| Index: webrtc/modules/audio_processing/include/audio_processing.h
|
| diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
|
| index 5eb3b62f98ca09dec1872516b1c588627c525acb..318b2f89533652b6fd1ad896d269f78deb0d004d 100644
|
| --- a/webrtc/modules/audio_processing/include/audio_processing.h
|
| +++ b/webrtc/modules/audio_processing/include/audio_processing.h
|
| @@ -264,15 +264,6 @@ class AudioProcessing {
|
| // ensures the options are applied immediately.
|
| virtual void SetExtraOptions(const Config& config) = 0;
|
|
|
| - // DEPRECATED.
|
| - // TODO(ajm): Remove after Chromium has upgraded to using Initialize().
|
| - virtual int set_sample_rate_hz(int rate) = 0;
|
| - // TODO(ajm): Remove after voice engine no longer requires it to resample
|
| - // the reverse stream to the forward rate.
|
| - virtual int input_sample_rate_hz() const = 0;
|
| - // TODO(ajm): Remove after Chromium no longer depends on it.
|
| - virtual int sample_rate_hz() const = 0;
|
| -
|
| // TODO(ajm): Only intended for internal use. Make private and friend the
|
| // necessary classes?
|
| virtual int proc_sample_rate_hz() const = 0;
|
| @@ -286,7 +277,6 @@ class AudioProcessing {
|
| // but some components may change behavior based on this information.
|
| // Default false.
|
| virtual void set_output_will_be_muted(bool muted) = 0;
|
| - virtual bool output_will_be_muted() const = 0;
|
|
|
| // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
|
| // this is the near-end (or captured) audio.
|
| @@ -387,7 +377,6 @@ class AudioProcessing {
|
| // Call to signal that a key press occurred (true) or did not occur (false)
|
| // with this chunk of audio.
|
| virtual void set_stream_key_pressed(bool key_pressed) = 0;
|
| - virtual bool stream_key_pressed() const = 0;
|
|
|
| // Sets a delay |offset| in ms to add to the values passed in through
|
| // set_stream_delay_ms(). May be positive or negative.
|
|
|