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rtc播放时，比对采样率等信息

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This commit is contained in:
xiongziliang 2021-04-04 23:58:59 +08:00
parent fe02f2cf1c
commit 6070654860
3 changed files with 54 additions and 9 deletions
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47
webrtc/Sdp.cpp
View File

@ -1331,6 +1331,10 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
if (!configure.enable) {
return;
}
bool failed = false;
RETRY:
for (auto &codec : configure.preferred_codec) {
for (auto &offer_media : medias) {
if (offer_media.type != type) {
@ -1342,12 +1346,15 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
}
const RtcCodecPlan *offer_plan_ptr = nullptr;
for (auto &plan : offer_media.plan) {
if (getCodecId(plan.codec) != codec) {
continue;
}
//命中偏好的编码格式
if (!onMatchCodecPlan(plan, codec)) {
continue;
if (!failed) {
//如果匹配失败了，那么随便选择一个plan
if (getCodecId(plan.codec) != codec) {
continue;
}
//命中偏好的编码格式
if (!onMatchCodecPlan(plan, codec)) {
continue;
}
}
//找到中意的codec
offer_plan_ptr = &plan;
@ -1476,4 +1483,32 @@ void RtcConfigure::matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const
return;
}
}
if (!failed) {
//只重试一次
failed = true;
goto RETRY;
}
}
void RtcConfigure::setPlayRtspInfo(const string &sdp){
RtcSession session;
session.loadFrom(sdp, false);
for (auto &m : session.media) {
switch (m.type) {
case TrackVideo : _rtsp_video_plan = std::make_shared<RtcCodecPlan>(m.plan[0]); break;
case TrackAudio : _rtsp_audio_plan = std::make_shared<RtcCodecPlan>(m.plan[0]); break;
default: break;
}
}
}
bool RtcConfigure::onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec){
if (_rtsp_audio_plan && codec == getCodecId(_rtsp_audio_plan->codec)) {
if (plan.sample_rate != _rtsp_audio_plan->sample_rate || plan.channel != _rtsp_audio_plan->channel) {
//音频采样率和通道数必须相同
return false;
}
}
return true;
}

9
webrtc/Sdp.h
View File

@ -574,6 +574,7 @@ public:
//rtc传输编码方案
class RtcCodecPlan{
public:
using Ptr = shared_ptr<RtcCodecPlan>;
uint8_t pt;
string codec;
uint32_t sample_rate;
@ -702,9 +703,15 @@ public:
shared_ptr<RtcSession> createAnswer(const RtcSession &offer);
void setPlayRtspInfo(const string &sdp);
private:
void matchMedia(shared_ptr<RtcSession> &ret, TrackType type, const vector<RtcMedia> &medias, const RtcTrackConfigure &configure);
bool onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec) { return true; }
bool onMatchCodecPlan(const RtcCodecPlan &plan, CodecId codec);
private:
RtcCodecPlan::Ptr _rtsp_video_plan;
RtcCodecPlan::Ptr _rtsp_audio_plan;
};

7
webrtc/WebRtcTransport.cpp
View File

@ -324,16 +324,19 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
//这是播放
configure.video.direction = RtpDirection::sendonly;
configure.audio.direction = RtpDirection::sendonly;
configure.setPlayRtspInfo(_src->getSdp());
_rtsp_send_sdp.loadFrom(_src->getSdp(), false);
//根据rtsp流的相关信息，设置rtc最佳编码
for (auto &m : _rtsp_send_sdp.media) {
switch (m.type) {
case TrackVideo: {
configure.video.preferred_codec.insert(configure.video.preferred_codec.begin(), getCodecId(m.plan[0].codec));
configure.video.preferred_codec.clear();
configure.video.preferred_codec.emplace_back(getCodecId(m.plan[0].codec));
break;
}
case TrackAudio: {
configure.audio.preferred_codec.insert(configure.audio.preferred_codec.begin(),getCodecId(m.plan[0].codec));
configure.audio.preferred_codec.clear();
configure.audio.preferred_codec.emplace_back(getCodecId(m.plan[0].codec));
break;
}
default: