| Index: talk/media/webrtc/webrtcvoiceengine.h
|
| diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
|
| index 3222861b21e7c61b5a087b59076396e7942896ad..1de4fb972af70a9453d198ffec5fc45c2e1ed88c 100644
|
| --- a/talk/media/webrtc/webrtcvoiceengine.h
|
| +++ b/talk/media/webrtc/webrtcvoiceengine.h
|
| @@ -29,7 +29,6 @@
|
| #define TALK_MEDIA_WEBRTCVOICEENGINE_H_
|
|
|
| #include <map>
|
| -#include <set>
|
| #include <string>
|
| #include <vector>
|
|
|
| @@ -72,9 +71,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
| VoiceMediaChannel* CreateChannel(webrtc::Call* call,
|
| const AudioOptions& options);
|
|
|
| - AudioOptions GetOptions() const { return options_; }
|
| - bool SetOptions(const AudioOptions& options);
|
| - bool SetDevices(const Device* in_device, const Device* out_device);
|
| bool GetOutputVolume(int* level);
|
| bool SetOutputVolume(int level);
|
| int GetInputLevel();
|
| @@ -118,15 +114,11 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
| // ignored. This allows us to selectively turn on and off different options
|
| // easily at any time.
|
| bool ApplyOptions(const AudioOptions& options);
|
| + void SetDefaultDevices();
|
|
|
| // webrtc::TraceCallback:
|
| void Print(webrtc::TraceLevel level, const char* trace, int length) override;
|
|
|
| - // Given the device type, name, and id, find device id. Return true and
|
| - // set the output parameter rtc_id if successful.
|
| - bool FindWebRtcAudioDeviceId(
|
| - bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
|
| -
|
| void StartAecDump(const std::string& filename);
|
| int CreateVoEChannel();
|
|
|
| @@ -138,16 +130,13 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
| rtc::scoped_refptr<webrtc::AudioState> audio_state_;
|
| // The external audio device manager
|
| webrtc::AudioDeviceModule* adm_ = nullptr;
|
| - bool is_dumping_aec_ = false;
|
| std::vector<AudioCodec> codecs_;
|
| std::vector<WebRtcVoiceMediaChannel*> channels_;
|
| - webrtc::AgcConfig default_agc_config_;
|
| -
|
| webrtc::Config voe_config_;
|
| -
|
| bool initialized_ = false;
|
| - AudioOptions options_;
|
| + bool is_dumping_aec_ = false;
|
|
|
| + webrtc::AgcConfig default_agc_config_;
|
| // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
|
| // values, and apply them in case they are missing in the audio options. We
|
| // need to do this because SetExtraOptions() will revert to defaults for
|
|
|