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:
Arun Raghavan
@ -16,6 +16,9 @@
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namespace webrtc {
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namespace {
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using Agc1Config = AudioProcessing::Config::GainController1;
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using Agc2Config = AudioProcessing::Config::GainController2;
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std::string NoiseSuppressionLevelToString(
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const AudioProcessing::Config::NoiseSuppression::Level& level) {
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switch (level) {
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@ -28,36 +31,19 @@ std::string NoiseSuppressionLevelToString(
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case AudioProcessing::Config::NoiseSuppression::Level::kVeryHigh:
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return "VeryHigh";
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}
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RTC_CHECK_NOTREACHED();
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}
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std::string GainController1ModeToString(
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const AudioProcessing::Config::GainController1::Mode& mode) {
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std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
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switch (mode) {
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case AudioProcessing::Config::GainController1::Mode::kAdaptiveAnalog:
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case Agc1Config::Mode::kAdaptiveAnalog:
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return "AdaptiveAnalog";
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case AudioProcessing::Config::GainController1::Mode::kAdaptiveDigital:
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case Agc1Config::Mode::kAdaptiveDigital:
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return "AdaptiveDigital";
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case AudioProcessing::Config::GainController1::Mode::kFixedDigital:
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case Agc1Config::Mode::kFixedDigital:
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return "FixedDigital";
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}
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}
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std::string GainController2LevelEstimatorToString(
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const AudioProcessing::Config::GainController2::LevelEstimator& level) {
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switch (level) {
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case AudioProcessing::Config::GainController2::LevelEstimator::kRms:
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return "Rms";
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case AudioProcessing::Config::GainController2::LevelEstimator::kPeak:
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return "Peak";
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}
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}
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int GetDefaultMaxInternalRate() {
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#ifdef WEBRTC_ARCH_ARM_FAMILY
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return 32000;
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#else
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return 48000;
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#endif
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RTC_CHECK_NOTREACHED();
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}
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} // namespace
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@ -67,25 +53,92 @@ constexpr int AudioProcessing::kNativeSampleRatesHz[];
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void CustomProcessing::SetRuntimeSetting(
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AudioProcessing::RuntimeSetting setting) {}
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AudioProcessing::Config::Pipeline::Pipeline()
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: maximum_internal_processing_rate(GetDefaultMaxInternalRate()) {}
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bool Agc1Config::operator==(const Agc1Config& rhs) const {
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const auto& analog_lhs = analog_gain_controller;
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const auto& analog_rhs = rhs.analog_gain_controller;
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return enabled == rhs.enabled && mode == rhs.mode &&
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target_level_dbfs == rhs.target_level_dbfs &&
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compression_gain_db == rhs.compression_gain_db &&
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enable_limiter == rhs.enable_limiter &&
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analog_lhs.enabled == analog_rhs.enabled &&
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analog_lhs.startup_min_volume == analog_rhs.startup_min_volume &&
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analog_lhs.clipped_level_min == analog_rhs.clipped_level_min &&
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analog_lhs.enable_digital_adaptive ==
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analog_rhs.enable_digital_adaptive &&
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analog_lhs.clipped_level_step == analog_rhs.clipped_level_step &&
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analog_lhs.clipped_ratio_threshold ==
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analog_rhs.clipped_ratio_threshold &&
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analog_lhs.clipped_wait_frames == analog_rhs.clipped_wait_frames &&
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analog_lhs.clipping_predictor.mode ==
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analog_rhs.clipping_predictor.mode &&
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analog_lhs.clipping_predictor.window_length ==
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analog_rhs.clipping_predictor.window_length &&
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analog_lhs.clipping_predictor.reference_window_length ==
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analog_rhs.clipping_predictor.reference_window_length &&
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analog_lhs.clipping_predictor.reference_window_delay ==
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analog_rhs.clipping_predictor.reference_window_delay &&
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analog_lhs.clipping_predictor.clipping_threshold ==
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analog_rhs.clipping_predictor.clipping_threshold &&
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analog_lhs.clipping_predictor.crest_factor_margin ==
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analog_rhs.clipping_predictor.crest_factor_margin &&
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analog_lhs.clipping_predictor.use_predicted_step ==
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analog_rhs.clipping_predictor.use_predicted_step;
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}
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bool Agc2Config::AdaptiveDigital::operator==(
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const Agc2Config::AdaptiveDigital& rhs) const {
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return enabled == rhs.enabled && headroom_db == rhs.headroom_db &&
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max_gain_db == rhs.max_gain_db &&
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initial_gain_db == rhs.initial_gain_db &&
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max_gain_change_db_per_second == rhs.max_gain_change_db_per_second &&
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max_output_noise_level_dbfs == rhs.max_output_noise_level_dbfs;
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}
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bool Agc2Config::InputVolumeController::operator==(
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const Agc2Config::InputVolumeController& rhs) const {
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return enabled == rhs.enabled;
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}
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bool Agc2Config::operator==(const Agc2Config& rhs) const {
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return enabled == rhs.enabled &&
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fixed_digital.gain_db == rhs.fixed_digital.gain_db &&
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adaptive_digital == rhs.adaptive_digital &&
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input_volume_controller == rhs.input_volume_controller;
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}
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bool AudioProcessing::Config::CaptureLevelAdjustment::operator==(
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const AudioProcessing::Config::CaptureLevelAdjustment& rhs) const {
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return enabled == rhs.enabled && pre_gain_factor == rhs.pre_gain_factor &&
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post_gain_factor == rhs.post_gain_factor &&
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analog_mic_gain_emulation == rhs.analog_mic_gain_emulation;
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}
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bool AudioProcessing::Config::CaptureLevelAdjustment::AnalogMicGainEmulation::
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operator==(const AudioProcessing::Config::CaptureLevelAdjustment::
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AnalogMicGainEmulation& rhs) const {
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return enabled == rhs.enabled && initial_level == rhs.initial_level;
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}
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std::string AudioProcessing::Config::ToString() const {
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char buf[1024];
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char buf[2048];
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rtc::SimpleStringBuilder builder(buf);
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builder << "AudioProcessing::Config{ "
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"pipeline: {"
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"pipeline: { "
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"maximum_internal_processing_rate: "
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<< pipeline.maximum_internal_processing_rate
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<< ", multi_channel_render: " << pipeline.multi_channel_render
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<< ", "
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", multi_channel_capture: "
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<< pipeline.multi_channel_capture
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<< "}, "
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"pre_amplifier: { enabled: "
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<< pre_amplifier.enabled
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<< ", multi_channel_capture: " << pipeline.multi_channel_capture
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<< " }, pre_amplifier: { enabled: " << pre_amplifier.enabled
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<< ", fixed_gain_factor: " << pre_amplifier.fixed_gain_factor
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<< " }, high_pass_filter: { enabled: " << high_pass_filter.enabled
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<< " },capture_level_adjustment: { enabled: "
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<< capture_level_adjustment.enabled
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<< ", pre_gain_factor: " << capture_level_adjustment.pre_gain_factor
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<< ", post_gain_factor: " << capture_level_adjustment.post_gain_factor
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<< ", analog_mic_gain_emulation: { enabled: "
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<< capture_level_adjustment.analog_mic_gain_emulation.enabled
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<< ", initial_level: "
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<< capture_level_adjustment.analog_mic_gain_emulation.initial_level
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<< " }}, high_pass_filter: { enabled: " << high_pass_filter.enabled
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<< " }, echo_canceller: { enabled: " << echo_canceller.enabled
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<< ", mobile_mode: " << echo_canceller.mobile_mode
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<< ", enforce_high_pass_filtering: "
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@ -95,29 +148,62 @@ std::string AudioProcessing::Config::ToString() const {
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<< NoiseSuppressionLevelToString(noise_suppression.level)
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<< " }, transient_suppression: { enabled: "
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<< transient_suppression.enabled
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<< " }, voice_detection: { enabled: " << voice_detection.enabled
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<< " }, gain_controller1: { enabled: " << gain_controller1.enabled
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<< ", mode: " << GainController1ModeToString(gain_controller1.mode)
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<< ", target_level_dbfs: " << gain_controller1.target_level_dbfs
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<< ", compression_gain_db: " << gain_controller1.compression_gain_db
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<< ", enable_limiter: " << gain_controller1.enable_limiter
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<< ", analog_level_minimum: " << gain_controller1.analog_level_minimum
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<< ", analog_level_maximum: " << gain_controller1.analog_level_maximum
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<< " }, gain_controller2: { enabled: " << gain_controller2.enabled
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<< ", analog_gain_controller { enabled: "
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<< gain_controller1.analog_gain_controller.enabled
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<< ", startup_min_volume: "
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<< gain_controller1.analog_gain_controller.startup_min_volume
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<< ", clipped_level_min: "
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<< gain_controller1.analog_gain_controller.clipped_level_min
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<< ", enable_digital_adaptive: "
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<< gain_controller1.analog_gain_controller.enable_digital_adaptive
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<< ", clipped_level_step: "
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<< gain_controller1.analog_gain_controller.clipped_level_step
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<< ", clipped_ratio_threshold: "
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<< gain_controller1.analog_gain_controller.clipped_ratio_threshold
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<< ", clipped_wait_frames: "
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<< gain_controller1.analog_gain_controller.clipped_wait_frames
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<< ", clipping_predictor: { enabled: "
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<< gain_controller1.analog_gain_controller.clipping_predictor.enabled
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<< ", mode: "
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<< gain_controller1.analog_gain_controller.clipping_predictor.mode
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<< ", window_length: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.window_length
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<< ", reference_window_length: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.reference_window_length
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<< ", reference_window_delay: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.reference_window_delay
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<< ", clipping_threshold: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.clipping_threshold
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<< ", crest_factor_margin: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.crest_factor_margin
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<< ", use_predicted_step: "
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<< gain_controller1.analog_gain_controller.clipping_predictor
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.use_predicted_step
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<< " }}}, gain_controller2: { enabled: " << gain_controller2.enabled
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<< ", fixed_digital: { gain_db: "
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<< gain_controller2.fixed_digital.gain_db
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<< " }, adaptive_digital: { enabled: "
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<< gain_controller2.adaptive_digital.enabled << ", level_estimator: "
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<< GainController2LevelEstimatorToString(
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gain_controller2.adaptive_digital.level_estimator)
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<< ", use_saturation_protector: "
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<< gain_controller2.adaptive_digital.use_saturation_protector
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<< ", extra_saturation_margin_db: "
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<< gain_controller2.adaptive_digital.extra_saturation_margin_db
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<< " } }, residual_echo_detector: { enabled: "
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<< residual_echo_detector.enabled
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<< " }, level_estimation: { enabled: " << level_estimation.enabled
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<< " } }";
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<< gain_controller2.adaptive_digital.enabled
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<< ", headroom_db: " << gain_controller2.adaptive_digital.headroom_db
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<< ", max_gain_db: " << gain_controller2.adaptive_digital.max_gain_db
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<< ", initial_gain_db: "
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<< gain_controller2.adaptive_digital.initial_gain_db
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<< ", max_gain_change_db_per_second: "
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<< gain_controller2.adaptive_digital.max_gain_change_db_per_second
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<< ", max_output_noise_level_dbfs: "
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<< gain_controller2.adaptive_digital.max_output_noise_level_dbfs
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<< " }, input_volume_control : { enabled "
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<< gain_controller2.input_volume_controller.enabled << "}}";
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return builder.str();
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}
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