AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M131 release

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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
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27
webrtc/modules/audio_processing/gain_controller2.cc
View File

@ -13,6 +13,7 @@
#include <memory>
#include <utility>
#include "api/audio/audio_frame.h"
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/cpu_features.h"
@ -63,11 +64,11 @@ struct SpeechLevel {
};
// Computes the audio levels for the first channel in `frame`.
AudioLevels ComputeAudioLevels(AudioFrameView<float> frame,
AudioLevels ComputeAudioLevels(DeinterleavedView<float> frame,
ApmDataDumper& data_dumper) {
float peak = 0.0f;
float rms = 0.0f;
for (const auto& x : frame.channel(0)) {
for (const auto& x : frame[0]) {
peak = std::max(std::fabs(x), peak);
rms += x * x;
}
@ -94,7 +95,9 @@ GainController2::GainController2(
fixed_gain_applier_(
/*hard_clip_samples=*/false,
/*initial_gain_factor=*/DbToRatio(config.fixed_digital.gain_db)),
limiter_(sample_rate_hz, &data_dumper_, /*histogram_name_prefix=*/"Agc2"),
limiter_(&data_dumper_,
SampleRateToDefaultChannelSize(sample_rate_hz),
/*histogram_name_prefix=*/"Agc2"),
calls_since_last_limiter_log_(0) {
RTC_DCHECK(Validate(config));
data_dumper_.InitiateNewSetOfRecordings();
@ -153,7 +156,7 @@ void GainController2::SetFixedGainDb(float gain_db) {
void GainController2::Analyze(int applied_input_volume,
const AudioBuffer& audio_buffer) {
recommended_input_volume_ = absl::nullopt;
recommended_input_volume_ = std::nullopt;
RTC_DCHECK_GE(applied_input_volume, 0);
RTC_DCHECK_LE(applied_input_volume, 255);
@ -164,10 +167,10 @@ void GainController2::Analyze(int applied_input_volume,
}
}
void GainController2::Process(absl::optional<float> speech_probability,
void GainController2::Process(std::optional<float> speech_probability,
bool input_volume_changed,
AudioBuffer* audio) {
recommended_input_volume_ = absl::nullopt;
recommended_input_volume_ = std::nullopt;
data_dumper_.DumpRaw("agc2_applied_input_volume_changed",
input_volume_changed);
@ -179,8 +182,8 @@ void GainController2::Process(absl::optional<float> speech_probability,
saturation_protector_->Reset();
}
AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
audio->num_frames());
DeinterleavedView<float> float_frame = audio->view();
// Compute speech probability.
if (vad_) {
// When the VAD component runs, `speech_probability` should not be specified
@ -200,13 +203,13 @@ void GainController2::Process(absl::optional<float> speech_probability,
// Compute audio, noise and speech levels.
AudioLevels audio_levels = ComputeAudioLevels(float_frame, data_dumper_);
absl::optional<float> noise_rms_dbfs;
std::optional<float> noise_rms_dbfs;
if (noise_level_estimator_) {
// TODO(bugs.webrtc.org/7494): Pass `audio_levels` to remove duplicated
// computation in `noise_level_estimator_`.
noise_rms_dbfs = noise_level_estimator_->Analyze(float_frame);
}
absl::optional<SpeechLevel> speech_level;
std::optional<SpeechLevel> speech_level;
if (speech_level_estimator_) {
RTC_DCHECK(speech_probability.has_value());
speech_level_estimator_->Update(
@ -225,8 +228,8 @@ void GainController2::Process(absl::optional<float> speech_probability,
input_volume_controller_->RecommendInputVolume(
*speech_probability,
speech_level->is_confident
? absl::optional<float>(speech_level->rms_dbfs)
: absl::nullopt);
? std::optional<float>(speech_level->rms_dbfs)
: std::nullopt);
}
}