Reland: Making WebRTC able to play and record audio to files for tests.

By specifying the define WEBRTC_DUMMY_FILE_DEVICES (which is similar to
WEBRTC_DUMMY_AUDIO_BUILD) an application will be able to tell WebRTC to
play out audio to a file and feed audio in from a file. We want to do
so we can better test WebRTC-using applications by recording what the
audio stack outputs and feeding known audio in for quality tests.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6403 4adac7df-926f-26a2-2b94-8c16560cd09d

7 files changed, 898 insertions, 2 deletions

diff --git a/modules/audio_device/Android.mk b/modules/audio_device/Android.mk
index affa5e1c..4b3b9124 100644
--- a/modules/audio_device/Android.mk
+++ b/modules/audio_device/Android.mk
@@ -25,7 +25,8 @@ LOCAL_SRC_FILES := \
     android/audio_device_android_opensles.cc \
     android/audio_device_utility_android.cc \
     dummy/audio_device_utility_dummy.cc \
-    dummy/audio_device_dummy.cc
+    dummy/audio_device_dummy.cc \
+    dummy/file_audio_device.cc
 
 # Flags passed to both C and C++ files.
 LOCAL_CFLAGS := \
diff --git a/modules/audio_device/audio_device.gypi b/modules/audio_device/audio_device.gypi
index 944f4222..a64856b5 100644
--- a/modules/audio_device/audio_device.gypi
+++ b/modules/audio_device/audio_device.gypi
@@ -20,7 +20,7 @@
         '.',
         '../interface',
         'include',
-        'dummy', # dummy audio device
+        'dummy',  # Contains dummy audio device implementations.
       ],
       'direct_dependent_settings': {
         'include_dirs': [
@@ -45,6 +45,8 @@
         'dummy/audio_device_dummy.h',
         'dummy/audio_device_utility_dummy.cc',
         'dummy/audio_device_utility_dummy.h',
+        'dummy/file_audio_device.cc',
+        'dummy/file_audio_device.h',
       ],
       'conditions': [
         ['OS=="linux"', {
@@ -77,6 +79,13 @@
             'WEBRTC_DUMMY_AUDIO_BUILD',
           ],
         }],
+        ['build_with_chromium==0', {
+          'sources': [
+            # Don't link these into Chrome since they contain static data.
+            'dummy/file_audio_device_factory.cc',
+            'dummy/file_audio_device_factory.h',
+          ],
+        }],
         ['include_internal_audio_device==1', {
           'sources': [
             'linux/alsasymboltable_linux.cc',
diff --git a/modules/audio_device/audio_device_impl.cc b/modules/audio_device/audio_device_impl.cc
index a2e5cba7..58411e3b 100644
--- a/modules/audio_device/audio_device_impl.cc
+++ b/modules/audio_device/audio_device_impl.cc
@@ -45,8 +45,14 @@
     #include "audio_device_utility_mac.h"
     #include "audio_device_mac.h"
 #endif
+
+#if defined(WEBRTC_DUMMY_FILE_DEVICES)
+#include "webrtc/modules/audio_device/dummy/file_audio_device_factory.h"
+#endif
+
 #include "webrtc/modules/audio_device/dummy/audio_device_dummy.h"
 #include "webrtc/modules/audio_device/dummy/audio_device_utility_dummy.h"
+#include "webrtc/modules/audio_device/dummy/file_audio_device.h"
 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
@@ -203,6 +209,14 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
     {
         ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
     }
+#elif defined(WEBRTC_DUMMY_FILE_DEVICES)
+    ptrAudioDevice = FileAudioDeviceFactory::CreateFileAudioDevice(Id());
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 "Will use file-playing dummy device.");
+    if (ptrAudioDevice != NULL)
+    {
+        ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
+    }
 #else
     const AudioLayer audioLayer(PlatformAudioLayer());
 
diff --git a/modules/audio_device/dummy/file_audio_device.cc b/modules/audio_device/dummy/file_audio_device.cc
new file mode 100644
index 00000000..e7771c66
--- /dev/null
+++ b/modules/audio_device/dummy/file_audio_device.cc
@@ -0,0 +1,586 @@
+/*
+ *  Copyright (c) 2014 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 <iostream>
+#include "webrtc/modules/audio_device/dummy/file_audio_device.h"
+#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/system_wrappers/interface/thread_wrapper.h"
+
+namespace webrtc {
+
+int kRecordingFixedSampleRate = 48000;
+int kRecordingNumChannels = 2;
+int kPlayoutFixedSampleRate = 48000;
+int kPlayoutNumChannels = 2;
+int kPlayoutBufferSize = kPlayoutFixedSampleRate / 100
+                         * kPlayoutNumChannels * 2;
+int kRecordingBufferSize = kRecordingFixedSampleRate / 100
+                           * kRecordingNumChannels * 2;
+
+FileAudioDevice::FileAudioDevice(const int32_t id,
+                                 const char* inputFilename,
+                                 const char* outputFile):
+    _ptrAudioBuffer(NULL),
+    _recordingBuffer(NULL),
+    _playoutBuffer(NULL),
+    _recordingFramesLeft(0),
+    _playoutFramesLeft(0),
+    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+    _recordingBufferSizeIn10MS(0),
+    _recordingFramesIn10MS(0),
+    _playoutFramesIn10MS(0),
+    _ptrThreadRec(NULL),
+    _ptrThreadPlay(NULL),
+    _recThreadID(0),
+    _playThreadID(0),
+    _playing(false),
+    _recording(false),
+    _lastCallPlayoutMillis(0),
+    _lastCallRecordMillis(0),
+    _outputFile(*FileWrapper::Create()),
+    _inputFile(*FileWrapper::Create()),
+    _outputFilename(outputFile),
+    _inputFilename(inputFilename),
+    _clock(Clock::GetRealTimeClock()) {
+}
+
+FileAudioDevice::~FileAudioDevice() {
+  _outputFile.Flush();
+  _outputFile.CloseFile();
+  delete &_outputFile;
+  _inputFile.Flush();
+  _inputFile.CloseFile();
+  delete &_inputFile;
+}
+
+int32_t FileAudioDevice::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::Init() { return 0; }
+
+int32_t FileAudioDevice::Terminate() { return 0; }
+
+bool FileAudioDevice::Initialized() const { return true; }
+
+int16_t FileAudioDevice::PlayoutDevices() {
+  return 1;
+}
+
+int16_t FileAudioDevice::RecordingDevices() {
+  return 1;
+}
+
+int32_t FileAudioDevice::PlayoutDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  const char* kName = "dummy_device";
+  const char* kGuid = "dummy_device_unique_id";
+  if (index < 1) {
+    memset(name, 0, kAdmMaxDeviceNameSize);
+    memset(guid, 0, kAdmMaxGuidSize);
+    memcpy(name, kName, strlen(kName));
+    memcpy(guid, kGuid, strlen(guid));
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::RecordingDeviceName(uint16_t index,
+                                              char name[kAdmMaxDeviceNameSize],
+                                              char guid[kAdmMaxGuidSize]) {
+  const char* kName = "dummy_device";
+  const char* kGuid = "dummy_device_unique_id";
+  if (index < 1) {
+    memset(name, 0, kAdmMaxDeviceNameSize);
+    memset(guid, 0, kAdmMaxGuidSize);
+    memcpy(name, kName, strlen(kName));
+    memcpy(guid, kGuid, strlen(guid));
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetPlayoutDevice(uint16_t index) {
+  if (index == 0) {
+    _playout_index = index;
+    return 0;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetRecordingDevice(uint16_t index) {
+  if (index == 0) {
+    _record_index = index;
+    return _record_index;
+  }
+  return -1;
+}
+
+int32_t FileAudioDevice::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t FileAudioDevice::PlayoutIsAvailable(bool& available) {
+  if (_playout_index == 0) {
+    available = true;
+    return _playout_index;
+  }
+  available = false;
+  return -1;
+}
+
+int32_t FileAudioDevice::InitPlayout() {
+  if (_ptrAudioBuffer)
+  {
+      // Update webrtc audio buffer with the selected parameters
+      _ptrAudioBuffer->SetPlayoutSampleRate(kPlayoutFixedSampleRate);
+      _ptrAudioBuffer->SetPlayoutChannels(kPlayoutNumChannels);
+  }
+  return 0;
+}
+
+bool FileAudioDevice::PlayoutIsInitialized() const {
+  return true;
+}
+
+int32_t FileAudioDevice::RecordingIsAvailable(bool& available) {
+  if (_record_index == 0) {
+    available = true;
+    return _record_index;
+  }
+  available = false;
+  return -1;
+}
+
+int32_t FileAudioDevice::InitRecording() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_recording) {
+    return -1;
+  }
+
+  _recordingFramesIn10MS = kRecordingFixedSampleRate/100;
+
+  if (_ptrAudioBuffer) {
+    _ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);
+    _ptrAudioBuffer->SetRecordingChannels(kRecordingNumChannels);
+  }
+  return 0;
+}
+
+bool FileAudioDevice::RecordingIsInitialized() const {
+  return true;
+}
+
+int32_t FileAudioDevice::StartPlayout() {
+  if (_playing)
+  {
+      return 0;
+  }
+
+  _playing = true;
+  _playoutFramesLeft = 0;
+
+  if (!_playoutBuffer)
+      _playoutBuffer = new int8_t[2 *
+                                  kPlayoutNumChannels *
+                                  kPlayoutFixedSampleRate/100];
+  if (!_playoutBuffer)
+  {
+    _playing = false;
+    return -1;
+  }
+
+  // PLAYOUT
+  const char* threadName = "webrtc_audio_module_play_thread";
+  _ptrThreadPlay =  ThreadWrapper::CreateThread(PlayThreadFunc,
+                                                this,
+                                                kRealtimePriority,
+                                                threadName);
+  if (_ptrThreadPlay == NULL)
+  {
+      _playing = false;
+      delete [] _playoutBuffer;
+      _playoutBuffer = NULL;
+      return -1;
+  }
+
+  if (_outputFile.OpenFile(_outputFilename.c_str(),
+                           false, false, false) == -1) {
+    printf("Failed to open playout file %s!", _outputFilename.c_str());
+    _playing = false;
+    delete [] _playoutBuffer;
+    _playoutBuffer = NULL;
+    return -1;
+  }
+
+  unsigned int threadID(0);
+  if (!_ptrThreadPlay->Start(threadID))
+  {
+      _playing = false;
+      delete _ptrThreadPlay;
+      _ptrThreadPlay = NULL;
+      delete [] _playoutBuffer;
+      _playoutBuffer = NULL;
+      return -1;
+  }
+  _playThreadID = threadID;
+
+  return 0;
+}
+
+int32_t FileAudioDevice::StopPlayout() {
+  {
+      CriticalSectionScoped lock(&_critSect);
+      _playing = false;
+  }
+
+  // stop playout thread first
+  if (_ptrThreadPlay && !_ptrThreadPlay->Stop())
+  {
+      return -1;
+  }
+  else {
+      delete _ptrThreadPlay;
+      _ptrThreadPlay = NULL;
+  }
+
+  CriticalSectionScoped lock(&_critSect);
+
+  _playoutFramesLeft = 0;
+  delete [] _playoutBuffer;
+  _playoutBuffer = NULL;
+  _outputFile.Flush();
+  _outputFile.CloseFile();
+   return 0;
+}
+
+bool FileAudioDevice::Playing() const {
+  return true;
+}
+
+int32_t FileAudioDevice::StartRecording() {
+  _recording = true;
+
+  // Make sure we only create the buffer once.
+  _recordingBufferSizeIn10MS = _recordingFramesIn10MS *
+                               kRecordingNumChannels *
+                               2;
+  if (!_recordingBuffer) {
+      _recordingBuffer = new int8_t[_recordingBufferSizeIn10MS];
+  }
+
+  if (_inputFile.OpenFile(_inputFilename.c_str(), true,
+                              true, false) == -1) {
+    printf("Failed to open audio input file %s!\n",
+           _inputFilename.c_str());
+    _recording = false;
+    delete[] _recordingBuffer;
+    _recordingBuffer = NULL;
+    return -1;
+  }
+
+  const char* threadName = "webrtc_audio_module_capture_thread";
+  _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc,
+                                              this,
+                                              kRealtimePriority,
+                                              threadName);
+  if (_ptrThreadRec == NULL)
+  {
+      _recording = false;
+      delete [] _recordingBuffer;
+      _recordingBuffer = NULL;
+      return -1;
+  }
+
+  unsigned int threadID(0);
+  if (!_ptrThreadRec->Start(threadID))
+  {
+      _recording = false;
+      delete _ptrThreadRec;
+      _ptrThreadRec = NULL;
+      delete [] _recordingBuffer;
+      _recordingBuffer = NULL;
+      return -1;
+  }
+  _recThreadID = threadID;
+
+  return 0;
+}
+
+
+int32_t FileAudioDevice::StopRecording() {
+  {
+    CriticalSectionScoped lock(&_critSect);
+    _recording = false;
+  }
+
+  if (_ptrThreadRec && !_ptrThreadRec->Stop())
+  {
+      return -1;
+  }
+  else {
+      delete _ptrThreadRec;
+      _ptrThreadRec = NULL;
+  }
+
+  CriticalSectionScoped lock(&_critSect);
+  _recordingFramesLeft = 0;
+  if (_recordingBuffer)
+  {
+      delete [] _recordingBuffer;
+      _recordingBuffer = NULL;
+  }
+  return 0;
+}
+
+bool FileAudioDevice::Recording() const {
+  return _recording;
+}
+
+int32_t FileAudioDevice::SetAGC(bool enable) { return -1; }
+
+bool FileAudioDevice::AGC() const { return false; }
+
+int32_t FileAudioDevice::SetWaveOutVolume(uint16_t volumeLeft,
+                                           uint16_t volumeRight) {
+  return -1;
+}
+
+int32_t FileAudioDevice::WaveOutVolume(uint16_t& volumeLeft,
+                                        uint16_t& volumeRight) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::InitSpeaker() { return -1; }
+
+bool FileAudioDevice::SpeakerIsInitialized() const { return false; }
+
+int32_t FileAudioDevice::InitMicrophone() { return 0; }
+
+bool FileAudioDevice::MicrophoneIsInitialized() const { return true; }
+
+int32_t FileAudioDevice::SpeakerVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetSpeakerVolume(uint32_t volume) { return -1; }
+
+int32_t FileAudioDevice::SpeakerVolume(uint32_t& volume) const { return -1; }
+
+int32_t FileAudioDevice::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MinSpeakerVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerVolumeStepSize(uint16_t& stepSize) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneVolumeIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetMicrophoneVolume(uint32_t volume) { return -1; }
+
+int32_t FileAudioDevice::MicrophoneVolume(uint32_t& volume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MinMicrophoneVolume(uint32_t& minVolume) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::MicrophoneVolumeStepSize(uint16_t& stepSize) const {
+  return -1;
+}
+
+int32_t FileAudioDevice::SpeakerMuteIsAvailable(bool& available) { return -1; }
+
+int32_t FileAudioDevice::SetSpeakerMute(bool enable) { return -1; }
+
+int32_t FileAudioDevice::SpeakerMute(bool& enabled) const { return -1; }
+
+int32_t FileAudioDevice::MicrophoneMuteIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetMicrophoneMute(bool enable) { return -1; }
+
+int32_t FileAudioDevice::MicrophoneMute(bool& enabled) const { return -1; }
+
+int32_t FileAudioDevice::MicrophoneBoostIsAvailable(bool& available) {
+  return -1;
+}
+
+int32_t FileAudioDevice::SetMicrophoneBoost(bool enable) { return -1; }
+
+int32_t FileAudioDevice::MicrophoneBoost(bool& enabled) const { return -1; }
+
+int32_t FileAudioDevice::StereoPlayoutIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+int32_t FileAudioDevice::SetStereoPlayout(bool enable) {
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoPlayout(bool& enabled) const {
+  enabled = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoRecordingIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::SetStereoRecording(bool enable) {
+  return 0;
+}
+
+int32_t FileAudioDevice::StereoRecording(bool& enabled) const {
+  enabled = true;
+  return 0;
+}
+
+int32_t FileAudioDevice::SetPlayoutBuffer(
+    const AudioDeviceModule::BufferType type,
+    uint16_t sizeMS) {
+  return 0;
+}
+
+int32_t FileAudioDevice::PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                        uint16_t& sizeMS) const {
+  type = _playBufType;
+  return 0;
+}
+
+int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const {
+  return 0;
+}
+
+int32_t FileAudioDevice::RecordingDelay(uint16_t& delayMS) const { return -1; }
+
+int32_t FileAudioDevice::CPULoad(uint16_t& load) const { return -1; }
+
+bool FileAudioDevice::PlayoutWarning() const { return false; }
+
+bool FileAudioDevice::PlayoutError() const { return false; }
+
+bool FileAudioDevice::RecordingWarning() const { return false; }
+
+bool FileAudioDevice::RecordingError() const { return false; }
+
+void FileAudioDevice::ClearPlayoutWarning() {}
+
+void FileAudioDevice::ClearPlayoutError() {}
+
+void FileAudioDevice::ClearRecordingWarning() {}
+
+void FileAudioDevice::ClearRecordingError() {}
+
+void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  CriticalSectionScoped lock(&_critSect);
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // Inform the AudioBuffer about default settings for this implementation.
+  // Set all values to zero here since the actual settings will be done by
+  // InitPlayout and InitRecording later.
+  _ptrAudioBuffer->SetRecordingSampleRate(0);
+  _ptrAudioBuffer->SetPlayoutSampleRate(0);
+  _ptrAudioBuffer->SetRecordingChannels(0);
+  _ptrAudioBuffer->SetPlayoutChannels(0);
+}
+
+bool FileAudioDevice::PlayThreadFunc(void* pThis)
+{
+    return (static_cast<FileAudioDevice*>(pThis)->PlayThreadProcess());
+}
+
+bool FileAudioDevice::RecThreadFunc(void* pThis)
+{
+    return (static_cast<FileAudioDevice*>(pThis)->RecThreadProcess());
+}
+
+bool FileAudioDevice::PlayThreadProcess()
+{
+    if(!_playing)
+        return false;
+
+    uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
+    _critSect.Enter();
+
+    if (_lastCallPlayoutMillis == 0 ||
+        currentTime - _lastCallPlayoutMillis >= 10)
+    {
+        _critSect.Leave();
+        _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
+        _critSect.Enter();
+
+        _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
+        assert(_playoutFramesLeft == _playoutFramesIn10MS);
+        if (_outputFile.Open()) {
+          _outputFile.Write(_playoutBuffer, kPlayoutBufferSize);
+          _outputFile.Flush();
+        }
+        _lastCallPlayoutMillis = currentTime;
+    }
+    _playoutFramesLeft = 0;
+    _critSect.Leave();
+    SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
+    return true;
+}
+
+bool FileAudioDevice::RecThreadProcess()
+{
+    if (!_recording)
+        return false;
+
+    uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
+    _critSect.Enter();
+
+    if (_lastCallRecordMillis == 0 ||
+        currentTime - _lastCallRecordMillis >= 10) {
+      if (_inputFile.Open()) {
+        if (_inputFile.Read(_recordingBuffer, kRecordingBufferSize) > 0) {
+          _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
+                                             _recordingFramesIn10MS);
+        } else {
+          _inputFile.Rewind();
+        }
+        _lastCallRecordMillis = currentTime;
+        _critSect.Leave();
+        _ptrAudioBuffer->DeliverRecordedData();
+        _critSect.Enter();
+      }
+    }
+
+    _critSect.Leave();
+    SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
+    return true;
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_device/dummy/file_audio_device.h b/modules/audio_device/dummy/file_audio_device.h
new file mode 100644
index 00000000..6f417eb2
--- /dev/null
+++ b/modules/audio_device/dummy/file_audio_device.h
@@ -0,0 +1,202 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
+#define WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
+
+#include <stdio.h>
+
+#include <string>
+
+#include "webrtc/modules/audio_device/audio_device_generic.h"
+#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/interface/file_wrapper.h"
+#include "webrtc/system_wrappers/interface/clock.h"
+
+namespace webrtc {
+class EventWrapper;
+class ThreadWrapper;
+
+// This is a fake audio device which plays audio from a file as its microphone
+// and plays out into a file.
+class FileAudioDevice : public AudioDeviceGeneric {
+ public:
+  // Constructs a file audio device with |id|. It will read audio from
+  // |inputFilename| and record output audio to |outputFilename|.
+  //
+  // The input file should be a readable 48k stereo raw file, and the output
+  // file should point to a writable location. The output format will also be
+  // 48k stereo raw audio.
+  FileAudioDevice(const int32_t id,
+                  const char* inputFilename,
+                  const char* outputFilename);
+  virtual ~FileAudioDevice();
+
+  // Retrieve the currently utilized audio layer
+  virtual int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const OVERRIDE;
+
+  // Main initializaton and termination
+  virtual int32_t Init() OVERRIDE;
+  virtual int32_t Terminate() OVERRIDE;
+  virtual bool Initialized() const OVERRIDE;
+
+  // Device enumeration
+  virtual int16_t PlayoutDevices() OVERRIDE;
+  virtual int16_t RecordingDevices() OVERRIDE;
+  virtual int32_t PlayoutDeviceName(uint16_t index,
+                                    char name[kAdmMaxDeviceNameSize],
+                                    char guid[kAdmMaxGuidSize]) OVERRIDE;
+  virtual int32_t RecordingDeviceName(uint16_t index,
+                                      char name[kAdmMaxDeviceNameSize],
+                                      char guid[kAdmMaxGuidSize]) OVERRIDE;
+
+  // Device selection
+  virtual int32_t SetPlayoutDevice(uint16_t index) OVERRIDE;
+  virtual int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
+  virtual int32_t SetRecordingDevice(uint16_t index) OVERRIDE;
+  virtual int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) OVERRIDE;
+
+  // Audio transport initialization
+  virtual int32_t PlayoutIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t InitPlayout() OVERRIDE;
+  virtual bool PlayoutIsInitialized() const OVERRIDE;
+  virtual int32_t RecordingIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t InitRecording() OVERRIDE;
+  virtual bool RecordingIsInitialized() const OVERRIDE;
+
+  // Audio transport control
+  virtual int32_t StartPlayout() OVERRIDE;
+  virtual int32_t StopPlayout() OVERRIDE;
+  virtual bool Playing() const OVERRIDE;
+  virtual int32_t StartRecording() OVERRIDE;
+  virtual int32_t StopRecording() OVERRIDE;
+  virtual bool Recording() const OVERRIDE;
+
+  // Microphone Automatic Gain Control (AGC)
+  virtual int32_t SetAGC(bool enable) OVERRIDE;
+  virtual bool AGC() const OVERRIDE;
+
+  // Volume control based on the Windows Wave API (Windows only)
+  virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
+                                   uint16_t volumeRight) OVERRIDE;
+  virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
+                                uint16_t& volumeRight) const OVERRIDE;
+
+  // Audio mixer initialization
+  virtual int32_t InitSpeaker() OVERRIDE;
+  virtual bool SpeakerIsInitialized() const OVERRIDE;
+  virtual int32_t InitMicrophone() OVERRIDE;
+  virtual bool MicrophoneIsInitialized() const OVERRIDE;
+
+  // Speaker volume controls
+  virtual int32_t SpeakerVolumeIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t SetSpeakerVolume(uint32_t volume) OVERRIDE;
+  virtual int32_t SpeakerVolume(uint32_t& volume) const OVERRIDE;
+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const OVERRIDE;
+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const OVERRIDE;
+  virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const OVERRIDE;
+
+  // Microphone volume controls
+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t SetMicrophoneVolume(uint32_t volume) OVERRIDE;
+  virtual int32_t MicrophoneVolume(uint32_t& volume) const OVERRIDE;
+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const OVERRIDE;
+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const OVERRIDE;
+  virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const OVERRIDE;
+
+  // Speaker mute control
+  virtual int32_t SpeakerMuteIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t SetSpeakerMute(bool enable) OVERRIDE;
+  virtual int32_t SpeakerMute(bool& enabled) const OVERRIDE;
+
+  // Microphone mute control
+  virtual int32_t MicrophoneMuteIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t SetMicrophoneMute(bool enable) OVERRIDE;
+  virtual int32_t MicrophoneMute(bool& enabled) const OVERRIDE;
+
+  // Microphone boost control
+  virtual int32_t MicrophoneBoostIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t SetMicrophoneBoost(bool enable) OVERRIDE;
+  virtual int32_t MicrophoneBoost(bool& enabled) const OVERRIDE;
+
+  // Stereo support
+  virtual int32_t StereoPlayoutIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t SetStereoPlayout(bool enable) OVERRIDE;
+  virtual int32_t StereoPlayout(bool& enabled) const OVERRIDE;
+  virtual int32_t StereoRecordingIsAvailable(bool& available) OVERRIDE;
+  virtual int32_t SetStereoRecording(bool enable) OVERRIDE;
+  virtual int32_t StereoRecording(bool& enabled) const OVERRIDE;
+
+  // Delay information and control
+  virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
+                                   uint16_t sizeMS) OVERRIDE;
+  virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                uint16_t& sizeMS) const OVERRIDE;
+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const OVERRIDE;
+  virtual int32_t RecordingDelay(uint16_t& delayMS) const OVERRIDE;
+
+  // CPU load
+  virtual int32_t CPULoad(uint16_t& load) const OVERRIDE;
+
+  virtual bool PlayoutWarning() const OVERRIDE;
+  virtual bool PlayoutError() const OVERRIDE;
+  virtual bool RecordingWarning() const OVERRIDE;
+  virtual bool RecordingError() const OVERRIDE;
+  virtual void ClearPlayoutWarning() OVERRIDE;
+  virtual void ClearPlayoutError() OVERRIDE;
+  virtual void ClearRecordingWarning() OVERRIDE;
+  virtual void ClearRecordingError() OVERRIDE;
+
+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) OVERRIDE;
+
+ private:
+  static bool RecThreadFunc(void*);
+  static bool PlayThreadFunc(void*);
+  bool RecThreadProcess();
+  bool PlayThreadProcess();
+
+  int32_t _playout_index;
+  int32_t _record_index;
+  AudioDeviceModule::BufferType _playBufType;
+  AudioDeviceBuffer* _ptrAudioBuffer;
+  int8_t* _recordingBuffer;  // In bytes.
+  int8_t* _playoutBuffer;  // In bytes.
+  uint32_t _recordingFramesLeft;
+  uint32_t _playoutFramesLeft;
+  CriticalSectionWrapper& _critSect;
+
+  uint32_t _recordingBufferSizeIn10MS;
+  uint32_t _recordingFramesIn10MS;
+  uint32_t _playoutFramesIn10MS;
+
+  ThreadWrapper* _ptrThreadRec;
+  ThreadWrapper* _ptrThreadPlay;
+  uint32_t _recThreadID;
+  uint32_t _playThreadID;
+
+  bool _playing;
+  bool _recording;
+  uint64_t _lastCallPlayoutMillis;
+  uint64_t _lastCallRecordMillis;
+
+  FileWrapper& _outputFile;
+  FileWrapper& _inputFile;
+  std::string _outputFilename;
+  std::string _inputFilename;
+
+  Clock* _clock;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_H
diff --git a/modules/audio_device/dummy/file_audio_device_factory.cc b/modules/audio_device/dummy/file_audio_device_factory.cc
new file mode 100644
index 00000000..db35bf11
--- /dev/null
+++ b/modules/audio_device/dummy/file_audio_device_factory.cc
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2014 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/modules/audio_device/dummy/file_audio_device_factory.h"
+
+#include <cstring>
+
+#include "webrtc/modules/audio_device/dummy/file_audio_device.h"
+
+namespace webrtc {
+
+char FileAudioDeviceFactory::_inputAudioFilename[MAX_FILENAME_LEN] = "";
+char FileAudioDeviceFactory::_outputAudioFilename[MAX_FILENAME_LEN] = "";
+
+FileAudioDevice* FileAudioDeviceFactory::CreateFileAudioDevice(
+    const int32_t id) {
+  // Bail out here if the files aren't set.
+  if (strlen(_inputAudioFilename) == 0 || strlen(_outputAudioFilename) == 0) {
+    printf("Was compiled with WEBRTC_DUMMY_AUDIO_PLAY_STATIC_FILE "
+           "but did not set input/output files to use. Bailing out.\n");
+    exit(1);
+  }
+  return new FileAudioDevice(id, _inputAudioFilename, _outputAudioFilename);
+}
+
+void FileAudioDeviceFactory::SetFilenamesToUse(
+    const char* inputAudioFilename, const char* outputAudioFilename) {
+  assert(strlen(inputAudioFilename) < MAX_FILENAME_LEN &&
+         strlen(outputAudioFilename) < MAX_FILENAME_LEN);
+
+  // Copy the strings since we don't know the lifetime of the input pointers.
+  strncpy(_inputAudioFilename, inputAudioFilename, MAX_FILENAME_LEN);
+  strncpy(_outputAudioFilename, outputAudioFilename, MAX_FILENAME_LEN);
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_device/dummy/file_audio_device_factory.h b/modules/audio_device/dummy/file_audio_device_factory.h
new file mode 100644
index 00000000..9975d7b9
--- /dev/null
+++ b/modules/audio_device/dummy/file_audio_device_factory.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H
+#define WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H
+
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+class FileAudioDevice;
+
+// This class is used by audio_device_impl.cc when WebRTC is compiled with
+// WEBRTC_DUMMY_FILE_DEVICES. The application must include this file and set the
+// filenames to use before the audio device module is initialized. This is
+// intended for test tools which use the audio device module.
+class FileAudioDeviceFactory {
+ public:
+  static FileAudioDevice* CreateFileAudioDevice(const int32_t id);
+
+  // The input file must be a readable 48k stereo raw file. The output
+  // file must be writable. The strings will be copied.
+  static void SetFilenamesToUse(const char* inputAudioFilename,
+                                const char* outputAudioFilename);
+
+ private:
+  static const uint32_t MAX_FILENAME_LEN = 256;
+  static char _inputAudioFilename[MAX_FILENAME_LEN];
+  static char _outputAudioFilename[MAX_FILENAME_LEN];
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_DEVICE_FILE_AUDIO_DEVICE_FACTORY_H