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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13
webrtc/common_audio/audio_converter.h
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@ -15,12 +15,10 @@
#include <memory>
#include "rtc_base/constructor_magic.h"
namespace webrtc {
// Format conversion (remixing and resampling) for audio. Only simple remixing
// conversions are supported: downmix to mono (i.e. |dst_channels| == 1) or
// conversions are supported: downmix to mono (i.e. `dst_channels` == 1) or
// upmix from mono (i.e. |src_channels == 1|).
//
// The source and destination chunks have the same duration in time; specifying
@ -35,8 +33,11 @@ class AudioConverter {
size_t dst_frames);
virtual ~AudioConverter() {}
// Convert |src|, containing |src_size| samples, to |dst|, having a sample
// capacity of |dst_capacity|. Both point to a series of buffers containing
AudioConverter(const AudioConverter&) = delete;
AudioConverter& operator=(const AudioConverter&) = delete;
// Convert `src`, containing `src_size` samples, to `dst`, having a sample
// capacity of `dst_capacity`. Both point to a series of buffers containing
// the samples for each channel. The sizes must correspond to the format
// passed to Create().
virtual void Convert(const float* const* src,
@ -64,8 +65,6 @@ class AudioConverter {
const size_t src_frames_;
const size_t dst_channels_;
const size_t dst_frames_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
};
} // namespace webrtc