rhubarb-lip-sync/rhubarb/lib/webrtc-8d2248ff/webrtc/api/test/fakeaudiocapturemodule.cc

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2016-06-21 20:13:05 +00:00
/*
* Copyright 2012 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/api/test/fakeaudiocapturemodule.h"
#include "webrtc/base/common.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/thread.h"
#include "webrtc/base/timeutils.h"
// Audio sample value that is high enough that it doesn't occur naturally when
// frames are being faked. E.g. NetEq will not generate this large sample value
// unless it has received an audio frame containing a sample of this value.
// Even simpler buffers would likely just contain audio sample values of 0.
static const int kHighSampleValue = 10000;
// Same value as src/modules/audio_device/main/source/audio_device_config.h in
// https://code.google.com/p/webrtc/
static const int kAdmMaxIdleTimeProcess = 1000;
// Constants here are derived by running VoE using a real ADM.
// The constants correspond to 10ms of mono audio at 44kHz.
static const int kTimePerFrameMs = 10;
static const uint8_t kNumberOfChannels = 1;
static const int kSamplesPerSecond = 44000;
static const int kTotalDelayMs = 0;
static const int kClockDriftMs = 0;
static const uint32_t kMaxVolume = 14392;
enum {
MSG_START_PROCESS,
MSG_RUN_PROCESS,
};
FakeAudioCaptureModule::FakeAudioCaptureModule()
: last_process_time_ms_(0),
audio_callback_(nullptr),
recording_(false),
playing_(false),
play_is_initialized_(false),
rec_is_initialized_(false),
current_mic_level_(kMaxVolume),
started_(false),
next_frame_time_(0),
frames_received_(0) {
}
FakeAudioCaptureModule::~FakeAudioCaptureModule() {
if (process_thread_) {
process_thread_->Stop();
}
}
rtc::scoped_refptr<FakeAudioCaptureModule> FakeAudioCaptureModule::Create() {
rtc::scoped_refptr<FakeAudioCaptureModule> capture_module(
new rtc::RefCountedObject<FakeAudioCaptureModule>());
if (!capture_module->Initialize()) {
return nullptr;
}
return capture_module;
}
int FakeAudioCaptureModule::frames_received() const {
rtc::CritScope cs(&crit_);
return frames_received_;
}
int64_t FakeAudioCaptureModule::TimeUntilNextProcess() {
const int64_t current_time = rtc::TimeMillis();
if (current_time < last_process_time_ms_) {
// TODO: wraparound could be handled more gracefully.
return 0;
}
const int64_t elapsed_time = current_time - last_process_time_ms_;
if (kAdmMaxIdleTimeProcess < elapsed_time) {
return 0;
}
return kAdmMaxIdleTimeProcess - elapsed_time;
}
void FakeAudioCaptureModule::Process() {
last_process_time_ms_ = rtc::TimeMillis();
}
int32_t FakeAudioCaptureModule::ActiveAudioLayer(
AudioLayer* /*audio_layer*/) const {
ASSERT(false);
return 0;
}
webrtc::AudioDeviceModule::ErrorCode FakeAudioCaptureModule::LastError() const {
ASSERT(false);
return webrtc::AudioDeviceModule::kAdmErrNone;
}
int32_t FakeAudioCaptureModule::RegisterEventObserver(
webrtc::AudioDeviceObserver* /*event_callback*/) {
// Only used to report warnings and errors. This fake implementation won't
// generate any so discard this callback.
return 0;
}
int32_t FakeAudioCaptureModule::RegisterAudioCallback(
webrtc::AudioTransport* audio_callback) {
rtc::CritScope cs(&crit_callback_);
audio_callback_ = audio_callback;
return 0;
}
int32_t FakeAudioCaptureModule::Init() {
// Initialize is called by the factory method. Safe to ignore this Init call.
return 0;
}
int32_t FakeAudioCaptureModule::Terminate() {
// Clean up in the destructor. No action here, just success.
return 0;
}
bool FakeAudioCaptureModule::Initialized() const {
ASSERT(false);
return 0;
}
int16_t FakeAudioCaptureModule::PlayoutDevices() {
ASSERT(false);
return 0;
}
int16_t FakeAudioCaptureModule::RecordingDevices() {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::PlayoutDeviceName(
uint16_t /*index*/,
char /*name*/[webrtc::kAdmMaxDeviceNameSize],
char /*guid*/[webrtc::kAdmMaxGuidSize]) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::RecordingDeviceName(
uint16_t /*index*/,
char /*name*/[webrtc::kAdmMaxDeviceNameSize],
char /*guid*/[webrtc::kAdmMaxGuidSize]) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetPlayoutDevice(uint16_t /*index*/) {
// No playout device, just playing from file. Return success.
return 0;
}
int32_t FakeAudioCaptureModule::SetPlayoutDevice(WindowsDeviceType /*device*/) {
if (play_is_initialized_) {
return -1;
}
return 0;
}
int32_t FakeAudioCaptureModule::SetRecordingDevice(uint16_t /*index*/) {
// No recording device, just dropping audio. Return success.
return 0;
}
int32_t FakeAudioCaptureModule::SetRecordingDevice(
WindowsDeviceType /*device*/) {
if (rec_is_initialized_) {
return -1;
}
return 0;
}
int32_t FakeAudioCaptureModule::PlayoutIsAvailable(bool* /*available*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::InitPlayout() {
play_is_initialized_ = true;
return 0;
}
bool FakeAudioCaptureModule::PlayoutIsInitialized() const {
return play_is_initialized_;
}
int32_t FakeAudioCaptureModule::RecordingIsAvailable(bool* /*available*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::InitRecording() {
rec_is_initialized_ = true;
return 0;
}
bool FakeAudioCaptureModule::RecordingIsInitialized() const {
return rec_is_initialized_;
}
int32_t FakeAudioCaptureModule::StartPlayout() {
if (!play_is_initialized_) {
return -1;
}
{
rtc::CritScope cs(&crit_);
playing_ = true;
}
bool start = true;
UpdateProcessing(start);
return 0;
}
int32_t FakeAudioCaptureModule::StopPlayout() {
bool start = false;
{
rtc::CritScope cs(&crit_);
playing_ = false;
start = ShouldStartProcessing();
}
UpdateProcessing(start);
return 0;
}
bool FakeAudioCaptureModule::Playing() const {
rtc::CritScope cs(&crit_);
return playing_;
}
int32_t FakeAudioCaptureModule::StartRecording() {
if (!rec_is_initialized_) {
return -1;
}
{
rtc::CritScope cs(&crit_);
recording_ = true;
}
bool start = true;
UpdateProcessing(start);
return 0;
}
int32_t FakeAudioCaptureModule::StopRecording() {
bool start = false;
{
rtc::CritScope cs(&crit_);
recording_ = false;
start = ShouldStartProcessing();
}
UpdateProcessing(start);
return 0;
}
bool FakeAudioCaptureModule::Recording() const {
rtc::CritScope cs(&crit_);
return recording_;
}
int32_t FakeAudioCaptureModule::SetAGC(bool /*enable*/) {
// No AGC but not needed since audio is pregenerated. Return success.
return 0;
}
bool FakeAudioCaptureModule::AGC() const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetWaveOutVolume(uint16_t /*volume_left*/,
uint16_t /*volume_right*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::WaveOutVolume(
uint16_t* /*volume_left*/,
uint16_t* /*volume_right*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::InitSpeaker() {
// No speaker, just playing from file. Return success.
return 0;
}
bool FakeAudioCaptureModule::SpeakerIsInitialized() const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::InitMicrophone() {
// No microphone, just playing from file. Return success.
return 0;
}
bool FakeAudioCaptureModule::MicrophoneIsInitialized() const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerVolumeIsAvailable(bool* /*available*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetSpeakerVolume(uint32_t /*volume*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerVolume(uint32_t* /*volume*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MaxSpeakerVolume(
uint32_t* /*max_volume*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MinSpeakerVolume(
uint32_t* /*min_volume*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerVolumeStepSize(
uint16_t* /*step_size*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneVolumeIsAvailable(
bool* /*available*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetMicrophoneVolume(uint32_t volume) {
rtc::CritScope cs(&crit_);
current_mic_level_ = volume;
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneVolume(uint32_t* volume) const {
rtc::CritScope cs(&crit_);
*volume = current_mic_level_;
return 0;
}
int32_t FakeAudioCaptureModule::MaxMicrophoneVolume(
uint32_t* max_volume) const {
*max_volume = kMaxVolume;
return 0;
}
int32_t FakeAudioCaptureModule::MinMicrophoneVolume(
uint32_t* /*min_volume*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneVolumeStepSize(
uint16_t* /*step_size*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerMuteIsAvailable(bool* /*available*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetSpeakerMute(bool /*enable*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SpeakerMute(bool* /*enabled*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneMuteIsAvailable(bool* /*available*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetMicrophoneMute(bool /*enable*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneMute(bool* /*enabled*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneBoostIsAvailable(
bool* /*available*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetMicrophoneBoost(bool /*enable*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::MicrophoneBoost(bool* /*enabled*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::StereoPlayoutIsAvailable(
bool* available) const {
// No recording device, just dropping audio. Stereo can be dropped just
// as easily as mono.
*available = true;
return 0;
}
int32_t FakeAudioCaptureModule::SetStereoPlayout(bool /*enable*/) {
// No recording device, just dropping audio. Stereo can be dropped just
// as easily as mono.
return 0;
}
int32_t FakeAudioCaptureModule::StereoPlayout(bool* /*enabled*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::StereoRecordingIsAvailable(
bool* available) const {
// Keep thing simple. No stereo recording.
*available = false;
return 0;
}
int32_t FakeAudioCaptureModule::SetStereoRecording(bool enable) {
if (!enable) {
return 0;
}
return -1;
}
int32_t FakeAudioCaptureModule::StereoRecording(bool* /*enabled*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetRecordingChannel(
const ChannelType channel) {
if (channel != AudioDeviceModule::kChannelBoth) {
// There is no right or left in mono. I.e. kChannelBoth should be used for
// mono.
ASSERT(false);
return -1;
}
return 0;
}
int32_t FakeAudioCaptureModule::RecordingChannel(ChannelType* channel) const {
// Stereo recording not supported. However, WebRTC ADM returns kChannelBoth
// in that case. Do the same here.
*channel = AudioDeviceModule::kChannelBoth;
return 0;
}
int32_t FakeAudioCaptureModule::SetPlayoutBuffer(const BufferType /*type*/,
uint16_t /*size_ms*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::PlayoutBuffer(BufferType* /*type*/,
uint16_t* /*size_ms*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::PlayoutDelay(uint16_t* delay_ms) const {
// No delay since audio frames are dropped.
*delay_ms = 0;
return 0;
}
int32_t FakeAudioCaptureModule::RecordingDelay(uint16_t* /*delay_ms*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::CPULoad(uint16_t* /*load*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::StartRawOutputFileRecording(
const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::StopRawOutputFileRecording() {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::StartRawInputFileRecording(
const char /*pcm_file_name_utf8*/[webrtc::kAdmMaxFileNameSize]) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::StopRawInputFileRecording() {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetRecordingSampleRate(
const uint32_t /*samples_per_sec*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::RecordingSampleRate(
uint32_t* /*samples_per_sec*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetPlayoutSampleRate(
const uint32_t /*samples_per_sec*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::PlayoutSampleRate(
uint32_t* /*samples_per_sec*/) const {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::ResetAudioDevice() {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::SetLoudspeakerStatus(bool /*enable*/) {
ASSERT(false);
return 0;
}
int32_t FakeAudioCaptureModule::GetLoudspeakerStatus(bool* /*enabled*/) const {
ASSERT(false);
return 0;
}
void FakeAudioCaptureModule::OnMessage(rtc::Message* msg) {
switch (msg->message_id) {
case MSG_START_PROCESS:
StartProcessP();
break;
case MSG_RUN_PROCESS:
ProcessFrameP();
break;
default:
// All existing messages should be caught. Getting here should never
// happen.
ASSERT(false);
}
}
bool FakeAudioCaptureModule::Initialize() {
// Set the send buffer samples high enough that it would not occur on the
// remote side unless a packet containing a sample of that magnitude has been
// sent to it. Note that the audio processing pipeline will likely distort the
// original signal.
SetSendBuffer(kHighSampleValue);
last_process_time_ms_ = rtc::TimeMillis();
return true;
}
void FakeAudioCaptureModule::SetSendBuffer(int value) {
Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_);
const size_t buffer_size_in_samples =
sizeof(send_buffer_) / kNumberBytesPerSample;
for (size_t i = 0; i < buffer_size_in_samples; ++i) {
buffer_ptr[i] = value;
}
}
void FakeAudioCaptureModule::ResetRecBuffer() {
memset(rec_buffer_, 0, sizeof(rec_buffer_));
}
bool FakeAudioCaptureModule::CheckRecBuffer(int value) {
const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_);
const size_t buffer_size_in_samples =
sizeof(rec_buffer_) / kNumberBytesPerSample;
for (size_t i = 0; i < buffer_size_in_samples; ++i) {
if (buffer_ptr[i] >= value) return true;
}
return false;
}
bool FakeAudioCaptureModule::ShouldStartProcessing() {
return recording_ || playing_;
}
void FakeAudioCaptureModule::UpdateProcessing(bool start) {
if (start) {
if (!process_thread_) {
process_thread_.reset(new rtc::Thread());
process_thread_->Start();
}
process_thread_->Post(RTC_FROM_HERE, this, MSG_START_PROCESS);
} else {
if (process_thread_) {
process_thread_->Stop();
process_thread_.reset(nullptr);
}
started_ = false;
}
}
void FakeAudioCaptureModule::StartProcessP() {
ASSERT(process_thread_->IsCurrent());
if (started_) {
// Already started.
return;
}
ProcessFrameP();
}
void FakeAudioCaptureModule::ProcessFrameP() {
ASSERT(process_thread_->IsCurrent());
if (!started_) {
next_frame_time_ = rtc::TimeMillis();
started_ = true;
}
{
rtc::CritScope cs(&crit_);
// Receive and send frames every kTimePerFrameMs.
if (playing_) {
ReceiveFrameP();
}
if (recording_) {
SendFrameP();
}
}
next_frame_time_ += kTimePerFrameMs;
const int64_t current_time = rtc::TimeMillis();
const int64_t wait_time =
(next_frame_time_ > current_time) ? next_frame_time_ - current_time : 0;
process_thread_->PostDelayed(RTC_FROM_HERE, wait_time, this, MSG_RUN_PROCESS);
}
void FakeAudioCaptureModule::ReceiveFrameP() {
ASSERT(process_thread_->IsCurrent());
{
rtc::CritScope cs(&crit_callback_);
if (!audio_callback_) {
return;
}
ResetRecBuffer();
size_t nSamplesOut = 0;
int64_t elapsed_time_ms = 0;
int64_t ntp_time_ms = 0;
if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
kNumberOfChannels, kSamplesPerSecond,
rec_buffer_, nSamplesOut,
&elapsed_time_ms, &ntp_time_ms) != 0) {
ASSERT(false);
}
ASSERT(nSamplesOut == kNumberSamples);
}
// The SetBuffer() function ensures that after decoding, the audio buffer
// should contain samples of similar magnitude (there is likely to be some
// distortion due to the audio pipeline). If one sample is detected to
// have the same or greater magnitude somewhere in the frame, an actual frame
// has been received from the remote side (i.e. faked frames are not being
// pulled).
if (CheckRecBuffer(kHighSampleValue)) {
rtc::CritScope cs(&crit_);
++frames_received_;
}
}
void FakeAudioCaptureModule::SendFrameP() {
ASSERT(process_thread_->IsCurrent());
rtc::CritScope cs(&crit_callback_);
if (!audio_callback_) {
return;
}
bool key_pressed = false;
uint32_t current_mic_level = 0;
MicrophoneVolume(&current_mic_level);
if (audio_callback_->RecordedDataIsAvailable(send_buffer_, kNumberSamples,
kNumberBytesPerSample,
kNumberOfChannels,
kSamplesPerSecond, kTotalDelayMs,
kClockDriftMs, current_mic_level,
key_pressed,
current_mic_level) != 0) {
ASSERT(false);
}
SetMicrophoneVolume(current_mic_level);
}