LEN) {
|
||||
//没有有效数据
|
||||
return;
|
||||
}
|
||||
writeAdtsHeader(*_adts, _adts->buffer);
|
||||
|
||||
//覆盖adts头
|
||||
dumpAacConfig(_aac_cfg, _adts->size(), (uint8_t *) _adts->data());
|
||||
RtpCodec::inputFrame(_adts);
|
||||
_adts = obtainFrame();
|
||||
}
|
||||
|
Loading…
Reference in New Issue
Block a user