Chromium Code Reviews| Index: talk/media/base/fakemediaengine.h |
| diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h |
| index b1f09aaf64908fb4e62bf001dc42d7ca282c82dc..7580c2bfae25e23abc0595af2f66d35e44541f0a 100644 |
| --- a/talk/media/base/fakemediaengine.h |
| +++ b/talk/media/base/fakemediaengine.h |
| @@ -35,12 +35,13 @@ |
| #include <vector> |
| #include "talk/media/base/audiorenderer.h" |
| +#include "talk/media/base/audiorenderer.h" |
|
perkj_webrtc
2015/12/13 19:26:47
nit: remove extra
tommi
2015/12/13 19:46:33
Done in the last patch set
|
| #include "talk/media/base/mediaengine.h" |
| #include "talk/media/base/rtputils.h" |
| #include "talk/media/base/streamparams.h" |
| -#include "webrtc/p2p/base/sessiondescription.h" |
| #include "webrtc/base/buffer.h" |
| #include "webrtc/base/stringutils.h" |
| +#include "webrtc/p2p/base/sessiondescription.h" |
| namespace cricket { |
| @@ -346,6 +347,11 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
| virtual bool GetStats(VoiceMediaInfo* info) { return false; } |
| + virtual void SetRawAudioSink(uint32_t ssrc, |
| + rtc::scoped_ptr<webrtc::AudioSink> sink) { |
| + sink_ = std::move(sink); |
| + } |
| + |
| private: |
| class VoiceChannelAudioSink : public AudioRenderer::Sink { |
| public: |
| @@ -418,6 +424,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
| int time_since_last_typing_; |
| AudioOptions options_; |
| std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_; |
| + rtc::scoped_ptr<webrtc::AudioSink> sink_; |
| }; |
| // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo. |