| 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.
|
|
|