mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
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89e21038c2
# Conflicts: # src/Common/MediaSource.h # src/Common/config.h # src/Rtsp/RtspMediaSourceImp.h # src/Rtsp/RtspMediaSourceMuxer.h # src/Rtsp/RtspMuxer.cpp # webrtc/WebRtcTransport.cpp
470 lines
16 KiB
C++
470 lines
16 KiB
C++
/*
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT-like 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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#include "AACRtp.h"
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#include "AACRtmp.h"
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#include "Common/Parser.h"
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#include "Extension/Factory.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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unsigned const samplingFrequencyTable[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0 };
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#ifndef ENABLE_MP4
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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 [AUTO-TRANSLATED:481aa349]
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// The following are the parameters that change in each frame
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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. [AUTO-TRANSLATED:3e975531]
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// no_raw_data_blocks_in_frame indicates that there are number_of_raw_data_blocks_in_frame + 1 AAC raw frames in the ADTS frame.
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// 所以说number_of_raw_data_blocks_in_frame == 0 [AUTO-TRANSLATED:1b8e9697]
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// So 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) [AUTO-TRANSLATED:4a09d783]
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// means that there is one AAC data block in the ADTS frame, not that there is none. (An AAC raw frame contains 1024 samples and related data over a period of time)
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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 bool parseAacConfig(const string &config, AdtsHeader &adts) {
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if (config.size() < 2) {
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return false;
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}
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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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return true;
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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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if (!parseAacConfig(config, header)) {
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return false;
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}
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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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* aac type SDP
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* [AUTO-TRANSLATED:c06f00b1]
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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 payload_type rtp payload type
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* @param sample_rate 音频采样率
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* @param channels 通道数
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* @param bitrate 比特率
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* Constructor
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* @param aac_cfg aac two-byte configuration description
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* @param payload_type rtp payload type
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* @param sample_rate audio sampling rate
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* @param channels number of channels
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* @param bitrate bitrate
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* [AUTO-TRANSLATED:6fe1f3b2]
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*/
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AACSdp(const string &aac_cfg, int payload_type, int sample_rate, int channels, int bitrate)
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: 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(CodecAAC) << "/" << 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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}
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string getSdp() const override { return _printer; }
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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(int samplerate, int channel, int profile) : _channel(channel), _sampleRate(samplerate) {
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uint8_t audioSpecificConfig[2] = { 0 };
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uint8_t audioObjectType = profile+1;
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int samplingFrequencyIndex = 0;
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for (size_t i = 0; i < sizeof(samplingFrequencyTable) / sizeof(samplingFrequencyTable[0]); i++) {
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if (samplingFrequencyTable[i] == (unsigned)samplerate) {
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samplingFrequencyIndex = i;
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break;
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}
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}
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_cfg.resize(2);
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_cfg[0] = (audioObjectType << 3) | (samplingFrequencyIndex >> 1);
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_cfg[1] = (samplingFrequencyIndex << 7) | (channel << 3);
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// update();
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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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update();
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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() const {
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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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static Frame::Ptr addADTSHeader(const Frame::Ptr &frame_in, const std::string &aac_config) {
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auto frame = FrameImp::create();
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frame->_codec_id = CodecAAC;
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// 生成adts头 [AUTO-TRANSLATED:c285b9b0]
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// Generate adts header
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char adts_header[32] = { 0 };
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auto size = dumpAacConfig(aac_config, frame_in->size(), (uint8_t *)adts_header, sizeof(adts_header));
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CHECK(size > 0, "Invalid adts config");
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frame->_prefix_size = size;
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frame->_dts = frame_in->dts();
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frame->_buffer.assign(adts_header, size);
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frame->_buffer.append(frame_in->data(), frame_in->size());
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frame->setIndex(frame_in->getIndex());
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return frame;
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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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CHECK(ready());
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return inputFrame_l(addADTSHeader(frame, _cfg));
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}
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bool ret = false;
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// 有adts头，尝试分帧 [AUTO-TRANSLATED:f691c4ce]
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// There is an adts header, try to frame
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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<FrameInternalBase<FrameFromPtr>>(frame, (char *)ptr, frame_len, dts, pts, ADTS_HEADER_LEN);
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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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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() && frame->prefixSize()) {
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// 未获取到aac_cfg信息，根据7个字节的adts头生成aac config [AUTO-TRANSLATED:1b80f562]
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// Unable to get aac_cfg information, generate aac config based on the 7-byte adts header
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_cfg = makeAacConfig((uint8_t *)(frame->data()), frame->prefixSize());
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update();
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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 [AUTO-TRANSLATED:5b7c088e]
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// There is an actual payload besides the adts header
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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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toolkit::Buffer::Ptr AACTrack::getExtraData() const {
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CHECK(ready());
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return std::make_shared<BufferString>(_cfg);
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}
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void AACTrack::setExtraData(const uint8_t *data, size_t size) {
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CHECK(size >= 2);
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_cfg.assign((char *)data, size);
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update();
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}
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bool AACTrack::update() {
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return parseAacConfig(_cfg, _sampleRate, _channel);
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}
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Track::Ptr AACTrack::clone() const {
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return std::make_shared<AACTrack>(*this);
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}
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Sdp::Ptr AACTrack::getSdp(uint8_t payload_type) const {
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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>(getExtraData()->toString(), payload_type, getAudioSampleRate(), getAudioChannel(), getBitRate() / 1024);
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}
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namespace {
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CodecId getCodec() {
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return CodecAAC;
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}
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Track::Ptr getTrackByCodecId(int sample_rate, int channels, int sample_bit) {
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if (sample_rate && channels)
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return std::make_shared<AACTrack>(sample_rate, channels, 1);
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return std::make_shared<AACTrack>();
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}
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Track::Ptr getTrackBySdp(const SdpTrack::Ptr &track) {
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string aac_cfg_str = findSubString(track->_fmtp.data(), "config=", ";");
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if (aac_cfg_str.empty()) {
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aac_cfg_str = findSubString(track->_fmtp.data(), "config=", nullptr);
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}
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if (aac_cfg_str.empty()) {
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// 如果sdp中获取不到aac config信息，那么在rtp也无法获取，那么忽略该Track [AUTO-TRANSLATED:995bc20d]
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// If aac config information cannot be obtained from sdp, then it cannot be obtained from rtp either, so ignore this Track
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return nullptr;
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}
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string aac_cfg;
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for (size_t i = 0; i < aac_cfg_str.size() / 2; ++i) {
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unsigned int cfg;
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sscanf(aac_cfg_str.substr(i * 2, 2).data(), "%02X", &cfg);
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cfg &= 0x00FF;
|
||
aac_cfg.push_back((char)cfg);
|
||
}
|
||
return std::make_shared<AACTrack>(aac_cfg);
|
||
}
|
||
|
||
RtpCodec::Ptr getRtpEncoderByCodecId(uint8_t pt) {
|
||
return std::make_shared<AACRtpEncoder>();
|
||
}
|
||
|
||
RtpCodec::Ptr getRtpDecoderByCodecId() {
|
||
return std::make_shared<AACRtpDecoder>();
|
||
}
|
||
|
||
RtmpCodec::Ptr getRtmpEncoderByTrack(const Track::Ptr &track) {
|
||
return std::make_shared<AACRtmpEncoder>(track);
|
||
}
|
||
|
||
RtmpCodec::Ptr getRtmpDecoderByTrack(const Track::Ptr &track) {
|
||
return std::make_shared<AACRtmpDecoder>(track);
|
||
}
|
||
|
||
size_t aacPrefixSize(const char *data, size_t bytes) {
|
||
uint8_t *ptr = (uint8_t *)data;
|
||
size_t prefix = 0;
|
||
if (!(bytes > ADTS_HEADER_LEN && ptr[0] == 0xFF && (ptr[1] & 0xF0) == 0xF0)) {
|
||
return 0;
|
||
}
|
||
return ADTS_HEADER_LEN;
|
||
}
|
||
|
||
Frame::Ptr getFrameFromPtr(const char *data, size_t bytes, uint64_t dts, uint64_t pts) {
|
||
return std::make_shared<FrameFromPtr>(CodecAAC, (char *)data, bytes, dts, pts, aacPrefixSize(data, bytes));
|
||
}
|
||
|
||
} // namespace
|
||
|
||
CodecPlugin aac_plugin = { getCodec,
|
||
getTrackByCodecId,
|
||
getTrackBySdp,
|
||
getRtpEncoderByCodecId,
|
||
getRtpDecoderByCodecId,
|
||
getRtmpEncoderByTrack,
|
||
getRtmpDecoderByTrack,
|
||
getFrameFromPtr };
|
||
|
||
} // namespace mediakit
|