Chromium Code Reviews| Index: webrtc/media/base/fakemediaengine.h |
| diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h |
| index afd262bb5e92263951cc85e7be3256e5d867801c..c3f16603cae69a33097af52adc304029403668e9 100644 |
| --- a/webrtc/media/base/fakemediaengine.h |
| +++ b/webrtc/media/base/fakemediaengine.h |
| @@ -21,7 +21,7 @@ |
| #include "webrtc/audio_sink.h" |
| #include "webrtc/base/buffer.h" |
| #include "webrtc/base/stringutils.h" |
| -#include "webrtc/media/base/audiorenderer.h" |
| +#include "webrtc/media/base/audiosource.h" |
| #include "webrtc/media/base/mediaengine.h" |
| #include "webrtc/media/base/rtputils.h" |
| #include "webrtc/media/base/streamparams.h" |
| @@ -253,14 +253,12 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
| set_playout(playout); |
| return true; |
| } |
| - virtual bool SetSend(SendFlags flag) { |
| - return set_sending(flag != SEND_NOTHING); |
| - } |
| + virtual void SetSend(bool send) { set_sending(send); } |
|
pthatcher1
2016/03/08 17:46:51
I like having it not be flags.
|
| virtual bool SetAudioSend(uint32_t ssrc, |
| bool enable, |
| const AudioOptions* options, |
| - AudioRenderer* renderer) { |
| - if (!SetLocalRenderer(ssrc, renderer)) { |
| + AudioSource* source) { |
| + if (!SetLocalSource(ssrc, source)) { |
| return false; |
| } |
| if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) { |
| @@ -338,15 +336,14 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
| } |
| private: |
| - class VoiceChannelAudioSink : public AudioRenderer::Sink { |
| + class VoiceChannelAudioSink : public AudioSource::Sink { |
| public: |
| - explicit VoiceChannelAudioSink(AudioRenderer* renderer) |
| - : renderer_(renderer) { |
| - renderer_->SetSink(this); |
| + explicit VoiceChannelAudioSink(AudioSource* source) : source_(source) { |
| + source_->SetSink(this); |
| } |
| virtual ~VoiceChannelAudioSink() { |
| - if (renderer_) { |
| - renderer_->SetSink(NULL); |
| + if (source_) { |
| + source_->SetSink(nullptr); |
| } |
| } |
| void OnData(const void* audio_data, |
| @@ -354,11 +351,11 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
| int sample_rate, |
| size_t number_of_channels, |
| size_t number_of_frames) override {} |
| - void OnClose() override { renderer_ = NULL; } |
| - AudioRenderer* renderer() const { return renderer_; } |
| + void OnClose() override { source_ = nullptr; } |
| + AudioSource* source() const { return source_; } |
| private: |
| - AudioRenderer* renderer_; |
| + AudioSource* source_; |
| }; |
| bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) { |
| @@ -383,19 +380,19 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
| options_.SetAll(options); |
| return true; |
| } |
| - bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer) { |
| - auto it = local_renderers_.find(ssrc); |
| - if (renderer) { |
| - if (it != local_renderers_.end()) { |
| - ASSERT(it->second->renderer() == renderer); |
| + bool SetLocalSource(uint32_t ssrc, AudioSource* source) { |
| + auto it = local_sinks_.find(ssrc); |
| + if (source) { |
| + if (it != local_sinks_.end()) { |
| + ASSERT(it->second->source() == source); |
| } else { |
| - local_renderers_.insert(std::make_pair( |
| - ssrc, new VoiceChannelAudioSink(renderer))); |
| + local_sinks_.insert( |
| + std::make_pair(ssrc, new VoiceChannelAudioSink(source))); |
| } |
| } else { |
| - if (it != local_renderers_.end()) { |
| + if (it != local_sinks_.end()) { |
| delete it->second; |
| - local_renderers_.erase(it); |
| + local_sinks_.erase(it); |
| } |
| } |
| return true; |
| @@ -408,7 +405,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> { |
| std::vector<DtmfInfo> dtmf_info_queue_; |
| int time_since_last_typing_; |
| AudioOptions options_; |
| - std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_; |
| + std::map<uint32_t, VoiceChannelAudioSink*> local_sinks_; |
| std::unique_ptr<webrtc::AudioSinkInterface> sink_; |
| }; |