Skip to content

Z
ZLMediaKit
Commit bacf8f10 by xiongziliang
Options

G711支持多种规格

parent 5c3418a4
正在显示 包含 204 行增加 和 197 行删除
src/Common/Device.cpp
......@@ -196,7 +196,7 @@ void DevChannel::initAudio(const AudioInfo& info) {
}
case CodecG711A :
case CodecG711U : addTrack(std::make_shared<G711Track>(info.codecId)); break;
case CodecG711U : addTrack(std::make_shared<G711Track>(info.codecId, info.iSampleRate, info.iChannel, info.iSampleBit)); break;
default: WarnL << "不支持该类型的音频编码类型:" << info.codecId; break;
}
}
......
src/Common/MediaSink.cpp
......@@ -42,8 +42,8 @@ void MediaSink::addTrack(const Track::Ptr &track_in) {
if (_allTrackReady) {
//运行至这里说明Track状态由未就绪切换为已就绪状态,那么这帧就不应该丢弃
onTrackFrame(frame);
} else {
ErrorL << "some track is unready，drop frame of: " << frame->getCodecName();
} else if(frame->keyFrame()){
WarnL << "some track is unready，drop key frame of: " << frame->getCodecName();
}
}));
}
......
src/Extension/AACRtmp.cpp
......@@ -104,7 +104,7 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
//header
uint8_t is_config = false;
rtmpPkt->strBuf.push_back(_ui8AudioFlags);
rtmpPkt->strBuf.push_back(_audio_flv_flags);
rtmpPkt->strBuf.push_back(!is_config);
//aac data
......@@ -117,44 +117,16 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
rtmpPkt->typeId = MSG_AUDIO;
RtmpCodec::inputRtmp(rtmpPkt, false);
}
}
void AACRtmpEncoder::makeAudioConfigPkt() {
makeAdtsHeader(_aac_cfg,*_adts);
int iSampleRate , iChannel , iSampleBit = 16;
getAACInfo(*_adts,iSampleRate,iChannel);
uint8_t flvStereoOrMono = (iChannel > 1);
uint8_t flvSampleRate;
switch (iSampleRate) {
case 48000:
case 44100:
flvSampleRate = 3;
break;
case 24000:
case 22050:
flvSampleRate = 2;
break;
case 12000:
case 11025:
flvSampleRate = 1;
break;
default:
flvSampleRate = 0;
break;
}
uint8_t flvSampleBit = iSampleBit == 16;
uint8_t flvAudioType = FLV_CODEC_AAC;
_ui8AudioFlags = (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
_audio_flv_flags = getAudioRtmpFlags(std::make_shared<AACTrack>(_aac_cfg));
RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
rtmpPkt->strBuf.clear();
//header
uint8_t is_config = true;
rtmpPkt->strBuf.push_back(_ui8AudioFlags);
rtmpPkt->strBuf.push_back(_audio_flv_flags);
rtmpPkt->strBuf.push_back(!is_config);
//aac config
rtmpPkt->strBuf.append(_aac_cfg);
......
src/Extension/AACRtmp.h
......@@ -79,7 +79,7 @@ public:
private:
void makeAudioConfigPkt();
private:
uint8_t _ui8AudioFlags;
uint8_t _audio_flv_flags;
AACTrack::Ptr _track;
};
......
src/Extension/Factory.cpp
......@@ -48,11 +48,11 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
}
if (strcasecmp(track->_codec.data(), "PCMA") == 0) {
return std::make_shared<G711Track>(CodecG711A);
return std::make_shared<G711Track>(CodecG711A, track->_samplerate, track->_channel, 16);
}
if (strcasecmp(track->_codec.data(), "PCMU") == 0) {
return std::make_shared<G711Track>(CodecG711U);
return std::make_shared<G711Track>(CodecG711U, track->_samplerate, track->_channel, 16);
}
if (strcasecmp(track->_codec.data(), "h264") == 0) {
......@@ -88,29 +88,6 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
return nullptr;
}
Track::Ptr Factory::getTrackByCodecId(CodecId codecId) {
switch (codecId){
case CodecH264:{
return std::make_shared<H264Track>();
}
case CodecH265:{
return std::make_shared<H265Track>();
}
case CodecAAC:{
return std::make_shared<AACTrack>();
}
case CodecG711A: {
return std::make_shared<G711Track>(CodecG711A);
}
case CodecG711U: {
return std::make_shared<G711Track>(CodecG711U);
}
default:
WarnL << "暂不支持该CodecId:" << codecId;
return nullptr;
}
}
RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
GET_CONFIG(uint32_t,audio_mtu,Rtp::kAudioMtuSize);
GET_CONFIG(uint32_t,video_mtu,Rtp::kVideoMtuSize);
......@@ -134,35 +111,23 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
auto interleaved = sdp->getTrackType() * 2;
auto codec_id = sdp->getCodecId();
switch (codec_id){
case CodecH264:
return std::make_shared<H264RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecH265:
return std::make_shared<H265RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecAAC:
return std::make_shared<AACRtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecG711A:
case CodecG711U:
return std::make_shared<G711RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
default:
WarnL << "暂不支持该CodecId:" << codec_id;
return nullptr;
case CodecH264 : return std::make_shared<H264RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecH265 : return std::make_shared<H265RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecAAC : return std::make_shared<AACRtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecG711A :
case CodecG711U : return std::make_shared<G711RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
default : WarnL << "暂不支持该CodecId:" << codec_id; return nullptr;
}
}
RtpCodec::Ptr Factory::getRtpDecoderByTrack(const Track::Ptr &track) {
switch (track->getCodecId()){
case CodecH264:
return std::make_shared<H264RtpDecoder>();
case CodecH265:
return std::make_shared<H265RtpDecoder>();
case CodecAAC:
return std::make_shared<AACRtpDecoder>(track->clone());
case CodecG711A:
case CodecG711U:
return std::make_shared<G711RtpDecoder>(track->clone());
default:
WarnL << "暂不支持该CodecId:" << track->getCodecName();
return nullptr;
case CodecH264 : return std::make_shared<H264RtpDecoder>();
case CodecH265 : return std::make_shared<H265RtpDecoder>();
case CodecAAC : return std::make_shared<AACRtpDecoder>(track->clone());
case CodecG711A :
case CodecG711U : return std::make_shared<G711RtpDecoder>(track->clone());
default : WarnL << "暂不支持该CodecId:" << track->getCodecName(); return nullptr;
}
}
......@@ -190,15 +155,25 @@ static CodecId getVideoCodecIdByAmf(const AMFValue &val){
case FLV_CODEC_H264: return CodecH264;
case FLV_CODEC_AAC: return CodecAAC;
case FLV_CODEC_H265: return CodecH265;
default:
WarnL << "暂不支持该Amf:" << type_id;
return CodecInvalid;
default : WarnL << "暂不支持该Amf:" << type_id; return CodecInvalid;
}
}
return CodecInvalid;
}
Track::Ptr getTrackByCodecId(CodecId codecId, int sample_rate = 0, int channels = 0, int sample_bit = 0) {
switch (codecId){
case CodecH264 : return std::make_shared<H264Track>();
case CodecH265 : return std::make_shared<H265Track>();
case CodecAAC : return std::make_shared<AACTrack>();
case CodecG711A :
case CodecG711U : return (sample_rate && channels && sample_bit) ? std::make_shared<G711Track>(codecId, sample_rate, channels, sample_bit) : nullptr;
default : WarnL << "暂不支持该CodecId:" << codecId; return nullptr;
}
}
Track::Ptr Factory::getVideoTrackByAmf(const AMFValue &amf) {
CodecId codecId = getVideoCodecIdByAmf(amf);
if(codecId == CodecInvalid){
......@@ -230,28 +205,22 @@ static CodecId getAudioCodecIdByAmf(const AMFValue &val) {
return CodecInvalid;
}
Track::Ptr Factory::getAudioTrackByAmf(const AMFValue& amf){
Track::Ptr Factory::getAudioTrackByAmf(const AMFValue& amf, int sample_rate, int channels, int sample_bit){
CodecId codecId = getAudioCodecIdByAmf(amf);
if (codecId == CodecInvalid) {
return nullptr;
}
return getTrackByCodecId(codecId);
return getTrackByCodecId(codecId, sample_rate, channels, sample_bit);
}
RtmpCodec::Ptr Factory::getRtmpCodecByTrack(const Track::Ptr &track) {
switch (track->getCodecId()){
case CodecH264:
return std::make_shared<H264RtmpEncoder>(track);
case CodecAAC:
return std::make_shared<AACRtmpEncoder>(track);
case CodecH265:
return std::make_shared<H265RtmpEncoder>(track);
case CodecG711A:
case CodecG711U:
return std::make_shared<G711RtmpEncoder>(track);
default:
WarnL << "暂不支持该CodecId:" << track->getCodecName();
return nullptr;
case CodecH264 : return std::make_shared<H264RtmpEncoder>(track);
case CodecAAC : return std::make_shared<AACRtmpEncoder>(track);
case CodecH265 : return std::make_shared<H265RtmpEncoder>(track);
case CodecG711A :
case CodecG711U : return std::make_shared<G711RtmpEncoder>(track);
default : WarnL << "暂不支持该CodecId:" << track->getCodecName(); return nullptr;
}
}
......
src/Extension/Factory.h
......@@ -24,13 +24,6 @@ namespace mediakit{
class Factory {
public:
/**
* 根据CodecId获取Track，该Track的ready()状态一般都为false
* @param codecId 编解码器id
*/
static Track::Ptr getTrackByCodecId(CodecId codecId);
////////////////////////////////rtsp相关//////////////////////////////////
/**
* 根据sdp生成Track对象
......@@ -61,7 +54,7 @@ public:
* 根据amf对象获取音频相应的Track
* @param amf rtmp metadata中的audiocodecid的值
*/
static Track::Ptr getAudioTrackByAmf(const AMFValue& amf);
static Track::Ptr getAudioTrackByAmf(const AMFValue& amf, int sample_rate, int channels, int sample_bit);
/**
* 根据Track获取Rtmp的编解码器
......
src/Extension/G711.cpp
......@@ -12,13 +12,12 @@
namespace mediakit{
Sdp::Ptr G711Track::getSdp() {
if(!ready()){
WarnL << getCodecName() << " Track未准备好";
return nullptr;
}
return std::make_shared<G711Sdp>(getCodecId());
return std::make_shared<G711Sdp>(getCodecId(), getAudioSampleRate(), getAudioChannel());
}
}//namespace mediakit
......
src/Extension/G711.h
......@@ -102,8 +102,11 @@ public:
/**
* G711A G711U
*/
G711Track(CodecId codecId){
G711Track(CodecId codecId,int sample_rate, int channels, int sample_bit){
_codecid = codecId;
_sample_rate = sample_rate;
_channels = channels;
_sample_bit = sample_bit;
}
/**
......@@ -114,7 +117,7 @@ public:
}
/**
* G711的Track不需要初始化
* 是否已经初始化
*/
bool ready() override {
return true;
......@@ -124,21 +127,21 @@ public:
* 返回音频采样率
*/
int getAudioSampleRate() const override{
return 8000;
return _sample_rate;
}
/**
* 返回音频采样位数，一般为16或8
*/
int getAudioSampleBit() const override{
return 16;
return _sample_bit;
}
/**
* 返回音频通道数
*/
int getAudioChannel() const override{
return 1;
return _channels;
}
private:
......@@ -149,7 +152,10 @@ private:
//生成sdp
Sdp::Ptr getSdp() override ;
private:
CodecId _codecid = CodecG711A;
CodecId _codecid;
int _sample_rate;
int _channels;
int _sample_bit;
};
/**
......@@ -160,11 +166,17 @@ public:
/**
* G711采样率固定为8000
* @param codecId G711A G711U
* @param sample_rate 音频采样率
* @param playload_type rtp playload
* @param bitrate 比特率
*/
G711Sdp(CodecId codecId) : Sdp(8000,payloadType(codecId)), _codecId(codecId){
int pt = payloadType(codecId);
_printer << "m=audio 0 RTP/AVP " << pt << "\r\n";
_printer << "a=rtpmap:" << pt << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << 8000 << "\r\n";
G711Sdp(CodecId codecId,
int sample_rate,
int channels,
int playload_type = 98,
int bitrate = 128) : Sdp(sample_rate,playload_type), _codecId(codecId){
_printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
_printer << "a=rtpmap:" << playload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate << "/" << channels << "\r\n";
_printer << "a=control:trackID=" << getTrackType() << "\r\n";
}
......@@ -180,14 +192,6 @@ public:
return _codecId;
}
int payloadType(CodecId codecId){
switch (codecId){
case CodecG711A : return 8;
case CodecG711U : return 0;
default : return -1;
}
}
private:
_StrPrinter _printer;
CodecId _codecId;
......
src/Extension/G711Rtmp.cpp
......@@ -27,7 +27,7 @@ G711Frame::Ptr G711RtmpDecoder::obtainFrame() {
bool G711RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool) {
//拷贝G711负载
_frame->buffer.assign(pkt->strBuf.data() + 2, pkt->strBuf.size() - 2);
_frame->buffer.assign(pkt->strBuf.data() + 1, pkt->strBuf.size() - 1);
_frame->timeStamp = pkt->timeStamp;
//写入环形缓存
RtmpCodec::inputFrame(_frame);
......@@ -38,55 +38,17 @@ bool G711RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool) {
/////////////////////////////////////////////////////////////////////////////////////
G711RtmpEncoder::G711RtmpEncoder(const Track::Ptr &track) : G711RtmpDecoder(track->getCodecId()) {
auto g711_track = dynamic_pointer_cast<AudioTrack>(track);
if(!g711_track){
WarnL << "无效的G711 track, 将忽略打包为RTMP";
return;
}
auto iSampleRate = g711_track->getAudioSampleRate() ;
auto iChannel = g711_track->getAudioChannel();
auto iSampleBit = g711_track->getAudioSampleBit();
uint8_t flvStereoOrMono = (iChannel > 1);
uint8_t flvSampleRate;
switch (iSampleRate) {
case 48000:
case 44100:
flvSampleRate = 3;
break;
case 24000:
case 22050:
flvSampleRate = 2;
break;
case 12000:
case 11025:
flvSampleRate = 1;
break;
default:
flvSampleRate = 0;
break;
}
uint8_t flvSampleBit = iSampleBit == 16;
uint8_t flvAudioType ;
switch (g711_track->getCodecId()){
case CodecG711A : flvAudioType = FLV_CODEC_G711A; break;
case CodecG711U : flvAudioType = FLV_CODEC_G711U; break;
default: WarnL << "无效的G711 track, 将忽略打包为RTMP"; return ;
}
_g711_flags = (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
_audio_flv_flags = getAudioRtmpFlags(track);
}
void G711RtmpEncoder::inputFrame(const Frame::Ptr &frame) {
if(!_g711_flags){
if(!_audio_flv_flags){
return;
}
RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
rtmpPkt->strBuf.clear();
//header
uint8_t is_config = false;
rtmpPkt->strBuf.push_back(_g711_flags);
rtmpPkt->strBuf.push_back(!is_config);
rtmpPkt->strBuf.push_back(_audio_flv_flags);
//g711 data
rtmpPkt->strBuf.append(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
......
src/Extension/G711Rtmp.h
......@@ -62,7 +62,7 @@ public:
*/
void inputFrame(const Frame::Ptr &frame) override;
private:
uint8_t _g711_flags = 0;
uint8_t _audio_flv_flags = 0;
};
}//namespace mediakit
......
src/Rtmp/Rtmp.cpp
......@@ -36,11 +36,68 @@ AudioMeta::AudioMeta(const AudioTrack::Ptr &audio,int datarate){
_metadata.set("audiosamplesize", audio->getAudioSampleBit());
}
if(audio->getAudioChannel() > 0){
_metadata.set("audiochannels", audio->getAudioChannel());
_metadata.set("stereo", audio->getAudioChannel() > 1);
}
_codecId = audio->getCodecId();
_metadata.set("audiocodecid", Factory::getAmfByCodecId(_codecId));
}
uint8_t getAudioRtmpFlags(const Track::Ptr &track){
switch (track->getTrackType()){
case TrackAudio : {
auto audioTrack = dynamic_pointer_cast<AudioTrack>(track);
if (!audioTrack) {
WarnL << "获取AudioTrack失败";
return 0;
}
auto iSampleRate = audioTrack->getAudioSampleRate();
auto iChannel = audioTrack->getAudioChannel();
auto iSampleBit = audioTrack->getAudioSampleBit();
uint8_t flvAudioType ;
switch (track->getCodecId()){
case CodecG711A : flvAudioType = FLV_CODEC_G711A; break;
case CodecG711U : flvAudioType = FLV_CODEC_G711U; break;
case CodecAAC : {
flvAudioType = FLV_CODEC_AAC;
//aac不通过flags获取音频相关信息
iSampleRate = 44100;
iSampleBit = 16;
iChannel = 2;
break;
}
default: WarnL << "该编码格式不支持转换为RTMP: " << track->getCodecName(); return 0;
}
uint8_t flvSampleRate;
switch (iSampleRate) {
case 44100:
flvSampleRate = 3;
break;
case 22050:
flvSampleRate = 2;
break;
case 11025:
flvSampleRate = 1;
break;
case 16000: // nellymoser only
case 8000: // nellymoser only
case 5512: // not MP3
flvSampleRate = 0;
break;
default:
WarnL << "FLV does not support sample rate " << iSampleRate << " ,choose from (44100, 22050, 11025)";
return 0;
}
uint8_t flvStereoOrMono = (iChannel > 1);
uint8_t flvSampleBit = iSampleBit == 16;
return (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
}
default : return 0;
}
}
}//namespace mediakit
\ No newline at end of file
src/Rtmp/Rtmp.h
......@@ -161,27 +161,26 @@ public:
strBuf = std::move(that.strBuf);
}
bool isVideoKeyFrame() const {
return typeId == MSG_VIDEO && (uint8_t) strBuf[0] >> 4 == FLV_KEY_FRAME
&& (uint8_t) strBuf[1] == 1;
return typeId == MSG_VIDEO && (uint8_t) strBuf[0] >> 4 == FLV_KEY_FRAME && (uint8_t) strBuf[1] == 1;
}
bool isCfgFrame() const {
return (typeId == MSG_VIDEO || typeId == MSG_AUDIO)
&& (uint8_t) strBuf[1] == 0;
switch (typeId){
case MSG_VIDEO : return strBuf[1] == 0;
case MSG_AUDIO : {
switch (getMediaType()){
case FLV_CODEC_AAC : return strBuf[1] == 0;
default : return false;
}
int getMediaType() const {
switch (typeId) {
case MSG_VIDEO: {
return (uint8_t) strBuf[0] & 0x0F;
}
break;
case MSG_AUDIO: {
return (uint8_t) strBuf[0] >> 4;
default : return false;
}
break;
default:
break;
}
return 0;
int getMediaType() const {
switch (typeId) {
case MSG_VIDEO : return (uint8_t) strBuf[0] & 0x0F;
case MSG_AUDIO : return (uint8_t) strBuf[0] >> 4;
default : return 0;
}
}
int getAudioSampleRate() const {
if (typeId != MSG_AUDIO) {
......@@ -314,6 +313,8 @@ private:
CodecId _codecId;
};
//根据音频track获取flags
uint8_t getAudioRtmpFlags(const Track::Ptr &track);
}//namespace mediakit
......
src/Rtmp/RtmpDemuxer.cpp
......@@ -15,14 +15,55 @@ namespace mediakit {
void RtmpDemuxer::loadMetaData(const AMFValue &val){
try {
makeVideoTrack(val["videocodecid"]);
makeAudioTrack(val["audiocodecid"]);
int audiosamplerate = 0;
int audiochannels = 0;
int audiosamplesize = 0;
const AMFValue *audiocodecid = nullptr;
const AMFValue *videocodecid = nullptr;
val.object_for_each([&](const string &key, const AMFValue &val) {
if (key == "duration") {
_fDuration = val.as_number();
return;
}
if(key == "audiosamplerate"){
audiosamplerate = val.as_integer();
return;
}
if(key == "audiosamplesize"){
audiosamplesize = val.as_integer();
return;
}
if(key == "stereo"){
audiochannels = val.as_boolean() ? 2 : 1;
return;
}
if(key == "videocodecid"){
//找到视频
videocodecid = &val;
return;
}
if(key == "audiocodecid"){
//找到音频
audiocodecid = &val;
return;
}
});
if(videocodecid){
//有视频
makeVideoTrack(*videocodecid);
}
if(audiocodecid){
//有音频
makeAudioTrack(*audiocodecid, audiosamplerate, audiochannels, audiosamplesize);
}
}catch (std::exception &ex){
WarnL << ex.what();
}
......@@ -46,7 +87,7 @@ bool RtmpDemuxer::inputRtmp(const RtmpPacket::Ptr &pkt) {
if(!_tryedGetAudioTrack) {
_tryedGetAudioTrack = true;
auto codec = AMFValue(pkt->getMediaType());
makeAudioTrack(codec);
makeAudioTrack(codec, pkt->getAudioSampleRate(), pkt->getAudioChannel(), pkt->getAudioSampleBit());
}
if(_audioRtmpDecoder){
_audioRtmpDecoder->inputRtmp(pkt, false);
......@@ -69,6 +110,7 @@ void RtmpDemuxer::makeVideoTrack(const AMFValue &videoCodec) {
//设置rtmp解码器代理，生成的frame写入该Track
_videoRtmpDecoder->addDelegate(_videoTrack);
onAddTrack(_videoTrack);
_tryedGetVideoTrack = true;
} else {
//找不到相应的rtmp解码器，该track无效
_videoTrack.reset();
......@@ -76,9 +118,9 @@ void RtmpDemuxer::makeVideoTrack(const AMFValue &videoCodec) {
}
}
void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec) {
void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec,int sample_rate, int channels, int sample_bit) {
//生成Track对象
_audioTrack = dynamic_pointer_cast<AudioTrack>(Factory::getAudioTrackByAmf(audioCodec));
_audioTrack = dynamic_pointer_cast<AudioTrack>(Factory::getAudioTrackByAmf(audioCodec, sample_rate, channels, sample_bit));
if (_audioTrack) {
//生成rtmpCodec对象以便解码rtmp
_audioRtmpDecoder = Factory::getRtmpCodecByTrack(_audioTrack);
......@@ -86,6 +128,7 @@ void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec) {
//设置rtmp解码器代理，生成的frame写入该Track
_audioRtmpDecoder->addDelegate(_audioTrack);
onAddTrack(_audioTrack);
_tryedGetAudioTrack = true;
} else {
//找不到相应的rtmp解码器，该track无效
_audioTrack.reset();
......
src/Rtmp/RtmpDemuxer.h
......@@ -40,7 +40,7 @@ public:
bool inputRtmp(const RtmpPacket::Ptr &pkt);
private:
void makeVideoTrack(const AMFValue &val);
void makeAudioTrack(const AMFValue &val);
void makeAudioTrack(const AMFValue &val, int sample_rate, int channels, int sample_bit);
private:
bool _tryedGetVideoTrack = false;
bool _tryedGetAudioTrack = false;
......
src/Rtmp/RtmpSession.cpp
......@@ -193,7 +193,6 @@ void RtmpSession::onCmd_deleteStream(AMFDecoder &dec) {
throw std::runtime_error(StrPrinter << "Stop publishing" << endl);
}
void RtmpSession::sendPlayResponse(const string &err,const RtmpMediaSource::Ptr &src){
bool authSuccess = err.empty();
bool ok = (src.operator bool() && authSuccess);
......
src/Rtsp/Rtsp.cpp
......@@ -172,6 +172,7 @@ void SdpParser::load(const string &sdp) {
if (4 == sscanf(opt_val.data(), " %15[^ ] %d %15[^ ] %d", type, &port, rtp, &pt)) {
track->_pt = pt;
track->_samplerate = RtpPayload::getClockRate(pt) ;
track->_channel = RtpPayload::getAudioChannel(pt);
track->_type = toTrackType(type);
track->_m = opt_val;
track->_port = port;
......@@ -214,9 +215,14 @@ void SdpParser::load(const string &sdp) {
it = track._attr.find("rtpmap");
if(it != track._attr.end()){
auto rtpmap = it->second;
int pt, samplerate;
int pt, samplerate, channel;
char codec[16] = {0};
if (3 == sscanf(rtpmap.data(), "%d %15[^/]/%d", &pt, codec, &samplerate)) {
if (4 == sscanf(rtpmap.data(), "%d %15[^/]/%d/%d", &pt, codec, &samplerate, &channel)) {
track._pt = pt;
track._codec = codec;
track._samplerate = samplerate;
track._channel = channel;
}else if (3 == sscanf(rtpmap.data(), "%d %15[^/]/%d", &pt, codec, &samplerate)) {
track._pt = pt;
track._codec = codec;
track._samplerate = samplerate;
......
src/Rtsp/Rtsp.h
......@@ -128,6 +128,7 @@ public:
int _pt;
string _codec;
int _samplerate;
int _channel;
string _fmtp;
string _control;
string _control_surffix;
......
Markdown 格式
0% 或
您添加了 0 人 到此讨论。请谨慎行事。
请先完成此评论的编辑！
请 注册 或者 后发表评论