Set max bitrate for audio send stream based on RtpParameters.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1370 matching lines @@
 
   void RecreateAudioSendStream() {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     if (stream_) {
       call_->DestroyAudioSendStream(stream_);
       stream_ = nullptr;
     }
     RTC_DCHECK(!stream_);
     if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
       config_.min_bitrate_bps = kOpusMinBitrateBps;
-      config_.max_bitrate_bps = kOpusBitrateFbBps;
+
+      // This means that when RtpParameters is reset, we may change the
+      // encoder's bit rate immediately (through call_->CreateAudioSendStream),
+      // meanwhile change the cap to the output of BWE.
+      // TODO(minyue): remove dependency on RtpParameters when the signaling to
+      // BWE is done through a dedicated API.

[stefan-webrtc, 2017/03/24 15:09:14]

I think this is something that should be here even after we add a BWE api. It's
still possible that someone would want to change the allocation of audio
bitrate.

[minyue-webrtc, 2017/03/24 21:13:59]

right, I will remove this todo now.

+      config_.max_bitrate_bps = rtp_parameters_.encodings[0].max_bitrate_bps ?
+          *rtp_parameters_.encodings[0].max_bitrate_bps : kOpusBitrateFbBps;
       // TODO(mflodman): Keep testing this and set proper values.
       // Note: This is an early experiment currently only supported by Opus.
       if (send_side_bwe_with_overhead_) {
         auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs(
             config_.send_codec_spec.codec_inst);
         if (!packet_sizes_ms.empty()) {
           int max_packet_size_ms =
               *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
           int min_packet_size_ms =
               *std::min_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
@@ skipping 12 matching lines @@
 
           // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
           constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
 
           int min_overhead_bps =
               kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;
 
           int max_overhead_bps =
               kOverheadPerPacket * 8 * 1000 / min_packet_size_ms;
 
-          config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps;
-          config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps;
+          config_.min_bitrate_bps += min_overhead_bps;
+          config_.max_bitrate_bps += max_overhead_bps;

[minyue-webrtc, 2017/03/24 21:13:59]

I see a problem here. To make sure that the payload rate will never go above 
rtp_parameters_.encodings[0].max_bitrate_bps, we have to add min_overhead_bps.

The other one is add max_overhead_bps.

[stefan-webrtc, 2017/03/27 12:11:46]

Sounds good to me. As long as the max BWE is set high enough to allow some
overhead it shouldn't affect calls on good networks.

         }
       }
     }
     stream_ = call_->CreateAudioSendStream(config_);
     RTC_CHECK(stream_);
     UpdateSendState();
   }
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::RaceChecker audio_capture_race_checker_;
@@ skipping 1218 matching lines @@
                       ssrc);
   if (it != unsignaled_recv_ssrcs_.end()) {
     unsignaled_recv_ssrcs_.erase(it);
     return true;
   }
   return false;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE