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