Add Cast PLI support on sender side.

media/cast/test/cast_benchmarks.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 //
 // This program benchmarks the theoretical throughput of the cast library.
 // It runs using a fake clock, simulated network and fake codecs. This allows
 // tests to run much faster than real time.
 // To run the program, run:
 // $ ./out/Release/cast_benchmarks | tee benchmarkoutput.asc
 // This may take a while, when it is done, you can view the data with
@@ skipping 78 matching lines @@
   void Init(CastTransportSender* transport,
             uint64_t* encoded_video_bytes,
             uint64_t* encoded_audio_bytes) {
     transport_.reset(transport);
     encoded_video_bytes_ = encoded_video_bytes;
     encoded_audio_bytes_ = encoded_audio_bytes;
   }
 
   void InitializeAudio(const CastTransportRtpConfig& config,
                        const RtcpCastMessageCallback& cast_message_cb,
-                       const RtcpRttCallback& rtt_cb) final {
+                       const RtcpRttCallback& rtt_cb,
+                       const RtcpPliCallback& key_frame_cb) final {
     audio_ssrc_ = config.ssrc;
-    transport_->InitializeAudio(config, cast_message_cb, rtt_cb);
+    transport_->InitializeAudio(config, cast_message_cb, rtt_cb, key_frame_cb);
   }
 
   void InitializeVideo(const CastTransportRtpConfig& config,
                        const RtcpCastMessageCallback& cast_message_cb,
-                       const RtcpRttCallback& rtt_cb) final {
+                       const RtcpRttCallback& rtt_cb,
+                       const RtcpPliCallback& key_frame_cb) final {
     video_ssrc_ = config.ssrc;
-    transport_->InitializeVideo(config, cast_message_cb, rtt_cb);
+    transport_->InitializeVideo(config, cast_message_cb, rtt_cb, key_frame_cb);
   }
 
   void InsertFrame(uint32_t ssrc, const EncodedFrame& frame) final {
     if (ssrc == audio_ssrc_) {
       *encoded_audio_bytes_ += frame.data.size();
     } else if (ssrc == video_ssrc_) {
       *encoded_video_bytes_ += frame.data.size();
     }
     transport_->InsertFrame(ssrc, frame);
   }
@@ skipping 36 matching lines @@
 
   void AddRtcpEvents(
       const ReceiverRtcpEventSubscriber::RtcpEvents& rtcp_events) final {
     transport_->AddRtcpEvents(rtcp_events);
   }
 
   void AddRtpReceiverReport(const RtcpReportBlock& rtp_report_block) final {
     transport_->AddRtpReceiverReport(rtp_report_block);
   }
 
+  void AddPli(const RtcpPliMessage& pli_message) final {
+    transport_->AddPli(pli_message);
+  }
+
   void SendRtcpFromRtpReceiver() final {
     transport_->SendRtcpFromRtpReceiver();
   }
 
   void SetOptions(const base::DictionaryValue& options) final {}
 
  private:
   scoped_ptr<CastTransportSender> transport_;
   uint32_t audio_ssrc_, video_ssrc_;
   uint64_t* encoded_video_bytes_;
@@ skipping 550 matching lines @@
   media::cast::CastBenchmark benchmark;
   if (getenv("PROFILE_FILE")) {
     std::string profile_file(getenv("PROFILE_FILE"));
     base::debug::StartProfiling(profile_file);
     benchmark.Run();
     base::debug::StopProfiling();
   } else {
     benchmark.Run();
   }
 }