Arun Raghavan b5c48b97f6 Bump to WebRTC M131 release
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
2024-12-26 12:55:16 -05:00

288 lines
8.9 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/echo_control_mobile_impl.h"
#include <string.h>
#include <cstdint>
#include "api/audio/audio_processing.h"
#include "modules/audio_processing/aecm/echo_control_mobile.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
int16_t MapSetting(EchoControlMobileImpl::RoutingMode mode) {
switch (mode) {
case EchoControlMobileImpl::kQuietEarpieceOrHeadset:
return 0;
case EchoControlMobileImpl::kEarpiece:
return 1;
case EchoControlMobileImpl::kLoudEarpiece:
return 2;
case EchoControlMobileImpl::kSpeakerphone:
return 3;
case EchoControlMobileImpl::kLoudSpeakerphone:
return 4;
}
RTC_DCHECK_NOTREACHED();
return -1;
}
AudioProcessing::Error MapError(int err) {
switch (err) {
case AECM_UNSUPPORTED_FUNCTION_ERROR:
return AudioProcessing::kUnsupportedFunctionError;
case AECM_NULL_POINTER_ERROR:
return AudioProcessing::kNullPointerError;
case AECM_BAD_PARAMETER_ERROR:
return AudioProcessing::kBadParameterError;
case AECM_BAD_PARAMETER_WARNING:
return AudioProcessing::kBadStreamParameterWarning;
default:
// AECM_UNSPECIFIED_ERROR
// AECM_UNINITIALIZED_ERROR
return AudioProcessing::kUnspecifiedError;
}
}
} // namespace
struct EchoControlMobileImpl::StreamProperties {
StreamProperties() = delete;
StreamProperties(int sample_rate_hz,
size_t num_reverse_channels,
size_t num_output_channels)
: sample_rate_hz(sample_rate_hz),
num_reverse_channels(num_reverse_channels),
num_output_channels(num_output_channels) {}
int sample_rate_hz;
size_t num_reverse_channels;
size_t num_output_channels;
};
class EchoControlMobileImpl::Canceller {
public:
Canceller() {
state_ = WebRtcAecm_Create();
RTC_CHECK(state_);
}
~Canceller() {
RTC_DCHECK(state_);
WebRtcAecm_Free(state_);
}
Canceller(const Canceller&) = delete;
Canceller& operator=(const Canceller&) = delete;
void* state() {
RTC_DCHECK(state_);
return state_;
}
void Initialize(int sample_rate_hz) {
RTC_DCHECK(state_);
int error = WebRtcAecm_Init(state_, sample_rate_hz);
RTC_DCHECK_EQ(AudioProcessing::kNoError, error);
}
private:
void* state_;
};
EchoControlMobileImpl::EchoControlMobileImpl()
: routing_mode_(kSpeakerphone), comfort_noise_enabled_(false) {}
EchoControlMobileImpl::~EchoControlMobileImpl() {}
void EchoControlMobileImpl::ProcessRenderAudio(
rtc::ArrayView<const int16_t> packed_render_audio) {
RTC_DCHECK(stream_properties_);
size_t buffer_index = 0;
size_t num_frames_per_band =
packed_render_audio.size() / (stream_properties_->num_output_channels *
stream_properties_->num_reverse_channels);
for (auto& canceller : cancellers_) {
WebRtcAecm_BufferFarend(canceller->state(),
&packed_render_audio[buffer_index],
num_frames_per_band);
buffer_index += num_frames_per_band;
}
}
void EchoControlMobileImpl::PackRenderAudioBuffer(
const AudioBuffer* audio,
size_t num_output_channels,
size_t num_channels,
std::vector<int16_t>* packed_buffer) {
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
audio->num_frames_per_band());
RTC_DCHECK_EQ(num_channels, audio->num_channels());
// The ordering convention must be followed to pass to the correct AECM.
packed_buffer->clear();
int render_channel = 0;
for (size_t i = 0; i < num_output_channels; i++) {
for (size_t j = 0; j < audio->num_channels(); j++) {
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> data_to_buffer;
FloatS16ToS16(audio->split_bands_const(render_channel)[kBand0To8kHz],
audio->num_frames_per_band(), data_to_buffer.data());
// Buffer the samples in the render queue.
packed_buffer->insert(
packed_buffer->end(), data_to_buffer.data(),
data_to_buffer.data() + audio->num_frames_per_band());
render_channel = (render_channel + 1) % audio->num_channels();
}
}
}
size_t EchoControlMobileImpl::NumCancellersRequired(
size_t num_output_channels,
size_t num_reverse_channels) {
return num_output_channels * num_reverse_channels;
}
int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
int stream_delay_ms) {
RTC_DCHECK(stream_properties_);
RTC_DCHECK_GE(160, audio->num_frames_per_band());
RTC_DCHECK_EQ(audio->num_channels(), stream_properties_->num_output_channels);
RTC_DCHECK_GE(cancellers_.size(), stream_properties_->num_reverse_channels *
audio->num_channels());
int err = AudioProcessing::kNoError;
// The ordering convention must be followed to pass to the correct AECM.
size_t handle_index = 0;
for (size_t capture = 0; capture < audio->num_channels(); ++capture) {
// TODO(ajm): improve how this works, possibly inside AECM.
// This is kind of hacked up.
RTC_DCHECK_LT(capture, low_pass_reference_.size());
const int16_t* noisy =
reference_copied_ ? low_pass_reference_[capture].data() : nullptr;
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
audio->num_frames_per_band());
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> split_bands_data;
int16_t* split_bands = split_bands_data.data();
const int16_t* clean = split_bands_data.data();
if (audio->split_bands(capture)[kBand0To8kHz]) {
FloatS16ToS16(audio->split_bands(capture)[kBand0To8kHz],
audio->num_frames_per_band(), split_bands_data.data());
} else {
clean = nullptr;
split_bands = nullptr;
}
if (noisy == NULL) {
noisy = clean;
clean = NULL;
}
for (size_t render = 0; render < stream_properties_->num_reverse_channels;
++render) {
err = WebRtcAecm_Process(cancellers_[handle_index]->state(), noisy, clean,
split_bands, audio->num_frames_per_band(),
stream_delay_ms);
if (split_bands) {
S16ToFloatS16(split_bands, audio->num_frames_per_band(),
audio->split_bands(capture)[kBand0To8kHz]);
}
if (err != AudioProcessing::kNoError) {
return MapError(err);
}
++handle_index;
}
for (size_t band = 1u; band < audio->num_bands(); ++band) {
memset(audio->split_bands_f(capture)[band], 0,
audio->num_frames_per_band() *
sizeof(audio->split_bands_f(capture)[band][0]));
}
}
return AudioProcessing::kNoError;
}
int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
if (MapSetting(mode) == -1) {
return AudioProcessing::kBadParameterError;
}
routing_mode_ = mode;
return Configure();
}
EchoControlMobileImpl::RoutingMode EchoControlMobileImpl::routing_mode() const {
return routing_mode_;
}
int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
comfort_noise_enabled_ = enable;
return Configure();
}
bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
return comfort_noise_enabled_;
}
void EchoControlMobileImpl::Initialize(int sample_rate_hz,
size_t num_reverse_channels,
size_t num_output_channels) {
low_pass_reference_.resize(num_output_channels);
for (auto& reference : low_pass_reference_) {
reference.fill(0);
}
stream_properties_.reset(new StreamProperties(
sample_rate_hz, num_reverse_channels, num_output_channels));
// AECM only supports 16 kHz or lower sample rates.
RTC_DCHECK_LE(stream_properties_->sample_rate_hz,
AudioProcessing::kSampleRate16kHz);
cancellers_.resize(
NumCancellersRequired(stream_properties_->num_output_channels,
stream_properties_->num_reverse_channels));
for (auto& canceller : cancellers_) {
if (!canceller) {
canceller.reset(new Canceller());
}
canceller->Initialize(sample_rate_hz);
}
Configure();
}
int EchoControlMobileImpl::Configure() {
AecmConfig config;
config.cngMode = comfort_noise_enabled_;
config.echoMode = MapSetting(routing_mode_);
int error = AudioProcessing::kNoError;
for (auto& canceller : cancellers_) {
int handle_error = WebRtcAecm_set_config(canceller->state(), config);
if (handle_error != AudioProcessing::kNoError) {
error = handle_error;
}
}
return error;
}
} // namespace webrtc