AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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4
webrtc/modules/audio_processing/aec3/render_delay_buffer.h
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@ -16,6 +16,7 @@
#include <vector>
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/block.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/render_buffer.h"
@ -41,8 +42,7 @@ class RenderDelayBuffer {
virtual void Reset() = 0;
// Inserts a block into the buffer.
virtual BufferingEvent Insert(
const std::vector<std::vector<std::vector<float>>>& block) = 0;
virtual BufferingEvent Insert(const Block& block) = 0;
// Updates the buffers one step based on the specified buffer delay. Returns
// an enum indicating whether there was a special event that occurred.