b5c48b97f6
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
371 lines
12 KiB
C++
371 lines
12 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/gain_control_impl.h"
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#include <cstdint>
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#include <optional>
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#include "api/audio/audio_processing.h"
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#include "modules/audio_processing/agc/legacy/gain_control.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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typedef void Handle;
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namespace {
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int16_t MapSetting(GainControl::Mode mode) {
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switch (mode) {
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case GainControl::kAdaptiveAnalog:
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return kAgcModeAdaptiveAnalog;
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case GainControl::kAdaptiveDigital:
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return kAgcModeAdaptiveDigital;
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case GainControl::kFixedDigital:
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return kAgcModeFixedDigital;
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}
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RTC_DCHECK_NOTREACHED();
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return -1;
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}
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// Applies the sub-frame `gains` to all the bands in `out` and clamps the output
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// in the signed 16 bit range.
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void ApplyDigitalGain(const int32_t gains[11],
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size_t num_bands,
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float* const* out) {
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constexpr float kScaling = 1.f / 65536.f;
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constexpr int kNumSubSections = 16;
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constexpr float kOneByNumSubSections = 1.f / kNumSubSections;
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float gains_scaled[11];
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for (int k = 0; k < 11; ++k) {
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gains_scaled[k] = gains[k] * kScaling;
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}
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for (size_t b = 0; b < num_bands; ++b) {
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float* out_band = out[b];
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for (int k = 0, sample = 0; k < 10; ++k) {
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const float delta =
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(gains_scaled[k + 1] - gains_scaled[k]) * kOneByNumSubSections;
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float gain = gains_scaled[k];
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for (int n = 0; n < kNumSubSections; ++n, ++sample) {
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RTC_DCHECK_EQ(k * kNumSubSections + n, sample);
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out_band[sample] *= gain;
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out_band[sample] =
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std::min(32767.f, std::max(-32768.f, out_band[sample]));
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gain += delta;
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}
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}
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}
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}
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} // namespace
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struct GainControlImpl::MonoAgcState {
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MonoAgcState() {
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state = WebRtcAgc_Create();
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RTC_CHECK(state);
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}
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~MonoAgcState() {
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RTC_DCHECK(state);
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WebRtcAgc_Free(state);
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}
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MonoAgcState(const MonoAgcState&) = delete;
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MonoAgcState& operator=(const MonoAgcState&) = delete;
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int32_t gains[11];
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Handle* state;
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};
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int GainControlImpl::instance_counter_ = 0;
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GainControlImpl::GainControlImpl()
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: data_dumper_(new ApmDataDumper(instance_counter_)),
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mode_(kAdaptiveAnalog),
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minimum_capture_level_(0),
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maximum_capture_level_(255),
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limiter_enabled_(true),
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target_level_dbfs_(3),
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compression_gain_db_(9),
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analog_capture_level_(0),
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was_analog_level_set_(false),
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stream_is_saturated_(false) {}
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GainControlImpl::~GainControlImpl() = default;
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void GainControlImpl::ProcessRenderAudio(
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rtc::ArrayView<const int16_t> packed_render_audio) {
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for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
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WebRtcAgc_AddFarend(mono_agcs_[ch]->state, packed_render_audio.data(),
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packed_render_audio.size());
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}
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}
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void GainControlImpl::PackRenderAudioBuffer(
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const AudioBuffer& audio,
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std::vector<int16_t>* packed_buffer) {
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RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
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std::array<int16_t, AudioBuffer::kMaxSplitFrameLength>
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mixed_16_kHz_render_data;
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rtc::ArrayView<const int16_t> mixed_16_kHz_render(
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mixed_16_kHz_render_data.data(), audio.num_frames_per_band());
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if (audio.num_channels() == 1) {
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FloatS16ToS16(audio.split_bands_const(0)[kBand0To8kHz],
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audio.num_frames_per_band(), mixed_16_kHz_render_data.data());
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} else {
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const int num_channels = static_cast<int>(audio.num_channels());
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for (size_t i = 0; i < audio.num_frames_per_band(); ++i) {
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int32_t sum = 0;
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for (int ch = 0; ch < num_channels; ++ch) {
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sum += FloatS16ToS16(audio.split_channels_const(kBand0To8kHz)[ch][i]);
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}
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mixed_16_kHz_render_data[i] = sum / num_channels;
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}
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}
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packed_buffer->clear();
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packed_buffer->insert(
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packed_buffer->end(), mixed_16_kHz_render.data(),
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(mixed_16_kHz_render.data() + audio.num_frames_per_band()));
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}
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int GainControlImpl::AnalyzeCaptureAudio(const AudioBuffer& audio) {
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RTC_DCHECK(num_proc_channels_);
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RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength, audio.num_frames_per_band());
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RTC_DCHECK_EQ(audio.num_channels(), *num_proc_channels_);
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RTC_DCHECK_LE(*num_proc_channels_, mono_agcs_.size());
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int16_t split_band_data[AudioBuffer::kMaxNumBands]
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[AudioBuffer::kMaxSplitFrameLength];
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int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
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split_band_data[0], split_band_data[1], split_band_data[2]};
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if (mode_ == kAdaptiveAnalog) {
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for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
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capture_levels_[ch] = analog_capture_level_;
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audio.ExportSplitChannelData(ch, split_bands);
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int err =
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WebRtcAgc_AddMic(mono_agcs_[ch]->state, split_bands,
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audio.num_bands(), audio.num_frames_per_band());
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if (err != AudioProcessing::kNoError) {
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return AudioProcessing::kUnspecifiedError;
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}
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}
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} else if (mode_ == kAdaptiveDigital) {
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for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
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int32_t capture_level_out = 0;
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audio.ExportSplitChannelData(ch, split_bands);
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int err =
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WebRtcAgc_VirtualMic(mono_agcs_[ch]->state, split_bands,
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audio.num_bands(), audio.num_frames_per_band(),
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analog_capture_level_, &capture_level_out);
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capture_levels_[ch] = capture_level_out;
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if (err != AudioProcessing::kNoError) {
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return AudioProcessing::kUnspecifiedError;
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}
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}
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}
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return AudioProcessing::kNoError;
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}
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int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio,
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bool stream_has_echo) {
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if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
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return AudioProcessing::kStreamParameterNotSetError;
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}
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RTC_DCHECK(num_proc_channels_);
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RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
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audio->num_frames_per_band());
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RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
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stream_is_saturated_ = false;
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bool error_reported = false;
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for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
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int16_t split_band_data[AudioBuffer::kMaxNumBands]
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[AudioBuffer::kMaxSplitFrameLength];
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int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
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split_band_data[0], split_band_data[1], split_band_data[2]};
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audio->ExportSplitChannelData(ch, split_bands);
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// The call to stream_has_echo() is ok from a deadlock perspective
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// as the capture lock is allready held.
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int32_t new_capture_level = 0;
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uint8_t saturation_warning = 0;
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int err_analyze = WebRtcAgc_Analyze(
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mono_agcs_[ch]->state, split_bands, audio->num_bands(),
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audio->num_frames_per_band(), capture_levels_[ch], &new_capture_level,
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stream_has_echo, &saturation_warning, mono_agcs_[ch]->gains);
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capture_levels_[ch] = new_capture_level;
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error_reported = error_reported || err_analyze != AudioProcessing::kNoError;
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stream_is_saturated_ = stream_is_saturated_ || saturation_warning == 1;
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}
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// Choose the minimun gain for application
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size_t index_to_apply = 0;
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for (size_t ch = 1; ch < mono_agcs_.size(); ++ch) {
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if (mono_agcs_[index_to_apply]->gains[10] < mono_agcs_[ch]->gains[10]) {
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index_to_apply = ch;
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}
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}
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for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
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ApplyDigitalGain(mono_agcs_[index_to_apply]->gains, audio->num_bands(),
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audio->split_bands(ch));
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}
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RTC_DCHECK_LT(0ul, *num_proc_channels_);
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if (mode_ == kAdaptiveAnalog) {
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// Take the analog level to be the minimum accross all channels.
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analog_capture_level_ = capture_levels_[0];
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for (size_t ch = 1; ch < mono_agcs_.size(); ++ch) {
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analog_capture_level_ =
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std::min(analog_capture_level_, capture_levels_[ch]);
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}
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}
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if (error_reported) {
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return AudioProcessing::kUnspecifiedError;
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}
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was_analog_level_set_ = false;
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return AudioProcessing::kNoError;
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}
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// TODO(ajm): ensure this is called under kAdaptiveAnalog.
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int GainControlImpl::set_stream_analog_level(int level) {
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data_dumper_->DumpRaw("gain_control_set_stream_analog_level", 1, &level);
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was_analog_level_set_ = true;
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if (level < minimum_capture_level_ || level > maximum_capture_level_) {
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return AudioProcessing::kBadParameterError;
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}
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analog_capture_level_ = level;
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return AudioProcessing::kNoError;
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}
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int GainControlImpl::stream_analog_level() const {
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data_dumper_->DumpRaw("gain_control_stream_analog_level", 1,
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&analog_capture_level_);
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return analog_capture_level_;
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}
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int GainControlImpl::set_mode(Mode mode) {
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if (MapSetting(mode) == -1) {
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return AudioProcessing::kBadParameterError;
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}
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mode_ = mode;
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RTC_DCHECK(num_proc_channels_);
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RTC_DCHECK(sample_rate_hz_);
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Initialize(*num_proc_channels_, *sample_rate_hz_);
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return AudioProcessing::kNoError;
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}
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int GainControlImpl::set_analog_level_limits(int minimum, int maximum) {
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if (minimum < 0 || maximum > 65535 || maximum < minimum) {
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return AudioProcessing::kBadParameterError;
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}
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minimum_capture_level_ = minimum;
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maximum_capture_level_ = maximum;
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RTC_DCHECK(num_proc_channels_);
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RTC_DCHECK(sample_rate_hz_);
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Initialize(*num_proc_channels_, *sample_rate_hz_);
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return AudioProcessing::kNoError;
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}
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int GainControlImpl::set_target_level_dbfs(int level) {
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if (level > 31 || level < 0) {
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return AudioProcessing::kBadParameterError;
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}
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target_level_dbfs_ = level;
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return Configure();
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}
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int GainControlImpl::set_compression_gain_db(int gain) {
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if (gain < 0 || gain > 90) {
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RTC_LOG(LS_ERROR) << "set_compression_gain_db(" << gain << ") failed.";
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return AudioProcessing::kBadParameterError;
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}
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compression_gain_db_ = gain;
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return Configure();
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}
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int GainControlImpl::enable_limiter(bool enable) {
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limiter_enabled_ = enable;
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return Configure();
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}
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void GainControlImpl::Initialize(size_t num_proc_channels, int sample_rate_hz) {
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data_dumper_->InitiateNewSetOfRecordings();
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RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000 ||
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sample_rate_hz == 48000);
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num_proc_channels_ = num_proc_channels;
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sample_rate_hz_ = sample_rate_hz;
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mono_agcs_.resize(*num_proc_channels_);
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capture_levels_.resize(*num_proc_channels_);
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for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
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if (!mono_agcs_[ch]) {
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mono_agcs_[ch].reset(new MonoAgcState());
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}
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int error = WebRtcAgc_Init(mono_agcs_[ch]->state, minimum_capture_level_,
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maximum_capture_level_, MapSetting(mode_),
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*sample_rate_hz_);
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RTC_DCHECK_EQ(error, 0);
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capture_levels_[ch] = analog_capture_level_;
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}
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Configure();
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}
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int GainControlImpl::Configure() {
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WebRtcAgcConfig config;
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// TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
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// change the interface.
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// RTC_DCHECK_LE(target_level_dbfs_, 0);
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// config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
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config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
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config.compressionGaindB = static_cast<int16_t>(compression_gain_db_);
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config.limiterEnable = limiter_enabled_;
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int error = AudioProcessing::kNoError;
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for (size_t ch = 0; ch < mono_agcs_.size(); ++ch) {
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int error_ch = WebRtcAgc_set_config(mono_agcs_[ch]->state, config);
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if (error_ch != AudioProcessing::kNoError) {
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error = error_ch;
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}
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}
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return error;
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}
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} // namespace webrtc
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