Bump to WebRTC M120 release
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:
@ -26,10 +26,11 @@ rtc_library("audio_frame_operations") {
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"../../api/audio:audio_frame_api",
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"../../common_audio",
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"../../rtc_base:checks",
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"../../rtc_base:deprecation",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:logging",
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"../../rtc_base:safe_conversions",
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"../../system_wrappers:field_trial",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
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}
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if (rtc_include_tests) {
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@ -44,7 +45,9 @@ if (rtc_include_tests) {
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":audio_frame_operations",
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"../../api/audio:audio_frame_api",
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"../../rtc_base:checks",
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"../../rtc_base:rtc_base_approved",
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"../../rtc_base:logging",
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"../../rtc_base:macromagic",
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"../../rtc_base:stringutils",
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"../../test:field_trial",
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"../../test:test_support",
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"//testing/gtest",
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@ -131,8 +131,8 @@ void AudioFrameOperations::DownmixChannels(const int16_t* src_audio,
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return;
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}
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RTC_NOTREACHED() << "src_channels: " << src_channels
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<< ", dst_channels: " << dst_channels;
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RTC_DCHECK_NOTREACHED() << "src_channels: " << src_channels
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<< ", dst_channels: " << dst_channels;
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}
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void AudioFrameOperations::DownmixChannels(size_t dst_channels,
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@ -149,8 +149,8 @@ void AudioFrameOperations::DownmixChannels(size_t dst_channels,
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int err = QuadToStereo(frame);
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RTC_DCHECK_EQ(err, 0);
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} else {
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RTC_NOTREACHED() << "src_channels: " << frame->num_channels_
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<< ", dst_channels: " << dst_channels;
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RTC_DCHECK_NOTREACHED() << "src_channels: " << frame->num_channels_
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<< ", dst_channels: " << dst_channels;
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}
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}
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@ -169,10 +169,10 @@ void AudioFrameOperations::UpmixChannels(size_t target_number_of_channels,
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if (!frame->muted()) {
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// Up-mixing done in place. Going backwards through the frame ensure nothing
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// is irrevocably overwritten.
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int16_t* frame_data = frame->mutable_data();
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for (int i = frame->samples_per_channel_ - 1; i >= 0; i--) {
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for (size_t j = 0; j < target_number_of_channels; ++j) {
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frame->mutable_data()[target_number_of_channels * i + j] =
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frame->data()[i];
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frame_data[target_number_of_channels * i + j] = frame_data[i];
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}
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}
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}
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@ -222,14 +222,14 @@ void AudioFrameOperations::Mute(AudioFrame* frame,
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size_t end = count;
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float start_g = 0.0f;
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if (current_frame_muted) {
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// Fade out the last |count| samples of frame.
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// Fade out the last `count` samples of frame.
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RTC_DCHECK(!previous_frame_muted);
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start = frame->samples_per_channel_ - count;
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end = frame->samples_per_channel_;
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start_g = 1.0f;
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inc = -inc;
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} else {
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// Fade in the first |count| samples of frame.
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// Fade in the first `count` samples of frame.
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RTC_DCHECK(previous_frame_muted);
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}
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@ -14,8 +14,8 @@
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#include <stddef.h>
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#include <stdint.h>
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#include "absl/base/attributes.h"
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#include "api/audio/audio_frame.h"
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#include "rtc_base/deprecation.h"
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namespace webrtc {
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@ -24,38 +24,40 @@ namespace webrtc {
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// than a class.
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class AudioFrameOperations {
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public:
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// Add samples in |frame_to_add| with samples in |result_frame|
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// putting the results in |results_frame|. The fields
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// |vad_activity_| and |speech_type_| of the result frame are
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// updated. If |result_frame| is empty (|samples_per_channel_|==0),
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// the samples in |frame_to_add| are added to it. The number of
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// Add samples in `frame_to_add` with samples in `result_frame`
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// putting the results in `results_frame`. The fields
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// `vad_activity_` and `speech_type_` of the result frame are
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// updated. If `result_frame` is empty (`samples_per_channel_`==0),
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// the samples in `frame_to_add` are added to it. The number of
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// channels and number of samples per channel must match except when
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// |result_frame| is empty.
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// `result_frame` is empty.
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static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
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// |frame.num_channels_| will be updated. This version checks for sufficient
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// buffer size and that |num_channels_| is mono. Use UpmixChannels
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// `frame.num_channels_` will be updated. This version checks for sufficient
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// buffer size and that `num_channels_` is mono. Use UpmixChannels
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// instead. TODO(bugs.webrtc.org/8649): remove.
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RTC_DEPRECATED static int MonoToStereo(AudioFrame* frame);
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ABSL_DEPRECATED("bugs.webrtc.org/8649")
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static int MonoToStereo(AudioFrame* frame);
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// |frame.num_channels_| will be updated. This version checks that
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// |num_channels_| is stereo. Use DownmixChannels
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// `frame.num_channels_` will be updated. This version checks that
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// `num_channels_` is stereo. Use DownmixChannels
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// instead. TODO(bugs.webrtc.org/8649): remove.
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RTC_DEPRECATED static int StereoToMono(AudioFrame* frame);
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ABSL_DEPRECATED("bugs.webrtc.org/8649")
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static int StereoToMono(AudioFrame* frame);
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// Downmixes 4 channels |src_audio| to stereo |dst_audio|. This is an in-place
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// operation, meaning |src_audio| and |dst_audio| may point to the same
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// Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place
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// operation, meaning `src_audio` and `dst_audio` may point to the same
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// buffer.
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static void QuadToStereo(const int16_t* src_audio,
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size_t samples_per_channel,
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int16_t* dst_audio);
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// |frame.num_channels_| will be updated. This version checks that
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// |num_channels_| is 4 channels.
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// `frame.num_channels_` will be updated. This version checks that
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// `num_channels_` is 4 channels.
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static int QuadToStereo(AudioFrame* frame);
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// Downmixes |src_channels| |src_audio| to |dst_channels| |dst_audio|.
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// This is an in-place operation, meaning |src_audio| and |dst_audio|
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// Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`.
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// This is an in-place operation, meaning `src_audio` and `dst_audio`
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// may point to the same buffer. Supported channel combinations are
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// Stereo to Mono, Quad to Mono, and Quad to Stereo.
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static void DownmixChannels(const int16_t* src_audio,
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@ -64,27 +66,27 @@ class AudioFrameOperations {
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size_t dst_channels,
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int16_t* dst_audio);
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// |frame.num_channels_| will be updated. This version checks that
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// |num_channels_| and |dst_channels| are valid and performs relevant downmix.
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// `frame.num_channels_` will be updated. This version checks that
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// `num_channels_` and `dst_channels` are valid and performs relevant downmix.
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// Supported channel combinations are N channels to Mono, and Quad to Stereo.
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static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
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// |frame.num_channels_| will be updated. This version checks that
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// |num_channels_| and |dst_channels| are valid and performs relevant
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// `frame.num_channels_` will be updated. This version checks that
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// `num_channels_` and `dst_channels` are valid and performs relevant
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// downmix. Supported channel combinations are Mono to N
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// channels. The single channel is replicated.
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static void UpmixChannels(size_t target_number_of_channels,
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AudioFrame* frame);
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// Swap the left and right channels of |frame|. Fails silently if |frame| is
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// Swap the left and right channels of `frame`. Fails silently if `frame` is
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// not stereo.
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static void SwapStereoChannels(AudioFrame* frame);
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// Conditionally zero out contents of |frame| for implementing audio mute:
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// |previous_frame_muted| && |current_frame_muted| - Zero out whole frame.
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// |previous_frame_muted| && !|current_frame_muted| - Fade-in at frame start.
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// !|previous_frame_muted| && |current_frame_muted| - Fade-out at frame end.
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// !|previous_frame_muted| && !|current_frame_muted| - Leave frame untouched.
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// Conditionally zero out contents of `frame` for implementing audio mute:
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// `previous_frame_muted` && `current_frame_muted` - Zero out whole frame.
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// `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start.
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// !`previous_frame_muted` && `current_frame_muted` - Fade-out at frame end.
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// !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched.
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static void Mute(AudioFrame* frame,
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bool previous_frame_muted,
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bool current_frame_muted);
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@ -92,7 +94,7 @@ class AudioFrameOperations {
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// Zero out contents of frame.
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static void Mute(AudioFrame* frame);
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// Halve samples in |frame|.
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// Halve samples in `frame`.
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static void ApplyHalfGain(AudioFrame* frame);
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static int Scale(float left, float right, AudioFrame* frame);
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