116 lines
3.8 KiB
C++
116 lines
3.8 KiB
C++
/*
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* Copyright (c) 2019 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 "test/pc/e2e/echo/echo_emulation.h"
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#include <limits>
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#include <utility>
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namespace webrtc {
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namespace webrtc_pc_e2e {
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namespace {
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constexpr int kSingleBufferDurationMs = 10;
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} // namespace
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EchoEmulatingCapturer::EchoEmulatingCapturer(
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std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
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PeerConnectionE2EQualityTestFixture::EchoEmulationConfig config)
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: delegate_(std::move(capturer)),
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config_(config),
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renderer_queue_(2 * config_.echo_delay.ms() / kSingleBufferDurationMs),
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queue_input_(TestAudioDeviceModule::SamplesPerFrame(
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delegate_->SamplingFrequency()) *
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delegate_->NumChannels()),
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queue_output_(TestAudioDeviceModule::SamplesPerFrame(
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delegate_->SamplingFrequency()) *
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delegate_->NumChannels()) {
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renderer_thread_.Detach();
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capturer_thread_.Detach();
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}
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void EchoEmulatingCapturer::OnAudioRendered(
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rtc::ArrayView<const int16_t> data) {
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RTC_DCHECK_RUN_ON(&renderer_thread_);
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if (!recording_started_) {
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// Because rendering can start before capturing in the beginning we can have
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// a set of empty audio data frames. So we will skip them and will start
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// fill the queue only after 1st non-empty audio data frame will arrive.
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bool is_empty = true;
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for (auto d : data) {
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if (d != 0) {
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is_empty = false;
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break;
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}
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}
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if (is_empty) {
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return;
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}
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recording_started_ = true;
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}
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queue_input_.assign(data.begin(), data.end());
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if (!renderer_queue_.Insert(&queue_input_)) {
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RTC_LOG(WARNING) << "Echo queue is full";
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}
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}
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bool EchoEmulatingCapturer::Capture(rtc::BufferT<int16_t>* buffer) {
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RTC_DCHECK_RUN_ON(&capturer_thread_);
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bool result = delegate_->Capture(buffer);
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// Now we have to reduce input signal to avoid saturation when mixing in the
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// fake echo.
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for (size_t i = 0; i < buffer->size(); ++i) {
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(*buffer)[i] /= 2;
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}
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// When we accumulated enough delay in the echo buffer we will pop from
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// that buffer on each ::Capture(...) call. If the buffer become empty it
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// will mean some bug, so we will crash during removing item from the queue.
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if (!delay_accumulated_) {
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delay_accumulated_ =
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renderer_queue_.SizeAtLeast() >=
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static_cast<size_t>(config_.echo_delay.ms() / kSingleBufferDurationMs);
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}
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if (delay_accumulated_) {
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RTC_CHECK(renderer_queue_.Remove(&queue_output_));
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for (size_t i = 0; i < buffer->size() && i < queue_output_.size(); ++i) {
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int32_t res = (*buffer)[i] + queue_output_[i];
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if (res < std::numeric_limits<int16_t>::min()) {
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res = std::numeric_limits<int16_t>::min();
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}
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if (res > std::numeric_limits<int16_t>::max()) {
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res = std::numeric_limits<int16_t>::max();
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}
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(*buffer)[i] = static_cast<int16_t>(res);
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}
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}
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return result;
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}
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EchoEmulatingRenderer::EchoEmulatingRenderer(
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std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
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EchoEmulatingCapturer* echo_emulating_capturer)
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: delegate_(std::move(renderer)),
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echo_emulating_capturer_(echo_emulating_capturer) {
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RTC_DCHECK(echo_emulating_capturer_);
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}
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bool EchoEmulatingRenderer::Render(rtc::ArrayView<const int16_t> data) {
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if (data.size() > 0) {
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echo_emulating_capturer_->OnAudioRendered(data);
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}
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return delegate_->Render(data);
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}
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} // namespace webrtc_pc_e2e
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} // namespace webrtc
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