WebRTC清晰度和流畅度

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,
};
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18

接口是

  // See https://crbug.com/653531 and https://w3c.github.io/mst-content-hint.
  enum class ContentHint { kNone, kFluid, kDetailed, kText };
1
2

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;
}
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38

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;
				}					
			}
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27

上述代码video_track创建好之后，调用

video_track->set_content_hint(webrtc::VideoTrackInterface::ContentHint::kDetailed);
1

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