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Unified Diff: talk/media/webrtc/webrtcvideoengine2.cc

Issue 1418123003: Adding reduced size RTCP configuration down to the video stream level. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing TODO comments. Created 5 years, 1 month ago
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Index: talk/media/webrtc/webrtcvideoengine2.cc
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 101ed15bddd996d4e51d462d537f4d50ee613dd4..62e3724eb88f7ca3fb568b8fb3986fb77781f704 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -863,20 +863,42 @@ bool WebRtcVideoChannel2::ReceiveCodecsHaveChanged(
return false;
}
-bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
+bool WebRtcVideoChannel2::SetSendParameters(
+ const VideoSendParameters& send_params) {
// TODO(pbos): Refactor this to only recreate the send streams once
// instead of 4 times.
- return (SetSendCodecs(params.codecs) &&
- SetSendRtpHeaderExtensions(params.extensions) &&
- SetMaxSendBandwidth(params.max_bandwidth_bps) &&
- SetOptions(params.options));
+ if (!SetSendCodecs(send_params.codecs) ||
+ !SetSendRtpHeaderExtensions(send_params.extensions) ||
+ !SetMaxSendBandwidth(send_params.max_bandwidth_bps) ||
+ !SetOptions(send_params.options)) {
+ return false;
+ }
+ if (send_params_.rtcp.reduced_size != send_params.rtcp.reduced_size) {
+ rtc::CritScope stream_lock(&stream_crit_);
+ for (auto& kv : send_streams_) {
+ kv.second->SetSendParameters(send_params);
+ }
+ }
+ send_params_ = send_params;
+ return true;
}
-bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) {
+bool WebRtcVideoChannel2::SetRecvParameters(
+ const VideoRecvParameters& recv_params) {
// TODO(pbos): Refactor this to only recreate the recv streams once
// instead of twice.
- return (SetRecvCodecs(params.codecs) &&
- SetRecvRtpHeaderExtensions(params.extensions));
+ if (!SetRecvCodecs(recv_params.codecs) ||
+ !SetRecvRtpHeaderExtensions(recv_params.extensions)) {
+ return false;
+ }
+ if (recv_params_.rtcp.reduced_size != recv_params.rtcp.reduced_size) {
+ rtc::CritScope stream_lock(&stream_crit_);
+ for (auto& kv : receive_streams_) {
+ kv.second->SetRecvParameters(recv_params);
+ }
+ }
+ recv_params_ = recv_params;
+ return true;
}
std::string WebRtcVideoChannel2::CodecSettingsVectorToString(
@@ -1094,15 +1116,10 @@ bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
webrtc::VideoSendStream::Config config(this);
config.overuse_callback = this;
- WebRtcVideoSendStream* stream =
- new WebRtcVideoSendStream(call_,
- sp,
- config,
- external_encoder_factory_,
- options_,
- bitrate_config_.max_bitrate_bps,
- send_codec_,
- send_rtp_extensions_);
+ WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
+ call_, sp, config, external_encoder_factory_, options_,
+ bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
+ send_params_);
uint32_t ssrc = sp.first_ssrc();
RTC_DCHECK(ssrc != 0);
@@ -1242,6 +1259,9 @@ void WebRtcVideoChannel2::ConfigureReceiverRtp(
config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
config->rtp.extensions = recv_rtp_extensions_;
+ config->rtp.rtcp_mode = recv_params_.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
// TODO(pbos): This protection is against setting the same local ssrc as
// remote which is not permitted by the lower-level API. RTCP requires a
@@ -1733,7 +1753,10 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
const VideoOptions& options,
int max_bitrate_bps,
const rtc::Optional<VideoCodecSettings>& codec_settings,
- const std::vector<webrtc::RtpExtension>& rtp_extensions)
+ const std::vector<webrtc::RtpExtension>& rtp_extensions,
+ // TODO(pbos): Don't duplicate information between send_params,
+ // rtp_extensions, options, etc.
+ const VideoSendParameters& send_params)
: ssrcs_(sp.ssrcs),
ssrc_groups_(sp.ssrc_groups),
call_(call),
@@ -1754,6 +1777,9 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
&parameters_.config.rtp.rtx.ssrcs);
parameters_.config.rtp.c_name = sp.cname;
parameters_.config.rtp.extensions = rtp_extensions;
+ parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
if (codec_settings) {
SetCodec(*codec_settings);
@@ -2070,6 +2096,18 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpExtensions(
}
}
+void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
+ const VideoSendParameters& send_params) {
+ rtc::CritScope cs(&lock_);
+ parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
+ if (stream_ != nullptr) {
+ LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
+ RecreateWebRtcStream();
+ }
+}
+
webrtc::VideoEncoderConfig
WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
const Dimensions& dimensions,
@@ -2497,6 +2535,15 @@ void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRtpExtensions(
RecreateWebRtcStream();
}
+void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvParameters(
+ const VideoRecvParameters& recv_params) {
+ config_.rtp.rtcp_mode = recv_params.rtcp.reduced_size
+ ? webrtc::RtcpMode::kReducedSize
+ : webrtc::RtcpMode::kCompound;
+ LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvParameters";
+ RecreateWebRtcStream();
+}
+
void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
if (stream_ != NULL) {
call_->DestroyVideoReceiveStream(stream_);

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