Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 551 insertions, 0 deletions

diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc
new file mode 100644
index 0000000000..9c7bf069d8
--- /dev/null
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -0,0 +1,551 @@
+/*
+ *  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/include/sleep.h"
+#include "webrtc/system_wrappers/include/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* outputFilename):
+    _ptrAudioBuffer(NULL),
+    _recordingBuffer(NULL),
+    _playoutBuffer(NULL),
+    _recordingFramesLeft(0),
+    _playoutFramesLeft(0),
+    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+    _recordingBufferSizeIn10MS(0),
+    _recordingFramesIn10MS(0),
+    _playoutFramesIn10MS(0),
+    _playing(false),
+    _recording(false),
+    _lastCallPlayoutMillis(0),
+    _lastCallRecordMillis(0),
+    _outputFile(*FileWrapper::Create()),
+    _inputFile(*FileWrapper::Create()),
+    _outputFilename(outputFilename),
+    _inputFilename(inputFilename),
+    _clock(Clock::GetRealTimeClock()) {
+}
+
+FileAudioDevice::~FileAudioDevice() {
+  if (_outputFile.Open()) {
+      _outputFile.Flush();
+      _outputFile.CloseFile();
+  }
+  delete &_outputFile;
+  if (_inputFile.Open()) {
+      _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 = static_cast<size_t>(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;
+  }
+
+  _playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);
+  _playing = true;
+  _playoutFramesLeft = 0;
+
+  if (!_playoutBuffer) {
+      _playoutBuffer = new int8_t[2 *
+                                  kPlayoutNumChannels *
+                                  kPlayoutFixedSampleRate/100];
+  }
+  if (!_playoutBuffer) {
+    _playing = false;
+    return -1;
+  }
+
+  // PLAYOUT
+  if (!_outputFilename.empty() && _outputFile.OpenFile(
+        _outputFilename.c_str(), false, false, false) == -1) {
+    printf("Failed to open playout file %s!\n", _outputFilename.c_str());
+    _playing = false;
+    delete [] _playoutBuffer;
+    _playoutBuffer = NULL;
+    return -1;
+  }
+
+  const char* threadName = "webrtc_audio_module_play_thread";
+  _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
+                                               threadName);
+  if (!_ptrThreadPlay->Start()) {
+      _ptrThreadPlay.reset();
+      _playing = false;
+      delete [] _playoutBuffer;
+      _playoutBuffer = NULL;
+      return -1;
+  }
+  _ptrThreadPlay->SetPriority(kRealtimePriority);
+  return 0;
+}
+
+int32_t FileAudioDevice::StopPlayout() {
+  {
+      CriticalSectionScoped lock(&_critSect);
+      _playing = false;
+  }
+
+  // stop playout thread first
+  if (_ptrThreadPlay) {
+      _ptrThreadPlay->Stop();
+      _ptrThreadPlay.reset();
+  }
+
+  CriticalSectionScoped lock(&_critSect);
+
+  _playoutFramesLeft = 0;
+  delete [] _playoutBuffer;
+  _playoutBuffer = NULL;
+  if (_outputFile.Open()) {
+      _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 (!_inputFilename.empty() && _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, threadName);
+
+  if (!_ptrThreadRec->Start()) {
+      _ptrThreadRec.reset();
+      _recording = false;
+      delete [] _recordingBuffer;
+      _recordingBuffer = NULL;
+      return -1;
+  }
+  _ptrThreadRec->SetPriority(kRealtimePriority);
+
+  return 0;
+}
+
+
+int32_t FileAudioDevice::StopRecording() {
+  {
+    CriticalSectionScoped lock(&_critSect);
+    _recording = false;
+  }
+
+  if (_ptrThreadRec) {
+      _ptrThreadRec->Stop();
+      _ptrThreadRec.reset();
+  }
+
+  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