Prepare the AudioSendStream to be hooked up to send-side BWE.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
+#include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
+#include "webrtc/call/congestion_controller.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/file_player.h"
 #include "webrtc/modules/utility/include/file_recorder.h"
 #include "webrtc/voice_engine/dtmf_inband.h"
 #include "webrtc/voice_engine/dtmf_inband_queue.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/level_indicator.h"
@@ skipping 35 matching lines @@
 struct SenderInfo;
 
 namespace voe {
 
 class OutputMixer;
 class Statistics;
 class StatisticsProxy;
 class TransmitMixer;
 class VoERtcpObserver;
 
+class PacketSenderProxy : public RtpPacketSender,

[the sun, 2015/11/30 12:37:20]

You can put this class in an anonymous namespace in the .cc

[stefan-webrtc, 2015/11/30 15:22:02]

Done.

+                          public TransportFeedbackObserver,
+                          public TransportSequenceNumberAllocator {
+ public:
+  void SetCongestionControlObjects(
+      RtpPacketSender* rtp_packet_sender,
+      TransportFeedbackObserver* transport_feedback_observer,
+      TransportSequenceNumberAllocator* seq_num_allocator) {
+    rtc::CritScope lock(&crit_);
+    packet_sender_ = rtp_packet_sender;
+    feedback_observer_ = transport_feedback_observer;
+    seq_num_allocator_ = seq_num_allocator;
+  }
+
+  // Implements RtpPacketSender.
+  void InsertPacket(Priority priority,
+                    uint32_t ssrc,
+                    uint16_t sequence_number,
+                    int64_t capture_time_ms,
+                    size_t bytes,
+                    bool retransmission) override {
+    rtc::CritScope lock(&crit_);
+    if (packet_sender_ == nullptr)

[the sun, 2015/11/30 12:37:20]

Looks like you have a crash bug here; packet_sender_ may be uninitialized.

[stefan-webrtc, 2015/11/30 15:22:02]

Definitely could crash here, good catch.

+      return;
+    packet_sender_->InsertPacket(priority, ssrc, sequence_number,
+                                 capture_time_ms, bytes, retransmission);
+  }
+
+  // Implements TransportFeedbackObserver.
+  void AddPacket(uint16_t sequence_number,

[the sun, 2015/11/30 12:37:20]

I don't see where this is called from? Where is it hooked up?

[stefan-webrtc, 2015/11/30 15:22:02]

Here:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

[the sun, 2015/12/01 10:25:36]

Yes, but I don't see how transport_feedback_observer_ is being configured. You
don't set it in the RtpRtcp::Configuration.

[stefan-webrtc, 2015/12/01 16:19:33]

Gooood catch. :)

It would have been caught later when we actually hook up voice to the BWE,
currently there is no test for that.

[the sun, 2015/12/01 16:48:23]

Is it possible to add a test? At some level?

[stefan-webrtc, 2015/12/02 16:14:17]

It is, but it requires setting up a call and monitoring the outgoing packets. At
least that is the way I was planning on doing it. We already have such a test
for video and my plan is to adapt those tests to include audio. The tests can be
found here:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

Do you think it is worthwhile to write a smaller version of this now before
adapting the above test?

[stefan-webrtc, 2015/12/02 16:17:39]

https://code.google.com/p/chromium/codesearch#chromium/src/third_party/webrtc...

that link is probably a better example.

+                 size_t length,
+                 bool was_paced) override {
+    rtc::CritScope lock(&crit_);
+    if (feedback_observer_ == nullptr)
+      return;
+    feedback_observer_->AddPacket(sequence_number, length, was_paced);
+  }
+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override {
+    rtc::CritScope lock(&crit_);
+    if (feedback_observer_ == nullptr)
+      return;
+    feedback_observer_->OnTransportFeedback(feedback);
+  }
+
+  // Implements TransportSequenceNumberAllocator.
+  uint16_t AllocateSequenceNumber() override {
+    rtc::CritScope lock(&crit_);
+    RTC_DCHECK(seq_num_allocator_);
+    return seq_num_allocator_->AllocateSequenceNumber();
+  }
+
+ private:
+  rtc::CriticalSection crit_;

[the sun, 2015/11/30 12:37:20]

So what are the threads this can be called on? Please document using a
ThreadChecker.

[stefan-webrtc, 2015/11/30 15:22:02]

Done.

+  RtpPacketSender* packet_sender_ GUARDED_BY(&crit_);
+  TransportFeedbackObserver* feedback_observer_ GUARDED_BY(&crit_);
+  TransportSequenceNumberAllocator* seq_num_allocator_ GUARDED_BY(&crit_);
+};
+
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
 // Example: a member can be set on thread T1 and read by an internal audio
 // thread T2. Accessing the member via this class ensures that we are
 // safe and also avoid TSan v2 warnings.
 class ChannelState {
  public:
     struct State {
         State() : rx_apm_is_enabled(false),
                   input_external_media(false),
@@ skipping 228 matching lines @@
 #endif
 
     // VoERTP_RTCP
     int SetLocalSSRC(unsigned int ssrc);
     int GetLocalSSRC(unsigned int& ssrc);
     int GetRemoteSSRC(unsigned int& ssrc);
     int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
     int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
     int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
     int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
+    void SetSendTransportSequenceNumber(int id);
+
+    void SetCongestionControlObjects(
+        RtpPacketSender* rtp_packet_sender,
+        TransportFeedbackObserver* transport_feedback_observer,
+        PacketRouter* packet_router);
+
     void SetRTCPStatus(bool enable);
     int GetRTCPStatus(bool& enabled);
     int SetRTCP_CNAME(const char cName[256]);
     int GetRemoteRTCP_CNAME(char cName[256]);
     int GetRemoteRTCPData(unsigned int& NTPHigh, unsigned int& NTPLow,
                           unsigned int& timestamp,
                           unsigned int& playoutTimestamp, unsigned int* jitter,
                           unsigned short* fractionLost);
     int SendApplicationDefinedRTCPPacket(unsigned char subType,
                                          unsigned int name, const char* data,
@@ skipping 243 matching lines @@
     bool _RxVadDetection;
     bool _rxAgcIsEnabled;
     bool _rxNsIsEnabled;
     bool restored_packet_in_use_;
     // RtcpBandwidthObserver
     rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
+
+    rtc::scoped_ptr<PacketSenderProxy> packet_sender_proxy_;
+    PacketRouter* packet_router_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_