| Index: webrtc/modules/audio_processing/audio_processing_impl.h
|
| diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
|
| index 15c6f7572f9a6e5a8fedebd2308676b1b166be43..eeab34f8743ceb74f4fc8337af61cd3affd4626f 100644
|
| --- a/webrtc/modules/audio_processing/audio_processing_impl.h
|
| +++ b/webrtc/modules/audio_processing/audio_processing_impl.h
|
| @@ -68,16 +68,12 @@ class AudioProcessingImpl : public AudioProcessing {
|
| ChannelLayout reverse_layout) override;
|
| int Initialize(const ProcessingConfig& processing_config) override;
|
| void SetExtraOptions(const Config& config) override;
|
| - int set_sample_rate_hz(int rate) override;
|
| - int input_sample_rate_hz() const override;
|
| - int sample_rate_hz() const override;
|
| int proc_sample_rate_hz() const override;
|
| int proc_split_sample_rate_hz() const override;
|
| int num_input_channels() const override;
|
| int num_output_channels() const override;
|
| int num_reverse_channels() const override;
|
| void set_output_will_be_muted(bool muted) override;
|
| - bool output_will_be_muted() const override;
|
| int ProcessStream(AudioFrame* frame) override;
|
| int ProcessStream(const float* const* src,
|
| size_t samples_per_channel,
|
| @@ -106,7 +102,6 @@ class AudioProcessingImpl : public AudioProcessing {
|
| void set_delay_offset_ms(int offset) override;
|
| int delay_offset_ms() const override;
|
| void set_stream_key_pressed(bool key_pressed) override;
|
| - bool stream_key_pressed() const override;
|
| int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
|
| int StartDebugRecording(FILE* handle) override;
|
| int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
|
|
|