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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22
webrtc/modules/audio_processing/agc/agc_manager_direct.cc
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@ -69,11 +69,11 @@ using AnalogAgcConfig =
// string. Returns an unspecified value if the field trial is not specified, if
// disabled or if it cannot be parsed. Example:
// 'WebRTC-Audio-2ndAgcMinMicLevelExperiment/Enabled-80' => returns 80.
absl::optional<int> GetMinMicLevelOverride() {
std::optional<int> GetMinMicLevelOverride() {
constexpr char kMinMicLevelFieldTrial[] =
"WebRTC-Audio-2ndAgcMinMicLevelExperiment";
if (!webrtc::field_trial::IsEnabled(kMinMicLevelFieldTrial)) {
return absl::nullopt;
return std::nullopt;
}
const auto field_trial_string =
webrtc::field_trial::FindFullName(kMinMicLevelFieldTrial);
@ -84,7 +84,7 @@ absl::optional<int> GetMinMicLevelOverride() {
} else {
RTC_LOG(LS_WARNING) << "[agc] Invalid parameter for "
<< kMinMicLevelFieldTrial << ", ignored.";
return absl::nullopt;
return std::nullopt;
}
}
@ -189,8 +189,8 @@ void MonoAgc::Initialize() {
}
void MonoAgc::Process(rtc::ArrayView<const int16_t> audio,
absl::optional<int> rms_error_override) {
new_compression_to_set_ = absl::nullopt;
std::optional<int> rms_error_override) {
new_compression_to_set_ = std::nullopt;
if (check_volume_on_next_process_) {
check_volume_on_next_process_ = false;
@ -617,13 +617,13 @@ void AgcManagerDirect::AnalyzePreProcess(const AudioBuffer& audio_buffer) {
}
void AgcManagerDirect::Process(const AudioBuffer& audio_buffer) {
Process(audio_buffer, /*speech_probability=*/absl::nullopt,
/*speech_level_dbfs=*/absl::nullopt);
Process(audio_buffer, /*speech_probability=*/std::nullopt,
/*speech_level_dbfs=*/std::nullopt);
}
void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
absl::optional<float> speech_probability,
absl::optional<float> speech_level_dbfs) {
std::optional<float> speech_probability,
std::optional<float> speech_level_dbfs) {
AggregateChannelLevels();
const int volume_after_clipping_handling = recommended_input_volume_;
@ -632,7 +632,7 @@ void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
}
const size_t num_frames_per_band = audio_buffer.num_frames_per_band();
absl::optional<int> rms_error_override = absl::nullopt;
std::optional<int> rms_error_override = std::nullopt;
if (speech_probability.has_value() && speech_level_dbfs.has_value()) {
rms_error_override =
GetSpeechLevelErrorDb(*speech_level_dbfs, *speech_probability);
@ -656,7 +656,7 @@ void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
}
}
absl::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
std::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
return new_compressions_to_set_[channel_controlling_gain_];
}