Apply VideoOptions per stream.

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 598 matching lines @@
     WebRtcVideoEncoderFactory* external_encoder_factory,
     WebRtcVideoDecoderFactory* external_decoder_factory)
     : VideoMediaChannel(config),
       call_(call),
       unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
       signal_cpu_adaptation_(config.enable_cpu_overuse_detection),
       disable_prerenderer_smoothing_(config.disable_prerenderer_smoothing),
       external_encoder_factory_(external_encoder_factory),
       external_decoder_factory_(external_decoder_factory) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  options_.SetAll(options);
+
+  send_params_.options = options;
+
   rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
   sending_ = false;
   default_send_ssrc_ = 0;
   RTC_DCHECK(ValidateCodecFormats(recv_codecs));
   recv_codecs_ = FilterSupportedCodecs(MapCodecs(recv_codecs));
 }
 
 WebRtcVideoChannel2::~WebRtcVideoChannel2() {
   for (auto& kv : send_streams_)
     delete kv.second;
@@ skipping 99 matching lines @@
   if (params.max_bandwidth_bps != bitrate_config_.max_bitrate_bps &&
       params.max_bandwidth_bps >= 0) {
     // 0 uncaps max bitrate (-1).
     changed_params->max_bandwidth_bps = rtc::Optional<int>(
         params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps);
   }
 
   // Handle options.
   // TODO(pbos): Require VideoSendParameters to contain a full set of options
   // and check if params.options != options_ instead of applying a delta.
-  VideoOptions new_options = options_;
+  VideoOptions new_options = send_params_.options;
   new_options.SetAll(params.options);
-  if (!(new_options == options_)) {
+  if (!(new_options == send_params_.options)) {
     changed_params->options = rtc::Optional<VideoOptions>(new_options);
   }
 
   // Handle RTCP mode.
   if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) {
     changed_params->rtcp_mode = rtc::Optional<webrtc::RtcpMode>(
         params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
                                  : webrtc::RtcpMode::kCompound);
   }
 
@@ skipping 42 matching lines @@
       bitrate_config_.min_bitrate_bps = max_bitrate_bps;
     }
     bitrate_config_changed = true;
   }
 
   if (bitrate_config_changed) {
     call_->SetBitrateConfig(bitrate_config_);
   }
 
   if (changed_params.options)
-    options_.SetAll(*changed_params.options);
+    send_params_.options.SetAll(*changed_params.options);
 
   {
     rtc::CritScope stream_lock(&stream_crit_);
     for (auto& kv : send_streams_) {
       kv.second->SetSendParameters(changed_params);
     }
     if (changed_params.codec) {
       // Update receive feedback parameters from new codec.
       LOG(LS_INFO)
           << "SetFeedbackOptions on all the receive streams because the send "
@@ skipping 128 matching lines @@
   LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable
                << "options: " << (options ? options->ToString() : "nullptr")
                << ").";
 
   // TODO(solenberg): The state change should be fully rolled back if any one of
   //                  these calls fail.
   if (!MuteStream(ssrc, !enable)) {
     return false;
   }
   if (enable && options) {
-    VideoSendParameters new_params = send_params_;
-    new_params.options.SetAll(*options);
-    SetSendParameters(send_params_);
+    SetOptions(ssrc, *options);
   }
   return true;
 }
 
 bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
     const StreamParams& sp) const {
   for (uint32_t ssrc : sp.ssrcs) {
     if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
       LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists.";
       return false;
@@ skipping 23 matching lines @@
 
   if (!ValidateSendSsrcAvailability(sp))
     return false;
 
   for (uint32_t used_ssrc : sp.ssrcs)
     send_ssrcs_.insert(used_ssrc);
 
   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_,
-      send_params_);
+  WebRtcVideoSendStream* stream =
+      new WebRtcVideoSendStream(call_, sp, config, external_encoder_factory_,
+                                bitrate_config_.max_bitrate_bps, send_codec_,
+                                send_rtp_extensions_, send_params_);
 
   uint32_t ssrc = sp.first_ssrc();
   RTC_DCHECK(ssrc != 0);
   send_streams_[ssrc] = stream;
 
   if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
     rtcp_receiver_report_ssrc_ = ssrc;
     LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added "
                     "a send stream.";
     for (auto& kv : receive_streams_)
@@ skipping 248 matching lines @@
   }
   video_media_info->bw_estimations.push_back(bwe_info);
 }
 
 bool WebRtcVideoChannel2::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
   LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
                << (capturer != NULL ? "(capturer)" : "NULL");
   RTC_DCHECK(ssrc != 0);
   {
     rtc::CritScope stream_lock(&stream_crit_);
-    if (send_streams_.find(ssrc) == send_streams_.end()) {
+    auto kv = send_streams_.find(ssrc);

[pbos-webrtc, 2016/02/17 12:09:47]

auto& kv

+    if (kv == send_streams_.end()) {
       LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
       return false;
     }
-    if (!send_streams_[ssrc]->SetCapturer(capturer)) {
+    if (!kv->second->SetCapturer(capturer)) {
       return false;
     }
   }
   {
     rtc::CritScope lock(&capturer_crit_);
     capturers_[ssrc] = capturer;
   }
   return true;
 }
 
@@ skipping 73 matching lines @@
 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
   call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
   LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
                   << (mute ? "mute" : "unmute");
   RTC_DCHECK(ssrc != 0);
   rtc::CritScope stream_lock(&stream_crit_);
-  if (send_streams_.find(ssrc) == send_streams_.end()) {
+  auto kv = send_streams_.find(ssrc);

[pbos-webrtc, 2016/02/17 12:09:47]

auto& kv

[nisse-webrtc, 2016/02/17 12:27:49]

Done. const auto& was needed to make the compiler happy. Also found an unrelated
find call in WebRtcVideoChannel2::AddRecvStream, which I fixed similarly. 

+  if (kv == send_streams_.end()) {
     LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
     return false;
   }
 
-  send_streams_[ssrc]->MuteStream(mute);
+  kv->second->MuteStream(mute);
   return true;
 }
 
 // TODO(pbos): Remove SetOptions in favor of SetSendParameters.
-void WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
-  VideoSendParameters new_params = send_params_;
-  new_params.options.SetAll(options);
-  SetSendParameters(send_params_);
+void WebRtcVideoChannel2::SetOptions(uint32_t ssrc,
+                                     const VideoOptions& options) {
+  LOG(LS_INFO) << "SetOptions: ssrc " << ssrc << ": " << options.ToString();
+
+  rtc::CritScope stream_lock(&stream_crit_);
+  auto kv = send_streams_.find(ssrc);

[pbos-webrtc, 2016/02/17 12:09:47]

auto& kv

+  if (kv == send_streams_.end()) {
+    return;
+  }
+  kv->second->SetOptions(options);
 }
 
 void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) {
   MediaChannel::SetInterface(iface);
   // Set the RTP recv/send buffer to a bigger size
   MediaChannel::SetOption(NetworkInterface::ST_RTP,
                           rtc::Socket::OPT_RCVBUF,
                           kVideoRtpBufferSize);
 
   // Speculative change to increase the outbound socket buffer size.
@@ skipping 81 matching lines @@
     this->encoder =
         new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder);
   }
 }
 
 WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
     webrtc::Call* call,
     const StreamParams& sp,
     const webrtc::VideoSendStream::Config& config,
     WebRtcVideoEncoderFactory* external_encoder_factory,
-    const VideoOptions& options,
     int max_bitrate_bps,
     const rtc::Optional<VideoCodecSettings>& codec_settings,
     const std::vector<webrtc::RtpExtension>& rtp_extensions,
     // TODO(deadbeef): 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),
       external_encoder_factory_(external_encoder_factory),
       stream_(NULL),
-      parameters_(config, options, max_bitrate_bps, codec_settings),
+      parameters_(config, send_params.options, max_bitrate_bps, codec_settings),
       pending_encoder_reconfiguration_(false),
       allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
       capturer_(NULL),
       sending_(false),
       muted_(false),
       old_adapt_changes_(0),
       first_frame_timestamp_ms_(0),
       last_frame_timestamp_ms_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
 
@@ skipping 976 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