Apply VideoOptions per stream.

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 604 matching lines @@
     webrtc::Call* call,
     const VideoOptions& options,
     const std::vector<VideoCodec>& recv_codecs,
     WebRtcVideoEncoderFactory* external_encoder_factory,
     WebRtcVideoDecoderFactory* external_decoder_factory)
     : call_(call),
       unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
       external_encoder_factory_(external_encoder_factory),
       external_decoder_factory_(external_decoder_factory) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  SetDefaultOptions();
-  options_.SetAll(options);
-  if (options_.cpu_overuse_detection)
-    signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
+
+  signal_cpu_adaptation_ = options.cpu_overuse_detection.value_or(true);
   rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
   sending_ = false;
   default_send_ssrc_ = 0;
   SetRecvCodecs(recv_codecs);
 }
 
-void WebRtcVideoChannel2::SetDefaultOptions() {
-  options_.cpu_overuse_detection = rtc::Optional<bool>(true);
-  options_.dscp = rtc::Optional<bool>(false);
-  options_.suspend_below_min_bitrate = rtc::Optional<bool>(false);
-  options_.screencast_min_bitrate = rtc::Optional<int>(0);
+void WebRtcVideoChannel2::SetDefaultOptions(VideoOptions *options) {
+  options->cpu_overuse_detection = rtc::Optional<bool>(true);
+  options->dscp = rtc::Optional<bool>(false);
+  options->suspend_below_min_bitrate = rtc::Optional<bool>(false);
+  options->screencast_min_bitrate = rtc::Optional<int>(0);
 }
 
 WebRtcVideoChannel2::~WebRtcVideoChannel2() {
   for (auto& kv : send_streams_)
     delete kv.second;
   for (auto& kv : receive_streams_)
     delete kv.second;
 }
 
 bool WebRtcVideoChannel2::CodecIsExternallySupported(
@@ skipping 60 matching lines @@
   return false;
 }
 
 bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
   TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters");
   LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
   // TODO(pbos): Refactor this to only recreate the send streams once
   // instead of 4 times.
   if (!SetSendCodecs(params.codecs) ||
       !SetSendRtpHeaderExtensions(params.extensions) ||
-      !SetMaxSendBandwidth(params.max_bandwidth_bps) ||
-      !SetOptions(params.options)) {
+      !SetMaxSendBandwidth(params.max_bandwidth_bps)) {
     return false;
   }
+  SetSharedOptions(params.options);
   if (send_params_.rtcp.reduced_size != params.rtcp.reduced_size) {
     rtc::CritScope stream_lock(&stream_crit_);
     for (auto& kv : send_streams_) {
       kv.second->SetSendParameters(params);
     }
   }
   send_params_ = params;
   return true;
 }
 
@@ skipping 182 matching lines @@
 }
 
 bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable,
                                        const VideoOptions* options) {
   // 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) {
-    return SetOptions(*options);
+    return SetOptions(ssrc, *options);
   } else {
     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.";
@@ skipping 24 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;
 
+  // Initial options
+  VideoOptions options;
+  SetDefaultOptions(&options);
+
   WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
-      call_, sp, config, external_encoder_factory_, options_,
+      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);
   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 "
@@ skipping 109 matching lines @@
 
   // Set up A/V sync group based on sync label.
   config.sync_group = sp.sync_label;
 
   config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false;
   config.rtp.transport_cc =
       send_codec_ ? HasTransportCc(send_codec_->codec) : false;
 
   receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
       call_, sp, config, external_decoder_factory_, default_stream,
-      recv_codecs_, options_.disable_prerenderer_smoothing.value_or(false));
+      recv_codecs_,
+      // TODO(nisse): Used to pass

[pbos-webrtc, 2016/01/28 14:55:44]

This should be const on construction, this needs to propagate from
RTCConfiguration. If this isn't broken right now we're missing test coverage.

+      // options_.disable_prerenderer_smoothing.value_or(false),
+      // unclear if it needs to be configurable here, but there's no
+      // other method to change it.
+      false);
 
   return true;
 }
 
 void WebRtcVideoChannel2::ConfigureReceiverRtp(
     webrtc::VideoReceiveStream::Config* config,
     const StreamParams& sp) const {
   uint32_t ssrc = sp.first_ssrc();
 
   config->rtp.remote_ssrc = ssrc;
@@ skipping 346 matching lines @@
       bitrate_config_.min_bitrate_bps > max_bitrate_bps) {
     bitrate_config_.min_bitrate_bps = max_bitrate_bps;
   }
   call_->SetBitrateConfig(bitrate_config_);
   rtc::CritScope stream_lock(&stream_crit_);
   for (auto& kv : send_streams_)
     kv.second->SetMaxBitrateBps(max_bitrate_bps);
   return true;
 }
 
-bool WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetOptions");
-  LOG(LS_INFO) << "SetOptions: " << options.ToString();
-  VideoOptions old_options = options_;
-  options_.SetAll(options);
-  if (options_ == old_options) {
-    // No new options to set.
-    return true;
-  }
+void WebRtcVideoChannel2::SetSharedOptions(const VideoOptions& options) {
   {
     rtc::CritScope lock(&capturer_crit_);
-    if (options_.cpu_overuse_detection)
-      signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
+    if (options.cpu_overuse_detection)
+      signal_cpu_adaptation_ = *options.cpu_overuse_detection;
   }
   rtc::DiffServCodePoint dscp =
-      options_.dscp.value_or(false) ? rtc::DSCP_AF41 : rtc::DSCP_DEFAULT;
+      options.dscp.value_or(false) ? rtc::DSCP_AF41 : rtc::DSCP_DEFAULT;
   MediaChannel::SetDscp(dscp);
+}
+
+bool WebRtcVideoChannel2::SetOptions(uint32_t ssrc,
+                                     const VideoOptions& options) {
+  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetOptions");
+  LOG(LS_INFO) << "SetOptions: ssrc " << ssrc << ": " << options.ToString();
+  SetSharedOptions(options);
+
   rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
-           send_streams_.begin();
-       it != send_streams_.end(); ++it) {
-    it->second->SetOptions(options_);
+  if (send_streams_.find(ssrc) == send_streams_.end()) {
+    return false;
   }
+  send_streams_[ssrc]->SetOptions(options);
+
   return true;
 }
 
 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);
 
@@ skipping 299 matching lines @@
     bool apply_rotation) {
   rtc::CritScope cs(&lock_);
   if (capturer_ == NULL)
     return;
 
   capturer_->SetApplyRotation(apply_rotation);
 }
 
 void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
     const VideoOptions& options) {
+
   rtc::CritScope cs(&lock_);
   if (parameters_.codec_settings) {
     LOG(LS_INFO) << "SetCodecAndOptions because of SetOptions; options="
                  << options.ToString();
     SetCodecAndOptions(*parameters_.codec_settings, options);
   } else {
     parameters_.options = options;
   }
 }
 
@@ skipping 828 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO