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