
* Better param names * Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases. * Use arraysize() * Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers * reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead * Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition * Fix indenting * Use uint32_t for timestamps (matching how it's already a uint32_t in most places) * Spelling * RTC_CHECK_EQ(expected, actual) * Rewrap * Use .empty() * Be more pedantic about matching int/int32_t/ * Remove pointless consts on input parameters to functions * Add missing sanity checks All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first. BUG=none TEST=none Review URL: https://codereview.webrtc.org/1534193008 Cr-Commit-Position: refs/heads/master@{#11191}
408 lines
12 KiB
C++
408 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 "webrtc/modules/audio_processing/echo_control_mobile_impl.h"
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#include <assert.h>
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#include <string.h>
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#include "webrtc/modules/audio_processing/aecm/echo_control_mobile.h"
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#include "webrtc/modules/audio_processing/audio_buffer.h"
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#include "webrtc/system_wrappers/include/logging.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(EchoControlMobile::RoutingMode mode) {
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switch (mode) {
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case EchoControlMobile::kQuietEarpieceOrHeadset:
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return 0;
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case EchoControlMobile::kEarpiece:
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return 1;
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case EchoControlMobile::kLoudEarpiece:
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return 2;
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case EchoControlMobile::kSpeakerphone:
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return 3;
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case EchoControlMobile::kLoudSpeakerphone:
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return 4;
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}
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assert(false);
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return -1;
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}
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AudioProcessing::Error MapError(int err) {
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switch (err) {
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case AECM_UNSUPPORTED_FUNCTION_ERROR:
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return AudioProcessing::kUnsupportedFunctionError;
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case AECM_NULL_POINTER_ERROR:
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return AudioProcessing::kNullPointerError;
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case AECM_BAD_PARAMETER_ERROR:
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return AudioProcessing::kBadParameterError;
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case AECM_BAD_PARAMETER_WARNING:
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return AudioProcessing::kBadStreamParameterWarning;
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default:
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// AECM_UNSPECIFIED_ERROR
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// AECM_UNINITIALIZED_ERROR
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return AudioProcessing::kUnspecifiedError;
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}
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}
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// Maximum length that a frame of samples can have.
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static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
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// Maximum number of frames to buffer in the render queue.
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// TODO(peah): Decrease this once we properly handle hugely unbalanced
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// reverse and forward call numbers.
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static const size_t kMaxNumFramesToBuffer = 100;
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} // namespace
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size_t EchoControlMobile::echo_path_size_bytes() {
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return WebRtcAecm_echo_path_size_bytes();
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}
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EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessing* apm,
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rtc::CriticalSection* crit_render,
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rtc::CriticalSection* crit_capture)
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: ProcessingComponent(),
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apm_(apm),
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crit_render_(crit_render),
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crit_capture_(crit_capture),
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routing_mode_(kSpeakerphone),
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comfort_noise_enabled_(true),
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external_echo_path_(NULL),
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render_queue_element_max_size_(0) {
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RTC_DCHECK(apm);
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RTC_DCHECK(crit_render);
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RTC_DCHECK(crit_capture);
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}
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EchoControlMobileImpl::~EchoControlMobileImpl() {
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if (external_echo_path_ != NULL) {
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delete [] external_echo_path_;
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external_echo_path_ = NULL;
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}
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}
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int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) {
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rtc::CritScope cs_render(crit_render_);
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if (!is_component_enabled()) {
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return AudioProcessing::kNoError;
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}
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assert(audio->num_frames_per_band() <= 160);
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assert(audio->num_channels() == apm_->num_reverse_channels());
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int err = AudioProcessing::kNoError;
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// The ordering convention must be followed to pass to the correct AECM.
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size_t handle_index = 0;
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render_queue_buffer_.clear();
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for (int i = 0; i < apm_->num_output_channels(); i++) {
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for (int j = 0; j < audio->num_channels(); j++) {
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Handle* my_handle = static_cast<Handle*>(handle(handle_index));
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err = WebRtcAecm_GetBufferFarendError(
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my_handle, audio->split_bands_const(j)[kBand0To8kHz],
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audio->num_frames_per_band());
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if (err != AudioProcessing::kNoError)
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return MapError(err); // TODO(ajm): warning possible?);
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// Buffer the samples in the render queue.
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render_queue_buffer_.insert(render_queue_buffer_.end(),
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audio->split_bands_const(j)[kBand0To8kHz],
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(audio->split_bands_const(j)[kBand0To8kHz] +
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audio->num_frames_per_band()));
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handle_index++;
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}
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}
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// Insert the samples into the queue.
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if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
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// The data queue is full and needs to be emptied.
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ReadQueuedRenderData();
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// Retry the insert (should always work).
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RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
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}
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return AudioProcessing::kNoError;
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}
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// Read chunks of data that were received and queued on the render side from
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// a queue. All the data chunks are buffered into the farend signal of the AEC.
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void EchoControlMobileImpl::ReadQueuedRenderData() {
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rtc::CritScope cs_capture(crit_capture_);
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if (!is_component_enabled()) {
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return;
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}
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while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
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size_t handle_index = 0;
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size_t buffer_index = 0;
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const size_t num_frames_per_band =
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capture_queue_buffer_.size() /
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(apm_->num_output_channels() * apm_->num_reverse_channels());
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for (int i = 0; i < apm_->num_output_channels(); i++) {
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for (int j = 0; j < apm_->num_reverse_channels(); j++) {
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Handle* my_handle = static_cast<Handle*>(handle(handle_index));
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WebRtcAecm_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
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num_frames_per_band);
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buffer_index += num_frames_per_band;
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handle_index++;
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}
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}
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}
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}
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int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
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rtc::CritScope cs_capture(crit_capture_);
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if (!is_component_enabled()) {
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return AudioProcessing::kNoError;
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}
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if (!apm_->was_stream_delay_set()) {
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return AudioProcessing::kStreamParameterNotSetError;
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}
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assert(audio->num_frames_per_band() <= 160);
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assert(audio->num_channels() == apm_->num_output_channels());
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int err = AudioProcessing::kNoError;
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// The ordering convention must be followed to pass to the correct AECM.
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size_t handle_index = 0;
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for (int i = 0; i < audio->num_channels(); i++) {
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// TODO(ajm): improve how this works, possibly inside AECM.
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// This is kind of hacked up.
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const int16_t* noisy = audio->low_pass_reference(i);
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const int16_t* clean = audio->split_bands_const(i)[kBand0To8kHz];
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if (noisy == NULL) {
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noisy = clean;
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clean = NULL;
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}
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for (int j = 0; j < apm_->num_reverse_channels(); j++) {
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Handle* my_handle = static_cast<Handle*>(handle(handle_index));
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err = WebRtcAecm_Process(
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my_handle,
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noisy,
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clean,
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audio->split_bands(i)[kBand0To8kHz],
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audio->num_frames_per_band(),
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apm_->stream_delay_ms());
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if (err != AudioProcessing::kNoError)
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return MapError(err);
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handle_index++;
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}
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}
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return AudioProcessing::kNoError;
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}
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int EchoControlMobileImpl::Enable(bool enable) {
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// Ensure AEC and AECM are not both enabled.
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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// The is_enabled call is safe from a deadlock perspective
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// as both locks are allready held in the correct order.
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if (enable && apm_->echo_cancellation()->is_enabled()) {
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return AudioProcessing::kBadParameterError;
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}
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return EnableComponent(enable);
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}
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bool EchoControlMobileImpl::is_enabled() const {
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rtc::CritScope cs(crit_capture_);
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return is_component_enabled();
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}
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int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
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if (MapSetting(mode) == -1) {
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return AudioProcessing::kBadParameterError;
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}
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{
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rtc::CritScope cs(crit_capture_);
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routing_mode_ = mode;
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}
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return Configure();
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}
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EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode()
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const {
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rtc::CritScope cs(crit_capture_);
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return routing_mode_;
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}
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int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
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{
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rtc::CritScope cs(crit_capture_);
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comfort_noise_enabled_ = enable;
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}
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return Configure();
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}
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bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
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rtc::CritScope cs(crit_capture_);
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return comfort_noise_enabled_;
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}
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int EchoControlMobileImpl::SetEchoPath(const void* echo_path,
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size_t size_bytes) {
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{
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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if (echo_path == NULL) {
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return AudioProcessing::kNullPointerError;
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}
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if (size_bytes != echo_path_size_bytes()) {
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// Size mismatch
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return AudioProcessing::kBadParameterError;
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}
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if (external_echo_path_ == NULL) {
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external_echo_path_ = new unsigned char[size_bytes];
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}
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memcpy(external_echo_path_, echo_path, size_bytes);
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}
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return Initialize();
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}
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int EchoControlMobileImpl::GetEchoPath(void* echo_path,
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size_t size_bytes) const {
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rtc::CritScope cs(crit_capture_);
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if (echo_path == NULL) {
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return AudioProcessing::kNullPointerError;
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}
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if (size_bytes != echo_path_size_bytes()) {
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// Size mismatch
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return AudioProcessing::kBadParameterError;
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}
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if (!is_component_enabled()) {
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return AudioProcessing::kNotEnabledError;
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}
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// Get the echo path from the first channel
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Handle* my_handle = static_cast<Handle*>(handle(0));
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int32_t err = WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes);
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if (err != 0)
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return MapError(err);
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return AudioProcessing::kNoError;
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}
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int EchoControlMobileImpl::Initialize() {
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{
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rtc::CritScope cs_capture(crit_capture_);
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if (!is_component_enabled()) {
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return AudioProcessing::kNoError;
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}
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}
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if (apm_->proc_sample_rate_hz() > AudioProcessing::kSampleRate16kHz) {
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LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates";
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return AudioProcessing::kBadSampleRateError;
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}
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int err = ProcessingComponent::Initialize();
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if (err != AudioProcessing::kNoError) {
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return err;
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}
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AllocateRenderQueue();
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return AudioProcessing::kNoError;
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}
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void EchoControlMobileImpl::AllocateRenderQueue() {
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const size_t new_render_queue_element_max_size = std::max<size_t>(
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static_cast<size_t>(1),
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kMaxAllowedValuesOfSamplesPerFrame * num_handles_required());
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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// Reallocate the queue if the queue item size is too small to fit the
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// data to put in the queue.
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if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
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render_queue_element_max_size_ = new_render_queue_element_max_size;
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std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
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render_signal_queue_.reset(
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new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
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kMaxNumFramesToBuffer, template_queue_element,
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RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));
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render_queue_buffer_.resize(render_queue_element_max_size_);
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capture_queue_buffer_.resize(render_queue_element_max_size_);
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} else {
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render_signal_queue_->Clear();
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}
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}
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void* EchoControlMobileImpl::CreateHandle() const {
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return WebRtcAecm_Create();
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}
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void EchoControlMobileImpl::DestroyHandle(void* handle) const {
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// This method is only called in a non-concurrent manner during APM
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// destruction.
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WebRtcAecm_Free(static_cast<Handle*>(handle));
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}
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int EchoControlMobileImpl::InitializeHandle(void* handle) const {
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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assert(handle != NULL);
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Handle* my_handle = static_cast<Handle*>(handle);
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if (WebRtcAecm_Init(my_handle, apm_->proc_sample_rate_hz()) != 0) {
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return GetHandleError(my_handle);
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}
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if (external_echo_path_ != NULL) {
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if (WebRtcAecm_InitEchoPath(my_handle,
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external_echo_path_,
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echo_path_size_bytes()) != 0) {
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return GetHandleError(my_handle);
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}
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}
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return AudioProcessing::kNoError;
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}
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int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
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rtc::CritScope cs_render(crit_render_);
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rtc::CritScope cs_capture(crit_capture_);
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AecmConfig config;
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config.cngMode = comfort_noise_enabled_;
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config.echoMode = MapSetting(routing_mode_);
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return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
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}
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int EchoControlMobileImpl::num_handles_required() const {
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// Not locked as it only relies on APM public API which is threadsafe.
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return apm_->num_output_channels() * apm_->num_reverse_channels();
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}
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int EchoControlMobileImpl::GetHandleError(void* handle) const {
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// Not locked as it does not rely on anything in the state.
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assert(handle != NULL);
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return AudioProcessing::kUnspecifiedError;
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}
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} // namespace webrtc
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