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; |