WebRTC清晰度和流畅度

flyfish

WebRTC提供了4种模式DISABLED，MAINTAIN_FRAMERATE，MAINTAIN_RESOLUTION，BALANCED

// Based on the spec in
// https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
// These options are enforced on a best-effort basis. For instance, all of
// these options may suffer some frame drops in order to avoid queuing.
// TODO(sprang): Look into possibility of more strictly enforcing the
// maintain-framerate option.
// TODO(deadbeef): Default to "balanced", as the spec indicates?
enum class DegradationPreference {// Don't take any actions based on over-utilization signals. Not part of the// web API.DISABLED,// On over-use, request lower resolution, possibly causing down-scaling.MAINTAIN_FRAMERATE,// On over-use, request lower frame rate, possibly causing frame drops.MAINTAIN_RESOLUTION,// Try to strike a "pleasing" balance between frame rate or resolution.BALANCED,
};

接口是

  // See https://crbug.com/653531 and https://w3c.github.io/mst-content-hint.enum class ContentHint { kNone, kFluid, kDetailed, kText };

根据源码 接口这里不是一一对应的kDetailed和kText是类似的

webrtc::DegradationPreference
WebRtcVideoChannel::WebRtcVideoSendStream::GetDegradationPreference() const {// Do not adapt resolution for screen content as this will likely// result in blurry and unreadable text.// `this` acts like a VideoSource to make sure SinkWants are handled on the// correct thread.if (!enable_cpu_overuse_detection_) {return webrtc::DegradationPreference::DISABLED;}webrtc::DegradationPreference degradation_preference;if (rtp_parameters_.degradation_preference.has_value()) {degradation_preference = *rtp_parameters_.degradation_preference;} else {if (parameters_.options.content_hint ==webrtc::VideoTrackInterface::ContentHint::kFluid) {degradation_preference =webrtc::DegradationPreference::MAINTAIN_FRAMERATE;} else if (parameters_.options.is_screencast.value_or(false) ||parameters_.options.content_hint ==webrtc::VideoTrackInterface::ContentHint::kDetailed ||parameters_.options.content_hint ==webrtc::VideoTrackInterface::ContentHint::kText) {degradation_preference =webrtc::DegradationPreference::MAINTAIN_RESOLUTION;} else if (IsEnabled(call_->trials(), "WebRTC-Video-BalancedDegradation")) {// Standard wants balanced by default, but it needs to be tuned first.degradation_preference = webrtc::DegradationPreference::BALANCED;} else {// Keep MAINTAIN_FRAMERATE by default until BALANCED has been tuned for// all codecs and launched.degradation_preference =webrtc::DegradationPreference::MAINTAIN_FRAMERATE;}}return degradation_preference;
}

使用方法

// create a new webrtc stream
{std::lock_guard<std::mutex> mlock(m_streamMapMutex);std::map<std::string, std::pair<rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>, rtc::scoped_refptr<webrtc::AudioSourceInterface>>>::iterator it = m_stream_map.find(streamLabel);if (it != m_stream_map.end()){std::pair<rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>, rtc::scoped_refptr<webrtc::AudioSourceInterface>> pair = it->second;rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> videoSource(pair.first);if (!videoSource){RTC_LOG(LS_ERROR) << "Cannot create capturer video:" << videourl;}else{rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track = m_peer_connection_factory->CreateVideoTrack(streamLabel + "_video", videoSource.get());if ((video_track) && (!peer_connection->AddTrack(video_track, {streamLabel}).ok())){RTC_LOG(LS_ERROR) << "Adding VideoTrack to MediaStream failed";}else{RTC_LOG(LS_INFO) << "VideoTrack added to PeerConnection";ret = true;}					}

上述代码video_track创建好之后，调用

video_track->set_content_hint(webrtc::VideoTrackInterface::ContentHint::kDetailed);

参考
https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference
https://crbug.com/653531 and https://w3c.github.io/mst-content-hint