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. |