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

webrtc/voice_engine/channel.h

Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index f26fdb23d46cf16beb6e65022b210ad90f334bf7..38ec522e9a65de16874782ce649b3685915f3205 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -11,13 +11,17 @@
 #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"
@@ -73,6 +77,64 @@ 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
@@ -321,6 +383,13 @@ public:
     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]);
@@ -584,6 +653,9 @@ private:
     // 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