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

Issue 1500633002: Refactor handling of AudioOptions. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
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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
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