147 lines
4.7 KiB
C++
147 lines
4.7 KiB
C++
/*
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* Copyright (c) 2013 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/test/fake_audio_device.h"
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#include <algorithm>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/platform_thread.h"
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#include "webrtc/modules/media_file/media_file_utility.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/system_wrappers/include/event_wrapper.h"
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#include "webrtc/system_wrappers/include/file_wrapper.h"
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namespace webrtc {
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namespace test {
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FakeAudioDevice::FakeAudioDevice(Clock* clock,
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const std::string& filename,
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float speed)
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: audio_callback_(NULL),
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capturing_(false),
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captured_audio_(),
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playout_buffer_(),
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speed_(speed),
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last_playout_ms_(-1),
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clock_(clock, speed),
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tick_(EventTimerWrapper::Create()),
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thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
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file_utility_(new ModuleFileUtility(0)),
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input_stream_(FileWrapper::Create()) {
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memset(captured_audio_, 0, sizeof(captured_audio_));
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memset(playout_buffer_, 0, sizeof(playout_buffer_));
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// Open audio input file as read-only and looping.
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EXPECT_TRUE(input_stream_->OpenFile(filename.c_str(), true)) << filename;
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}
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FakeAudioDevice::~FakeAudioDevice() {
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Stop();
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thread_.Stop();
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}
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int32_t FakeAudioDevice::Init() {
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rtc::CritScope cs(&lock_);
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if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
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return -1;
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if (!tick_->StartTimer(true, 10 / speed_))
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return -1;
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thread_.Start();
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thread_.SetPriority(rtc::kHighPriority);
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return 0;
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}
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int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
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rtc::CritScope cs(&lock_);
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audio_callback_ = callback;
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return 0;
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}
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bool FakeAudioDevice::Playing() const {
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rtc::CritScope cs(&lock_);
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return capturing_;
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}
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int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
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*delay_ms = 0;
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return 0;
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}
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bool FakeAudioDevice::Recording() const {
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rtc::CritScope cs(&lock_);
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return capturing_;
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}
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bool FakeAudioDevice::Run(void* obj) {
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static_cast<FakeAudioDevice*>(obj)->CaptureAudio();
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return true;
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}
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void FakeAudioDevice::CaptureAudio() {
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{
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rtc::CritScope cs(&lock_);
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if (capturing_) {
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int bytes_read = file_utility_->ReadPCMData(
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*input_stream_.get(), captured_audio_, kBufferSizeBytes);
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if (bytes_read <= 0)
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return;
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// 2 bytes per sample.
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size_t num_samples = static_cast<size_t>(bytes_read / 2);
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uint32_t new_mic_level;
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EXPECT_EQ(0,
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audio_callback_->RecordedDataIsAvailable(captured_audio_,
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num_samples,
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2,
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1,
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kFrequencyHz,
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0,
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0,
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0,
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false,
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new_mic_level));
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size_t samples_needed = kFrequencyHz / 100;
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int64_t now_ms = clock_.TimeInMilliseconds();
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uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
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if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
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samples_needed = std::min(
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static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
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kBufferSizeBytes / 2);
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}
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size_t samples_out = 0;
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int64_t elapsed_time_ms = -1;
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int64_t ntp_time_ms = -1;
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EXPECT_EQ(0,
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audio_callback_->NeedMorePlayData(samples_needed,
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2,
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1,
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kFrequencyHz,
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playout_buffer_,
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samples_out,
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&elapsed_time_ms,
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&ntp_time_ms));
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}
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}
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tick_->Wait(WEBRTC_EVENT_INFINITE);
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}
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void FakeAudioDevice::Start() {
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rtc::CritScope cs(&lock_);
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capturing_ = true;
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}
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void FakeAudioDevice::Stop() {
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rtc::CritScope cs(&lock_);
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capturing_ = false;
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}
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} // namespace test
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} // namespace webrtc
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