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..5c38b989715828a0418336966c474ebbac66b3c0 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -11,13 +11,18 @@
 #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/base/thread_checker.h"

[the sun, 2015/12/03 11:10:28]

Not needed.

[stefan-webrtc, 2015/12/04 10:31:43]

Done.

+#include "webrtc/call/congestion_controller.h"

[the sun, 2015/12/03 11:10:28]

Not needed, in fact VoE::Channel shouldn't rely on anything in call/

[stefan-webrtc, 2015/12/04 10:31:43]

Done.

 #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"

[the sun, 2015/12/03 11:10:28]

Not needed?

[stefan-webrtc, 2015/12/04 10:31:43]

Done.

+#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"
@@ -321,6 +326,13 @@ public:
     int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
     int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
     int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
+    void EnableSendTransportSequenceNumber(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]);
@@ -456,6 +468,10 @@ protected:
     void OnIncomingFractionLoss(int fraction_lost);
 
 private:
+ RtpRtcp* CreateRtpRtcp(

[the sun, 2015/12/03 11:10:28]

Please indent like the rest of this class, it'll make it easier to clean up
later.

[stefan-webrtc, 2015/12/04 10:31:43]

Done.

+     RtpPacketSender* packet_sender,
+     TransportSequenceNumberAllocator* sequence_number_allocator,
+     TransportFeedbackObserver* transport_feedback_callback);
     bool ReceivePacket(const uint8_t* packet, size_t packet_length,
                        const RTPHeader& header, bool in_order);
     bool HandleRtxPacket(const uint8_t* packet,
@@ -584,6 +600,8 @@ private:
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
+
+    PacketRouter* packet_router_;
 };
 
 }  // namespace voe