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, |