mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-26 20:47:08 +08:00
348 lines
12 KiB
C++
348 lines
12 KiB
C++
/*
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* Copyright (c) 2016 The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/xia-chu/ZLMediaKit).
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*
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* Use of this source code is governed by MIT license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#include "AAC.h"
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#ifdef ENABLE_MP4
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#include "mpeg4-aac.h"
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#endif
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using namespace std;
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using namespace toolkit;
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namespace mediakit{
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#ifndef ENABLE_MP4
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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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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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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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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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int channel_configuration;
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audioObjectType = cfg1 >> 3;
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sampling_frequency_index = ((cfg1 & 0x07) << 1) | (cfg2 >> 7);
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channel_configuration = (cfg2 & 0x7F) >> 3;
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adts.syncword = 0x0FFF;
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adts.id = 0;
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adts.layer = 0;
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adts.protection_absent = 1;
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adts.profile = audioObjectType - 1;
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adts.sf_index = sampling_frequency_index;
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adts.private_bit = 0;
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adts.channel_configuration = channel_configuration;
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adts.original = 0;
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adts.home = 0;
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adts.copyright_identification_bit = 0;
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adts.copyright_identification_start = 0;
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adts.aac_frame_length = 7;
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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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#endif// ENABLE_MP4
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int getAacFrameLength(const uint8_t *data, size_t bytes) {
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uint16_t len;
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if (bytes < 7) return -1;
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if (0xFF != data[0] || 0xF0 != (data[1] & 0xF0)) {
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return -1;
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}
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len = ((uint16_t) (data[3] & 0x03) << 11) | ((uint16_t) data[4] << 3) | ((uint16_t) (data[5] >> 5) & 0x07);
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return len;
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}
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string makeAacConfig(const uint8_t *hex, size_t length){
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#ifndef ENABLE_MP4
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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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#else
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struct mpeg4_aac_t aac;
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memset(&aac, 0, sizeof(aac));
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if (mpeg4_aac_adts_load(hex, length, &aac) > 0) {
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char buf[32] = {0};
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int len = mpeg4_aac_audio_specific_config_save(&aac, (uint8_t *) buf, sizeof(buf));
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if (len > 0) {
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return string(buf, len);
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}
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}
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WarnL << "生成aac config失败, adts header:" << hexdump(hex, length);
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return "";
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#endif
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}
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int dumpAacConfig(const string &config, size_t length, uint8_t *out, size_t out_size) {
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#ifndef ENABLE_MP4
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AdtsHeader header;
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parseAacConfig(config, header);
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header.aac_frame_length = (decltype(header.aac_frame_length))(ADTS_HEADER_LEN + length);
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dumpAdtsHeader(header, out);
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return ADTS_HEADER_LEN;
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#else
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struct mpeg4_aac_t aac;
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memset(&aac, 0, sizeof(aac));
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int ret = mpeg4_aac_audio_specific_config_load((uint8_t *) config.data(), config.size(), &aac);
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if (ret > 0) {
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ret = mpeg4_aac_adts_save(&aac, length, out, out_size);
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}
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if (ret < 0) {
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WarnL << "生成adts头失败:" << ret << ", aac config:" << hexdump(config.data(), config.size());
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}
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assert((int)out_size >= ret);
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return ret;
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#endif
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}
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bool parseAacConfig(const string &config, int &samplerate, int &channels){
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#ifndef ENABLE_MP4
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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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return true;
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#else
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struct mpeg4_aac_t aac;
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memset(&aac, 0, sizeof(aac));
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int ret = mpeg4_aac_audio_specific_config_load((uint8_t *) config.data(), config.size(), &aac);
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if (ret > 0) {
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samplerate = aac.sampling_frequency;
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channels = aac.channels;
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return true;
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}
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WarnL << "获取aac采样率、声道数失败:" << hexdump(config.data(), config.size());
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return false;
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#endif
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}
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////////////////////////////////////////////////////////////////////////////////////////////////////
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/**
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* aac类型SDP
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*/
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class AACSdp : public Sdp {
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public:
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/**
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* 构造函数
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* @param aac_cfg aac两个字节的配置描述
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* @param sample_rate 音频采样率
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* @param payload_type rtp payload type 默认98
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* @param bitrate 比特率
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*/
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AACSdp(const string &aac_cfg,
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int sample_rate,
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int channels,
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int bitrate = 128,
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int payload_type = 98) : Sdp(sample_rate,payload_type){
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_printer << "m=audio 0 RTP/AVP " << payload_type << "\r\n";
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if (bitrate) {
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_printer << "b=AS:" << bitrate << "\r\n";
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}
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_printer << "a=rtpmap:" << payload_type << " " << getCodecName() << "/" << sample_rate << "/" << channels << "\r\n";
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string configStr;
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char buf[4] = {0};
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for(auto &ch : aac_cfg){
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snprintf(buf, sizeof(buf), "%02X", (uint8_t)ch);
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configStr.append(buf);
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}
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_printer << "a=fmtp:" << payload_type << " streamtype=5;profile-level-id=1;mode=AAC-hbr;"
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<< "sizelength=13;indexlength=3;indexdeltalength=3;config=" << configStr << "\r\n";
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_printer << "a=control:trackID=" << (int)TrackAudio << "\r\n";
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}
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string getSdp() const override {
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return _printer;
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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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private:
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_StrPrinter _printer;
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};
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////////////////////////////////////////////////////////////////////////////////////////////////////
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AACTrack::AACTrack(const string &aac_cfg) {
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if (aac_cfg.size() < 2) {
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throw std::invalid_argument("adts配置必须最少2个字节");
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}
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_cfg = aac_cfg;
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onReady();
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}
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const string &AACTrack::getAacCfg() const {
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return _cfg;
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}
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CodecId AACTrack::getCodecId() const {
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return CodecAAC;
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}
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bool AACTrack::ready() {
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return !_cfg.empty();
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}
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int AACTrack::getAudioSampleRate() const {
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return _sampleRate;
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}
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int AACTrack::getAudioSampleBit() const {
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return _sampleBit;
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}
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int AACTrack::getAudioChannel() const {
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return _channel;
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}
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bool AACTrack::inputFrame(const Frame::Ptr &frame) {
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if (!frame->prefixSize()) {
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return inputFrame_l(frame);
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}
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bool ret = false;
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//有adts头，尝试分帧
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int64_t dts = frame->dts();
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int64_t pts = frame->pts();
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auto ptr = frame->data();
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auto end = frame->data() + frame->size();
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while (ptr < end) {
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auto frame_len = getAacFrameLength((uint8_t *) ptr, end - ptr);
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if (frame_len < ADTS_HEADER_LEN) {
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break;
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}
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if (frame_len == (int)frame->size()) {
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return inputFrame_l(frame);
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}
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auto sub_frame = std::make_shared<FrameTSInternal<FrameFromPtr> >(frame, (char *) ptr, frame_len, ADTS_HEADER_LEN,dts,pts);
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ptr += frame_len;
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if (ptr > end) {
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WarnL << "invalid aac length in adts header: " << frame_len
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<< ", remain data size: " << end - (ptr - frame_len);
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break;
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}
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sub_frame->setCodecId(CodecAAC);
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if (inputFrame_l(sub_frame)) {
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ret = true;
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}
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dts += 1024*1000/getAudioSampleRate();
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pts += 1024*1000/getAudioSampleRate();
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}
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return ret;
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}
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bool AACTrack::inputFrame_l(const Frame::Ptr &frame) {
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if (_cfg.empty()) {
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//未获取到aac_cfg信息
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if (frame->prefixSize()) {
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//根据7个字节的adts头生成aac config
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_cfg = makeAacConfig((uint8_t *) (frame->data()), frame->prefixSize());
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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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}
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}
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if (frame->size() > frame->prefixSize()) {
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//除adts头外，有实际负载
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return AudioTrack::inputFrame(frame);
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}
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return false;
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}
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void AACTrack::onReady() {
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if (_cfg.size() < 2) {
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return;
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}
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parseAacConfig(_cfg, _sampleRate, _channel);
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}
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Track::Ptr AACTrack::clone() {
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return std::make_shared<std::remove_reference<decltype(*this)>::type>(*this);
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}
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Sdp::Ptr AACTrack::getSdp() {
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if(!ready()){
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WarnL << getCodecName() << " Track未准备好";
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return nullptr;
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}
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return std::make_shared<AACSdp>(getAacCfg(), getAudioSampleRate(), getAudioChannel(), getBitRate() / 1024);
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}
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}//namespace mediakit
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