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 1335 matching lines @@
     if (!ValidateRtpParameters(parameters)) {
       return false;
     }
     auto send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
                                         parameters.encodings[0].max_bitrate_bps,
                                         send_codec_spec_.codec_inst);
     if (!send_rate) {
       return false;
     }
 
+    const rtc::Optional<int> old_rtp_max_bitrate =
+        rtp_parameters_.encodings[0].max_bitrate_bps;
+
     rtp_parameters_ = parameters;
 
-    // parameters.encodings[0].encodings[0].max_bitrate_bps could have changed.
-    if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
+    if (parameters_.encodings[0].max_bitrate_bps != old_rtp_max_bitrate) {
       // Recreate AudioSendStream with new bit rate.
       config_.send_codec_spec.codec_inst.rate = *send_rate;
       RecreateAudioSendStream();
     } else {
       // parameters.encodings[0].active could have changed.
       UpdateSendState();
     }
     return true;
   }
 
@@ skipping 11 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

[minyue-webrtc, 2017/03/27 13:29:01]

I did not change the TODO here since we may need to reconsider this due to the
overhead issue.

[stefan-webrtc, 2017/03/27 14:04:42]

I still think it makes sense to have this wired up to the bw allocator, so
please rephrase this comment explaining what work is to be done

+      // BWE is done through a dedicated API.
+      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());
 
           // Audio network adaptor will just use 20ms and 60ms frame lengths.
           // The adaptor will only be active for the Opus encoder.
           if (config_.audio_network_adaptor_config &&
               IsCodec(config_.send_codec_spec.codec_inst, kOpusCodecName)) {
 #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
             max_packet_size_ms = 120;
 #else
             max_packet_size_ms = 60;
 #endif
-            min_packet_size_ms = 20;
           }
 
           // 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;
+          // We assume that |config_.max_bitrate_bps| before the next line is
+          // a hard limit on the payload bitrate, so we add min_overhead_bps to
+          // it to ensure that, when overhead is deducted, the payload rate
+          // never goes beyond the limit.
+          // Note: this also means that if a higher overhead is forced, we
+          // cannot reach the limit. We need to investigate this.
+          config_.max_bitrate_bps += min_overhead_bps;
 
-          config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps;
-          config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps;
+          // In contrast to max_bitrate_bps, we let min_bitrate_bps be always

[stefan-webrtc, 2017/03/27 14:04:42]

always be reachable

+          // reachable.
+          config_.min_bitrate_bps += min_overhead_bps;
         }
       }
     }
     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