AnotherFoxGuy/webrtc-audio-processing
1
0
Fork 0
Code Pull Requests Activity

Bump to WebRTC M131 release

Browse Source 
Ongoing fixes and improvements, transient suppressor is gone. Also,
dropping isac because it doesn't seem to be useful, and is just build
system deadweight now.

Upstream references:

  Version: 131.0.6778.200
  WebRTC: 79aff54b0fa9238ce3518dd9eaf9610cd6f22e82
  Chromium: 2a19506ad24af755f2a215a4c61f775393e0db42
This commit is contained in:
Arun Raghavan
2024-12-24 19:32:07 -05:00
parent 8bdb53d91c
commit b5c48b97f6
263 changed files with 4628 additions and 20416 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
webrtc/modules/audio_processing/audio_buffer.h
View File

@ -17,8 +17,10 @@
#include <memory>
#include <vector>
#include "api/audio/audio_processing.h"
#include "api/audio/audio_view.h"
#include "common_audio/channel_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "common_audio/include/audio_util.h"
namespace webrtc {
@ -32,7 +34,8 @@ enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
class AudioBuffer {
public:
static const int kSplitBandSize = 160;
static const int kMaxSampleRate = 384000;
// TODO(tommi): Remove this (`AudioBuffer::kMaxSampleRate`) constant.
static const int kMaxSampleRate = webrtc::kMaxSampleRateHz;
AudioBuffer(size_t input_rate,
size_t input_num_channels,
size_t buffer_rate,
@ -56,6 +59,13 @@ class AudioBuffer {
// reset at each call to CopyFrom or InterleaveFrom.
void set_num_channels(size_t num_channels);
// Returns a DeinterleavedView<> over the channel data.
DeinterleavedView<float> view() {
return DeinterleavedView<float>(
num_channels_ && buffer_num_frames_ ? channels()[0] : nullptr,
buffer_num_frames_, num_channels_);
}
size_t num_channels() const { return num_channels_; }
size_t num_frames() const { return buffer_num_frames_; }
size_t num_frames_per_band() const { return num_split_frames_; }