Move BitrateAllocator from BitrateController logic to Call.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 #include "webrtc/video/video_send_stream.h"
 
 #include <algorithm>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/congestion_controller.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
+#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/pacing/include/packet_router.h"
 #include "webrtc/video/video_capture_input.h"
 #include "webrtc/video_engine/call_stats.h"
 #include "webrtc/video_engine/encoder_state_feedback.h"
 #include "webrtc/video_engine/payload_router.h"
 #include "webrtc/video_engine/vie_channel.h"
 #include "webrtc/video_engine/vie_encoder.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
-class BitrateAllocator;
 class PacedSender;
 class RtcpIntraFrameObserver;
 class TransportFeedbackObserver;
 
 std::string
 VideoSendStream::Config::EncoderSettings::ToString() const {
   std::stringstream ss;
   ss << "{payload_name: " << payload_name;
   ss << ", payload_type: " << payload_type;
   ss << ", encoder: " << (encoder != nullptr ? "(VideoEncoder)" : "nullptr");
@@ skipping 61 matching lines @@
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 VideoSendStream::VideoSendStream(
     int num_cpu_cores,
     ProcessThread* module_process_thread,
     CallStats* call_stats,
     CongestionController* congestion_controller,
+    BitrateAllocator* bitrate_allocator,
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config,
     const std::map<uint32_t, RtpState>& suspended_ssrcs)
     : stats_proxy_(Clock::GetRealTimeClock(), config),
       transport_adapter_(config.send_transport),
       encoded_frame_proxy_(config.post_encode_callback),
       config_(config),
       suspended_ssrcs_(suspended_ssrcs),
       module_process_thread_(module_process_thread),
       call_stats_(call_stats),
@@ skipping 11 matching lines @@
           congestion_controller_->GetTransportFeedbackObserver();
       break;
     }
   }
 
   const std::vector<uint32_t>& ssrcs = config.rtp.ssrcs;
 
   vie_encoder_.reset(new ViEEncoder(
       num_cpu_cores, module_process_thread_, &stats_proxy_,
       config.pre_encode_callback, congestion_controller_->pacer(),
-      congestion_controller_->bitrate_allocator()));
+      bitrate_allocator));
   RTC_CHECK(vie_encoder_->Init());
 
   vie_channel_.reset(new ViEChannel(
       num_cpu_cores, config.send_transport, module_process_thread_,
       encoder_feedback_->GetRtcpIntraFrameObserver(),
       congestion_controller_->GetBitrateController()->
           CreateRtcpBandwidthObserver(),
       transport_feedback_observer,
       congestion_controller_->GetRemoteBitrateEstimator(false),
       call_stats_->rtcp_rtt_stats(), congestion_controller_->pacer(),
@@ skipping 367 matching lines @@
   uint32_t jitter;
   int64_t rtt_ms;
   if (vie_channel_->GetSendRtcpStatistics(&frac_lost, &cumulative_lost,
                                           &extended_max_sequence_number,
                                           &jitter, &rtt_ms) == 0) {
     return rtt_ms;
   }
   return -1;
 }
 
+int VideoSendStream::GetPaddingNeededBps() const {
+  return vie_encoder_->GetPaddingNeededBps();
+}
+
 bool VideoSendStream::SetSendCodec(VideoCodec video_codec) {
   static const int kEncoderMinBitrate = 30;
   if (video_codec.maxBitrate == 0) {
     // Unset max bitrate -> cap to one bit per pixel.
     video_codec.maxBitrate =
         (video_codec.width * video_codec.height * video_codec.maxFramerate) /
         1000;
   }
 
   if (video_codec.minBitrate < kEncoderMinBitrate)
@@ skipping 21 matching lines @@
   vie_encoder_->SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc