Implementing auto pausing of video streams.

webrtc/call/call.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 <string.h>
 
+#include <algorithm>
 #include <map>
 #include <memory>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
@@ skipping 670 matching lines @@
   uint32_t pacer_bitrate_bps =
       std::max(target_bitrate_bps, allocated_bitrate_bps);
   {
     rtc::CritScope lock(&bitrate_crit_);
     // We only update these stats if we have send streams, and assume that
     // OnNetworkChanged is called roughly with a fixed frequency.
     estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000;
     pacer_bitrate_sum_kbits_ += pacer_bitrate_bps / 1000;
     ++num_bitrate_updates_;
   }
+
+  // Make sure to not ask for more padding than the current BWE allows for.
+  pad_up_to_bitrate_bps = std::min(static_cast<uint32_t>(pad_up_to_bitrate_bps),
+                                   target_bitrate_bps);
   congestion_controller_->SetAllocatedSendBitrate(allocated_bitrate_bps,
                                                   pad_up_to_bitrate_bps);
 }
 
 void Call::ConfigureSync(const std::string& sync_group) {
   // Set sync only if there was no previous one.
   if (voice_engine() == nullptr || sync_group.empty())
     return;
 
   AudioReceiveStream* sync_audio_stream = nullptr;
@@ skipping 135 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc