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 |