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Unified Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

Issue 1360773002: Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 3ea59a80c14a5244ce467562f3bb55567b416cd4..bc95839392974a688b33896c3d64c53680bb1d12 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -187,7 +187,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
- public webrtc::VoEFile, public webrtc::VoEHardware,
+ public webrtc::VoEHardware,
public webrtc::VoEExternalMedia, public webrtc::VoENetEqStats,
public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
public webrtc::VoEVideoSync, public webrtc::VoEVolumeControl {
@@ -209,7 +209,6 @@ class FakeWebRtcVoiceEngine
volume_scale(1.0),
volume_pan_left(1.0),
volume_pan_right(1.0),
- file(false),
vad(false),
codec_fec(false),
max_encoding_bandwidth(0),
@@ -241,7 +240,6 @@ class FakeWebRtcVoiceEngine
float volume_scale;
float volume_pan_left;
float volume_pan_right;
- bool file;
bool vad;
bool codec_fec;
int max_encoding_bandwidth;
@@ -723,64 +721,6 @@ class FakeWebRtcVoiceEngine
return 0;
}
- // webrtc::VoEFile
- WEBRTC_FUNC(StartPlayingFileLocally, (int channel, const char* fileNameUTF8,
- bool loop, webrtc::FileFormats format,
- float volumeScaling, int startPointMs,
- int stopPointMs)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->file = true;
- return 0;
- }
- WEBRTC_FUNC(StartPlayingFileLocally, (int channel, webrtc::InStream* stream,
- webrtc::FileFormats format,
- float volumeScaling, int startPointMs,
- int stopPointMs)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->file = true;
- return 0;
- }
- WEBRTC_FUNC(StopPlayingFileLocally, (int channel)) {
- WEBRTC_CHECK_CHANNEL(channel);
- channels_[channel]->file = false;
- return 0;
- }
- WEBRTC_FUNC(IsPlayingFileLocally, (int channel)) {
- WEBRTC_CHECK_CHANNEL(channel);
- return (channels_[channel]->file) ? 1 : 0;
- }
- WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
- const char* fileNameUTF8,
- bool loop,
- bool mixWithMicrophone,
- webrtc::FileFormats format,
- float volumeScaling));
- WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
- webrtc::InStream* stream,
- bool mixWithMicrophone,
- webrtc::FileFormats format,
- float volumeScaling));
- WEBRTC_STUB(StopPlayingFileAsMicrophone, (int channel));
- WEBRTC_STUB(IsPlayingFileAsMicrophone, (int channel));
- WEBRTC_STUB(StartRecordingPlayout, (int channel, const char* fileNameUTF8,
- webrtc::CodecInst* compression,
- int maxSizeBytes));
- WEBRTC_STUB(StartRecordingPlayout, (int channel, webrtc::OutStream* stream,
- webrtc::CodecInst* compression));
- WEBRTC_STUB(StopRecordingPlayout, (int channel));
- WEBRTC_FUNC(StartRecordingMicrophone, (const char* fileNameUTF8,
- webrtc::CodecInst* compression,
- int maxSizeBytes)) {
- return 0;
- }
- WEBRTC_FUNC(StartRecordingMicrophone, (webrtc::OutStream* stream,
- webrtc::CodecInst* compression)) {
- return 0;
- }
- WEBRTC_FUNC(StopRecordingMicrophone, ()) {
- return 0;
- }
-
// webrtc::VoEHardware
WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
return GetNumDevices(num);
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