webRTC sdp b=AS:600 x-google-min-bitrate 带宽码率那些事

发送端两个sdp片段：SetLocalDescription，SetremoteDescription;SetLocalDescription:
b=AS:600
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:gY3K
a=ice-pwd:1HSgJcvP5SkTGcJ8uiixrdqe
a=ice-options:trickle
a=fingerprint:sha-256 BC:7A:18:73:C5:BF:A3:9D:06:BC:13:A6:AC:99:0C:6B:BE:C8:6A:B5:DF:78:2D:C9:CE:3C:22:0E:2B:02:53:20
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:aG6LXeJdnm3w9XiAXNENCH2AsDWvZKmUrzot 3f2908cd-12b0-4158-a9fc-9cf7c285993fSetremoteDescription:
b=AS:600
a=rtpmap:125 H264/90000
a=rtpmap:107 rtx/90000
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;x-google-min-bitrate=400;x-google-max-bitrate=600;profile-level-id=42e01f
a=fmtp:107 apt=125
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=rtcp-fb:125 goog-remb
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=setup:active
a=mid:1
a=recvonly
a=ice-ufrag:4689494443441386ohoc
a=ice-pwd:orts7pigqcfgxy51906986jdz5wtizq2
a=ice-options:renomination
a=rtcp-mux
a=rtcp-rsize

webrtc::VideoEncoderConfig
WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(const VideoCodec& codec) const {RTC_DCHECK_RUN_ON(&thread_checker_);webrtc::VideoEncoderConfig encoder_config;encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name);encoder_config.video_format =webrtc::SdpVideoFormat(codec.name, codec.params);bool is_screencast = parameters_.options.is_screencast.value_or(false);if (is_screencast) {encoder_config.min_transmit_bitrate_bps =1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0);encoder_config.content_type =webrtc::VideoEncoderConfig::ContentType::kScreen;} else {encoder_config.min_transmit_bitrate_bps = 0;encoder_config.content_type =webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;}// By default, the stream count for the codec configuration should match the// number of negotiated ssrcs. But if the codec is blacklisted for simulcast// or a screencast (and not in simulcast screenshare experiment), only// configure a single stream.encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();if (IsCodecBlacklistedForSimulcast(codec.name)) {encoder_config.number_of_streams = 1;}// parameters_.max_bitrate comes from the max bitrate set at the SDP// (m-section) level with the attribute "b=AS." Note that we override this// value below if the RtpParameters max bitrate set with// RtpSender::SetParameters has a lower value.int stream_max_bitrate = parameters_.max_bitrate_bps;// When simulcast is enabled (when there are multiple encodings),// encodings[i].max_bitrate_bps will be enforced by// encoder_config.simulcast_layers[i].max_bitrate_bps. Otherwise, it's// enforced by stream_max_bitrate, taking the minimum of the two maximums// (one coming from SDP, the other coming from RtpParameters).if (rtp_parameters_.encodings[0].max_bitrate_bps &&rtp_parameters_.encodings.size() == 1) {stream_max_bitrate =MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps),parameters_.max_bitrate_bps);}// The codec max bitrate comes from the "x-google-max-bitrate" parameter// attribute set in the SDP for a specific codec. As done in// WebRtcVideoChannel::SetSendParameters, this value does not override the// stream max_bitrate set above.int codec_max_bitrate_kbps;if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps) &&stream_max_bitrate == -1) {stream_max_bitrate = codec_max_bitrate_kbps * 1000;}encoder_config.max_bitrate_bps = stream_max_bitrate;// The encoder config's default bitrate priority is set to 1.0,// unless it is set through the sender's encoding parameters.// The bitrate priority, which is used in the bitrate allocation, is done// on a per sender basis, so we use the first encoding's value.encoder_config.bitrate_priority =rtp_parameters_.encodings[0].bitrate_priority;// Application-controlled state is held in the encoder_config's// simulcast_layers. Currently this is used to control which simulcast layers// are active and for configuring the min/max bitrate and max framerate.// The encoder_config's simulcast_layers is also used for non-simulcast (when// there is a single layer).RTC_DCHECK_GE(rtp_parameters_.encodings.size(),encoder_config.number_of_streams);RTC_DCHECK_GT(encoder_config.number_of_streams, 0);// Copy all provided constraints.encoder_config.simulcast_layers.resize(rtp_parameters_.encodings.size());for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i) {encoder_config.simulcast_layers[i].active =rtp_parameters_.encodings[i].active;if (rtp_parameters_.encodings[i].min_bitrate_bps) {encoder_config.simulcast_layers[i].min_bitrate_bps =*rtp_parameters_.encodings[i].min_bitrate_bps;}if (rtp_parameters_.encodings[i].max_bitrate_bps) {encoder_config.simulcast_layers[i].max_bitrate_bps =*rtp_parameters_.encodings[i].max_bitrate_bps;}if (rtp_parameters_.encodings[i].max_framerate) {encoder_config.simulcast_layers[i].max_framerate =*rtp_parameters_.encodings[i].max_framerate;}if (rtp_parameters_.encodings[i].scale_resolution_down_by) {encoder_config.simulcast_layers[i].scale_resolution_down_by =*rtp_parameters_.encodings[i].scale_resolution_down_by;}if (rtp_parameters_.encodings[i].num_temporal_layers) {encoder_config.simulcast_layers[i].num_temporal_layers =*rtp_parameters_.encodings[i].num_temporal_layers;}}int max_qp = kDefaultQpMax;codec.GetParam(kCodecParamMaxQuantization, &max_qp);encoder_config.video_stream_factory =new rtc::RefCountedObject<EncoderStreamFactory>(codec.name, max_qp, is_screencast, parameters_.conference_mode);return encoder_config;
}

1.sdp 中有些关于带宽码率的设置，AS:行,x-google-max-bitrate,x-google-start-bitrate,x-google-min-bitrate等等，这些参数有什么作用呢？

AS:代表可用带宽，代表这路音频或者视频最多可以占用多大的带宽，比如b=AS:600就代表这路媒体流可使用多少带宽。

x-google-max-bitrate和AS作用差不多，但是有一定差异，看上面代码段，当同时设置AS行和x-google-max-bitrate时，x-google-max-bitrate会覆盖AS行，另外一点值得注意的是x-google-max-bitrate代表编码器可使用最大码率，而AS行不仅是编码器输出码率，还包括重传带宽，比如当上行网络不好时，带宽分配模块会将AS值分给编码码率以及重传带宽。

2.如果setLocalDescription和setRemoteDescription同时设置了AS值，发送端会以哪个为准？

这时候发送端会以较小的AS值为准。

webrtc::VideoEncoderConfig
WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(const VideoCodec& codec) const {RTC_DCHECK_RUN_ON(&thread_checker_);webrtc::VideoEncoderConfig encoder_config;encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name);encoder_config.video_format =webrtc::SdpVideoFormat(codec.name, codec.params);bool is_screencast = parameters_.options.is_screencast.value_or(false);if (is_screencast) {encoder_config.min_transmit_bitrate_bps =1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0);encoder_config.content_type =webrtc::VideoEncoderConfig::ContentType::kScreen;} else {encoder_config.min_transmit_bitrate_bps = 0;encoder_config.content_type =webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;}// By default, the stream count for the codec configuration should match the// number of negotiated ssrcs. But if the codec is disabled for simulcast// or a screencast (and not in simulcast screenshare experiment), only// configure a single stream.encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();if (IsCodecDisabledForSimulcast(codec.name, call_->trials())) {encoder_config.number_of_streams = 1;}// parameters_.max_bitrate comes from the max bitrate set at the SDP// (m-section) level with the attribute "b=AS." Note that we override this// value below if the RtpParameters max bitrate set with// RtpSender::SetParameters has a lower value.int stream_max_bitrate = parameters_.max_bitrate_bps;// When simulcast is enabled (when there are multiple encodings),// encodings[i].max_bitrate_bps will be enforced by// encoder_config.simulcast_layers[i].max_bitrate_bps. Otherwise, it's// enforced by stream_max_bitrate, taking the minimum of the two maximums// (one coming from SDP, the other coming from RtpParameters).if (rtp_parameters_.encodings[0].max_bitrate_bps &&rtp_parameters_.encodings.size() == 1) {stream_max_bitrate =MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps),parameters_.max_bitrate_bps);}// The codec max bitrate comes from the "x-google-max-bitrate" parameter// attribute set in the SDP for a specific codec. As done in// WebRtcVideoChannel::SetSendParameters, this value does not override the// stream max_bitrate set above.int codec_max_bitrate_kbps;if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps) &&stream_max_bitrate == -1) {stream_max_bitrate = codec_max_bitrate_kbps * 1000;}encoder_config.max_bitrate_bps = stream_max_bitrate;// The encoder config's default bitrate priority is set to 1.0,// unless it is set through the sender's encoding parameters.// The bitrate priority, which is used in the bitrate allocation, is done// on a per sender basis, so we use the first encoding's value.encoder_config.bitrate_priority =rtp_parameters_.encodings[0].bitrate_priority;// Application-controlled state is held in the encoder_config's// simulcast_layers. Currently this is used to control which simulcast layers// are active and for configuring the min/max bitrate and max framerate.// The encoder_config's simulcast_layers is also used for non-simulcast (when// there is a single layer).RTC_DCHECK_GE(rtp_parameters_.encodings.size(),encoder_config.number_of_streams);RTC_DCHECK_GT(encoder_config.number_of_streams, 0);// Copy all provided constraints.encoder_config.simulcast_layers.resize(rtp_parameters_.encodings.size());for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i) {encoder_config.simulcast_layers[i].active =rtp_parameters_.encodings[i].active;encoder_config.simulcast_layers[i].scalability_mode =rtp_parameters_.encodings[i].scalability_mode;if (rtp_parameters_.encodings[i].min_bitrate_bps) {encoder_config.simulcast_layers[i].min_bitrate_bps =*rtp_parameters_.encodings[i].min_bitrate_bps;}if (rtp_parameters_.encodings[i].max_bitrate_bps) {encoder_config.simulcast_layers[i].max_bitrate_bps =*rtp_parameters_.encodings[i].max_bitrate_bps;}if (rtp_parameters_.encodings[i].max_framerate) {encoder_config.simulcast_layers[i].max_framerate =*rtp_parameters_.encodings[i].max_framerate;}if (rtp_parameters_.encodings[i].scale_resolution_down_by) {encoder_config.simulcast_layers[i].scale_resolution_down_by =*rtp_parameters_.encodings[i].scale_resolution_down_by;}if (rtp_parameters_.encodings[i].num_temporal_layers) {encoder_config.simulcast_layers[i].num_temporal_layers =*rtp_parameters_.encodings[i].num_temporal_layers;}}encoder_config.legacy_conference_mode = parameters_.conference_mode;encoder_config.is_quality_scaling_allowed =!disable_automatic_resize_ && !is_screencast &&(parameters_.config.rtp.ssrcs.size() == 1 ||NumActiveStreams(rtp_parameters_) == 1);int max_qp = kDefaultQpMax;codec.GetParam(kCodecParamMaxQuantization, &max_qp);// wx add begin// encoder_config.min_transmit_bitrate_bps = 800*1000;// encoder_config.max_bitrate_bps = 1200*1000;RTC_LOG(LS_ERROR) << "wx_test:"<< "codec:" << codec.ToString().c_str() << "encoder_config"<< encoder_config.ToString();// wx add startencoder_config.video_stream_factory =rtc::make_ref_counted<EncoderStreamFactory>(codec.name, max_qp, is_screencast, parameters_.conference_mode);return encoder_config;
}

3.x-google-min-bitrate这个只有什么作用？

这个参数是控制最小可用带宽的，比如现在x-google-max-bitrate=600，x-google-min-bitrate=400;（使用GCC时）当上行网络不好出现丢包时，接收端会反馈给发送端丢包率，接收端会根据丢包率来调整发送端带宽，如果不对最小带宽限制，GCC出的码率会降的很低，这时候如果需要发送关键帧将会卡主很久，因此这里设置最小码率对GCC做些限制，这也是为什么使用Remb而不是Transport-cc的探测带宽主要原因，可能T-cc会探测的更准确些，但是服务器控制不了，还是使用了Remb。

struct BitrateConstraints {int min_bitrate_bps = 0;int start_bitrate_bps = kDefaultStartBitrateBps;int max_bitrate_bps = -1;private:static constexpr int kDefaultStartBitrateBps = 300000;
};webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec) {webrtc::BitrateConstraints config;int bitrate_kbps = 0;if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&bitrate_kbps > 0) {config.min_bitrate_bps = bitrate_kbps * 1000;} else {config.min_bitrate_bps = 0;}if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&bitrate_kbps > 0) {config.start_bitrate_bps = bitrate_kbps * 1000;} else {// Do not reconfigure start bitrate unless it's specified and positive.config.start_bitrate_bps = -1;}if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&bitrate_kbps > 0) {config.max_bitrate_bps = bitrate_kbps * 1000;} else {config.max_bitrate_bps = -1;}return config;
}

改图是发送端的videoBwe图，最开始设置总带宽为600，实际编码码率在600以下波动，当发送端设置丢包时，GCC估计出的带宽迅速下降到400，因为我设置了x-google-min-bitrate=400，不然根据GCC的带宽估计算法持续反馈丢包率的时候会降到十几K。这时总带宽会降到400之后不在下降，然后码率分配模块会将总带宽分配给编码码率和重传带宽，这块带宽怎么分配的后面再讲。

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webRTC sdp b=AS:600 x-google-min-bitrate 带宽码率那些事

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