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

61
webrtc/modules/audio_processing/agc2/limiter.cc
View File

@ -46,22 +46,20 @@ void InterpolateFirstSubframe(float last_factor,
void ComputePerSampleSubframeFactors(
const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
int samples_per_channel,
rtc::ArrayView<float> per_sample_scaling_factors) {
const int num_subframes = scaling_factors.size() - 1;
const int subframe_size =
rtc::CheckedDivExact(samples_per_channel, num_subframes);
MonoView<float> per_sample_scaling_factors) {
const size_t num_subframes = scaling_factors.size() - 1;
const int subframe_size = rtc::CheckedDivExact(
SamplesPerChannel(per_sample_scaling_factors), num_subframes);
// Handle first sub-frame differently in case of attack.
const bool is_attack = scaling_factors[0] > scaling_factors[1];
if (is_attack) {
InterpolateFirstSubframe(
scaling_factors[0], scaling_factors[1],
rtc::ArrayView<float>(
per_sample_scaling_factors.subview(0, subframe_size)));
per_sample_scaling_factors.subview(0, subframe_size));
}
for (int i = is_attack ? 1 : 0; i < num_subframes; ++i) {
for (size_t i = is_attack ? 1 : 0; i < num_subframes; ++i) {
const int subframe_start = i * subframe_size;
const float scaling_start = scaling_factors[i];
const float scaling_end = scaling_factors[i + 1];
@ -73,39 +71,36 @@ void ComputePerSampleSubframeFactors(
}
}
void ScaleSamples(rtc::ArrayView<const float> per_sample_scaling_factors,
AudioFrameView<float> signal) {
void ScaleSamples(MonoView<const float> per_sample_scaling_factors,
DeinterleavedView<float> signal) {
const int samples_per_channel = signal.samples_per_channel();
RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size());
for (int i = 0; i < signal.num_channels(); ++i) {
rtc::ArrayView<float> channel = signal.channel(i);
RTC_DCHECK_EQ(samples_per_channel,
SamplesPerChannel(per_sample_scaling_factors));
for (size_t i = 0; i < signal.num_channels(); ++i) {
MonoView<float> channel = signal[i];
for (int j = 0; j < samples_per_channel; ++j) {
channel[j] = rtc::SafeClamp(channel[j] * per_sample_scaling_factors[j],
kMinFloatS16Value, kMaxFloatS16Value);
}
}
}
void CheckLimiterSampleRate(int sample_rate_hz) {
// Check that per_sample_scaling_factors_ is large enough.
RTC_DCHECK_LE(sample_rate_hz,
kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
}
} // namespace
Limiter::Limiter(int sample_rate_hz,
ApmDataDumper* apm_data_dumper,
Limiter::Limiter(ApmDataDumper* apm_data_dumper,
size_t samples_per_channel,
absl::string_view histogram_name)
: interp_gain_curve_(apm_data_dumper, histogram_name),
level_estimator_(sample_rate_hz, apm_data_dumper),
level_estimator_(samples_per_channel, apm_data_dumper),
apm_data_dumper_(apm_data_dumper) {
CheckLimiterSampleRate(sample_rate_hz);
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
}
Limiter::~Limiter() = default;
void Limiter::Process(AudioFrameView<float> signal) {
void Limiter::Process(DeinterleavedView<float> signal) {
RTC_DCHECK_LE(signal.samples_per_channel(),
kMaximalNumberOfSamplesPerChannel);
const std::array<float, kSubFramesInFrame> level_estimate =
level_estimator_.ComputeLevel(signal);
@ -116,13 +111,9 @@ void Limiter::Process(AudioFrameView<float> signal) {
return interp_gain_curve_.LookUpGainToApply(x);
});
const int samples_per_channel = signal.samples_per_channel();
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
auto per_sample_scaling_factors = rtc::ArrayView<float>(
&per_sample_scaling_factors_[0], samples_per_channel);
ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel,
per_sample_scaling_factors);
MonoView<float> per_sample_scaling_factors(&per_sample_scaling_factors_[0],
signal.samples_per_channel());
ComputePerSampleSubframeFactors(scaling_factors_, per_sample_scaling_factors);
ScaleSamples(per_sample_scaling_factors, signal);
last_scaling_factor_ = scaling_factors_.back();
@ -139,9 +130,9 @@ InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {
return interp_gain_curve_.get_stats();
}
void Limiter::SetSampleRate(int sample_rate_hz) {
CheckLimiterSampleRate(sample_rate_hz);
level_estimator_.SetSampleRate(sample_rate_hz);
void Limiter::SetSamplesPerChannel(size_t samples_per_channel) {
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
level_estimator_.SetSamplesPerChannel(samples_per_channel);
}
void Limiter::Reset() {