Chromium Code Reviews| Index: webrtc/audio/audio_receive_stream.h |
| diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h |
| index adac88387613a43a7f6220567a75c8e881fe313c..f53a138298ce517ca25e7cbb201be8436777ecd4 100644 |
| --- a/webrtc/audio/audio_receive_stream.h |
| +++ b/webrtc/audio/audio_receive_stream.h |
| @@ -18,7 +18,10 @@ |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/base/thread_checker.h" |
| #include "webrtc/call/audio_receive_stream.h" |
| +#include "webrtc/call/rtp_transport_controller_receive.h" |
| #include "webrtc/call/syncable.h" |
| +#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
| +#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
|
the sun
2017/04/18 09:59:00
We only need forward decl here.
nisse-webrtc
2017/04/19 08:35:43
Done.
|
| namespace webrtc { |
| class PacketRouter; |
| @@ -32,7 +35,8 @@ class ChannelProxy; |
| namespace internal { |
| class AudioSendStream; |
| -class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
| +class AudioReceiveStream final : public webrtc::RtpPacketReceiverInterface, |
| + public webrtc::AudioReceiveStream, |
| public AudioMixer::Source, |
| public Syncable { |
| public: |
| @@ -50,8 +54,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
| void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
| void SetGain(float gain) override; |
| - // TODO(nisse): Intended to be part of an RtpPacketReceiver interface. |
| - void OnRtpPacket(const RtpPacketReceived& packet); |
| + // webrtc::RtpPacketReceiverInterface implementation |
| + bool OnRtpPacketReceive(RtpPacketReceived* packet) override; |
| // AudioMixer::Source |
| AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, |
| @@ -78,6 +82,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
| rtc::ThreadChecker worker_thread_checker_; |
| rtc::ThreadChecker module_process_thread_checker_; |
| const webrtc::AudioReceiveStream::Config config_; |
| + RtpHeaderExtensionMap rtp_header_extensions_; |
| + |
| rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| std::unique_ptr<voe::ChannelProxy> channel_proxy_; |