| Index: webrtc/voice_engine/channel.h
|
| diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
|
| index 60e6f6040cd906f15c7a2b831db58d8b6be30580..297f0a8c161288cea2d4d79c7779b44e8922f8bf 100644
|
| --- a/webrtc/voice_engine/channel.h
|
| +++ b/webrtc/voice_engine/channel.h
|
| @@ -17,6 +17,7 @@
|
| #include "webrtc/api/call/audio_sink.h"
|
| #include "webrtc/base/criticalsection.h"
|
| #include "webrtc/base/optional.h"
|
| +#include "webrtc/call/rtp_transport_controller_receive.h"
|
| #include "webrtc/common_audio/resampler/include/push_resampler.h"
|
| #include "webrtc/common_types.h"
|
| #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
|
| @@ -128,6 +129,7 @@ class ChannelState {
|
| class Channel
|
| : public RtpData,
|
| public RtpFeedback,
|
| + public RtpPacketSinkInterface,
|
| public FileCallback, // receiving notification from file player &
|
| // recorder
|
| public Transport,
|
| @@ -203,7 +205,9 @@ class Channel
|
| const PacketTime& packet_time);
|
| // TODO(nisse, solenberg): Delete when VoENetwork is deleted.
|
| int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length);
|
| - void OnRtpPacket(const RtpPacketReceived& packet);
|
| +
|
| + // RtpPacketSinkInterface implementation.
|
| + void OnRtpPacket(const RtpPacketReceived& packet) override;
|
|
|
| // VoEFile
|
| int StartPlayingFileLocally(const char* fileName,
|
| @@ -268,9 +272,7 @@ class Channel
|
| int GetLocalSSRC(unsigned int& ssrc);
|
| int GetRemoteSSRC(unsigned int& ssrc);
|
| int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
|
| - int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
|
| void EnableSendTransportSequenceNumber(int id);
|
| - void EnableReceiveTransportSequenceNumber(int id);
|
|
|
| void RegisterSenderCongestionControlObjects(
|
| RtpPacketSender* rtp_packet_sender,
|
|
|