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 44ff7327ffeb849184869c7bfadc921438093821..18014868b9c1c6dd2633683adac6b8f1a382c29e 100644 |
--- a/webrtc/modules/audio_processing/include/audio_processing.h |
+++ b/webrtc/modules/audio_processing/include/audio_processing.h |
@@ -293,12 +293,12 @@ class AudioProcessing { |
// Initialize with unpacked parameters. See Initialize() above for details. |
// |
// TODO(mgraczyk): Remove once clients are updated to use the new interface. |
- virtual int Initialize(int input_sample_rate_hz, |
- int output_sample_rate_hz, |
- int reverse_sample_rate_hz, |
- ChannelLayout input_layout, |
- ChannelLayout output_layout, |
- ChannelLayout reverse_layout) = 0; |
+ virtual int Initialize(int capture_input_sample_rate_hz, |
+ int capture_output_sample_rate_hz, |
+ int render_sample_rate_hz, |
+ ChannelLayout capture_input_layout, |
+ ChannelLayout capture_output_layout, |
+ ChannelLayout render_input_layout) = 0; |
// Pass down additional options which don't have explicit setters. This |
// ensures the options are applied immediately. |
@@ -385,8 +385,8 @@ class AudioProcessing { |
// Accepts deinterleaved float audio with the range [-1, 1]. Each element of |
// |data| points to a channel buffer, arranged according to |reverse_config|. |
virtual int ProcessReverseStream(const float* const* src, |
- const StreamConfig& reverse_input_config, |
- const StreamConfig& reverse_output_config, |
+ const StreamConfig& input_config, |
+ const StreamConfig& output_config, |
float* const* dest) = 0; |
// This must be called if and only if echo processing is enabled. |