| Index: webrtc/media/engine/fakewebrtcvoiceengine.h
|
| diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| index b5ad81c6a98818cabde818d3b4a78c60cc3be03c..33629160bc116d1139c6f9dc59e43b9c6bac1921 100644
|
| --- a/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| +++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| @@ -456,22 +456,10 @@ class FakeWebRtcVoiceEngine
|
| WEBRTC_STUB(SetPlayoutDevice, (int));
|
| WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
|
| WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
|
| - WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
|
| - recording_sample_rate_ = samples_per_sec;
|
| - return 0;
|
| - }
|
| - WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) {
|
| - *samples_per_sec = recording_sample_rate_;
|
| - return 0;
|
| - }
|
| - WEBRTC_FUNC(SetPlayoutSampleRate, (unsigned int samples_per_sec)) {
|
| - playout_sample_rate_ = samples_per_sec;
|
| - return 0;
|
| - }
|
| - WEBRTC_FUNC_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec)) {
|
| - *samples_per_sec = playout_sample_rate_;
|
| - return 0;
|
| - }
|
| + WEBRTC_STUB(SetRecordingSampleRate, (unsigned int samples_per_sec));
|
| + WEBRTC_STUB_CONST(RecordingSampleRate, (unsigned int* samples_per_sec));
|
| + WEBRTC_STUB(SetPlayoutSampleRate, (unsigned int samples_per_sec));
|
| + WEBRTC_STUB_CONST(PlayoutSampleRate, (unsigned int* samples_per_sec));
|
| WEBRTC_STUB(EnableBuiltInAEC, (bool enable));
|
| virtual bool BuiltInAECIsAvailable() const { return false; }
|
| WEBRTC_STUB(EnableBuiltInAGC, (bool enable));
|
| @@ -729,8 +717,6 @@ class FakeWebRtcVoiceEngine
|
| webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
|
| webrtc::AgcConfig agc_config_;
|
| int playout_fail_channel_ = -1;
|
| - int recording_sample_rate_ = -1;
|
| - int playout_sample_rate_ = -1;
|
| FakeAudioProcessing audio_processing_;
|
| };
|
|
|
|
|