Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/engine/webrtcvideoengine2.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 800 matching lines @@
         RTC_DCHECK(kv.second != nullptr);
         kv.second->SetFeedbackParameters(HasNack(send_codec_->codec),
                                          HasRemb(send_codec_->codec),
                                          HasTransportCc(send_codec_->codec));
       }
     }
   }
   send_params_ = params;
   return true;
 }
+webrtc::RtpParameters WebRtcVideoChannel2::GetRtpParameters(
+    uint32_t ssrc) const {
+  rtc::CritScope stream_lock(&stream_crit_);
+  auto it = send_streams_.find(ssrc);
+  if (it == send_streams_.end()) {
+    LOG(LS_WARNING) << "Attempting to get RTP parameters for stream with ssrc "
+                    << ssrc << " which doesn't exist.";
+    return webrtc::RtpParameters();
+  }
+
+  return it->second->rtp_parameters();
+}
+
+bool WebRtcVideoChannel2::SetRtpParameters(
+    uint32_t ssrc,
+    const webrtc::RtpParameters& parameters) {
+  rtc::CritScope stream_lock(&stream_crit_);
+  auto it = send_streams_.find(ssrc);
+  if (it == send_streams_.end()) {
+    LOG(LS_ERROR) << "Attempting to set RTP parameters for stream with ssrc "
+                  << ssrc << " which doesn't exist.";
+    return false;
+  }
+
+  return it->second->SetRtpParameters(parameters);
+}
 
 bool WebRtcVideoChannel2::GetChangedRecvParameters(
     const VideoRecvParameters& params,
     ChangedRecvParameters* changed_params) const {
   if (!ValidateCodecFormats(params.codecs) ||
       !ValidateRtpExtensions(params.extensions)) {
     return false;
   }
 
   // Handle receive codecs.
@@ skipping 638 matching lines @@
     : worker_thread_(rtc::Thread::Current()),
       ssrcs_(sp.ssrcs),
       ssrc_groups_(sp.ssrc_groups),
       call_(call),
       cpu_restricted_counter_(0),
       number_of_cpu_adapt_changes_(0),
       capturer_(nullptr),
       external_encoder_factory_(external_encoder_factory),
       stream_(nullptr),
       parameters_(config, options, max_bitrate_bps, codec_settings),
+      rtp_parameters_(CreateRtpParametersWithOneEncoding()),
       pending_encoder_reconfiguration_(false),
       allocated_encoder_(nullptr, webrtc::kVideoCodecUnknown, false),
       sending_(false),
       muted_(false),
       first_frame_timestamp_ms_(0),
       last_frame_timestamp_ms_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
   parameters_.conference_mode = send_params.conference_mode;
 
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
@@ skipping 267 matching lines @@
     bool recreate_stream = false;
     if (params.rtcp_mode) {
       parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
       recreate_stream = true;
     }
     if (params.rtp_header_extensions) {
       parameters_.config.rtp.extensions = *params.rtp_header_extensions;
       recreate_stream = true;
     }
     if (params.max_bandwidth_bps) {
-      // Max bitrate has changed, reconfigure encoder settings on the next frame
-      // or stream recreation.
       parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
       pending_encoder_reconfiguration_ = true;
     }
     if (params.conference_mode) {
       parameters_.conference_mode = *params.conference_mode;
     }
 
     // Set codecs and options.
     if (params.codec) {
       SetCodec(*params.codec);
@@ skipping 13 matching lines @@
   // that might cause a lock order inversion.
   if (params.rtp_header_extensions) {
     sink_wants_.rotation_applied = !ContainsHeaderExtension(
         *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension);
     if (capturer_) {
       capturer_->AddOrUpdateSink(this, sink_wants_);
     }
   }
 }
 
+bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpParameters(
+    const webrtc::RtpParameters& new_parameters) {
+  if (!ValidateRtpParameters(new_parameters)) {
+    return false;
+  }
+
+  rtc::CritScope cs(&lock_);
+  if (new_parameters.encodings[0].max_bitrate_bps !=
+      rtp_parameters_.encodings[0].max_bitrate_bps) {
+    pending_encoder_reconfiguration_ = true;
+  }
+  rtp_parameters_ = new_parameters;
+  return true;
+}
+
+bool WebRtcVideoChannel2::WebRtcVideoSendStream::ValidateRtpParameters(
+    const webrtc::RtpParameters& rtp_parameters) {
+  if (rtp_parameters.encodings.size() != 1) {
+    LOG(LS_ERROR)
+        << "Attempted to set RtpParameters without exactly one encoding";
+    return false;
+  }
+  return true;
+}
+
 webrtc::VideoEncoderConfig
 WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
     const Dimensions& dimensions,
     const VideoCodec& codec) const {
   webrtc::VideoEncoderConfig encoder_config;
   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 =
@@ skipping 19 matching lines @@
   clamped_codec.height = height;
 
   // 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, only configure a single stream.
   size_t stream_count = parameters_.config.rtp.ssrcs.size();
   if (IsCodecBlacklistedForSimulcast(codec.name) || is_screencast) {
     stream_count = 1;
   }
 
-  encoder_config.streams =
-      CreateVideoStreams(clamped_codec, parameters_.options,
-                         parameters_.max_bitrate_bps, stream_count);
+  int stream_max_bitrate =
+      MinPositive(rtp_parameters_.encodings[0].max_bitrate_bps,
+                  parameters_.max_bitrate_bps);
+  encoder_config.streams = CreateVideoStreams(
+      clamped_codec, parameters_.options, stream_max_bitrate, stream_count);
 
   // Conference mode screencast uses 2 temporal layers split at 100kbit.
   if (parameters_.conference_mode && is_screencast &&
       encoder_config.streams.size() == 1) {
     ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
 
     // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
     // on the VideoCodec struct as target and max bitrates, respectively.
     // See eg. webrtc::VP8EncoderImpl::SetRates().
     encoder_config.streams[0].target_bitrate_bps =
@@ skipping 664 matching lines @@
         rtx_mapping[video_codecs[i].codec.id] !=
             fec_settings.red_payload_type) {
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket