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, 584 insertions, 0 deletions

diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
new file mode 100644
index 0000000000..e7b487d687
--- /dev/null
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -0,0 +1,584 @@
+/*
+ *  Copyright (c) 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/modules/audio_device/audio_device_buffer.h"
+
+#include <assert.h>
+#include <string.h>
+
+#include "webrtc/base/format_macros.h"
+#include "webrtc/modules/audio_device/audio_device_config.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+
+static const int kHighDelayThresholdMs = 300;
+static const int kLogHighDelayIntervalFrames = 500;  // 5 seconds.
+
+// ----------------------------------------------------------------------------
+//  ctor
+// ----------------------------------------------------------------------------
+
+AudioDeviceBuffer::AudioDeviceBuffer() :
+    _id(-1),
+    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+    _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
+    _ptrCbAudioTransport(NULL),
+    _recSampleRate(0),
+    _playSampleRate(0),
+    _recChannels(0),
+    _playChannels(0),
+    _recChannel(AudioDeviceModule::kChannelBoth),
+    _recBytesPerSample(0),
+    _playBytesPerSample(0),
+    _recSamples(0),
+    _recSize(0),
+    _playSamples(0),
+    _playSize(0),
+    _recFile(*FileWrapper::Create()),
+    _playFile(*FileWrapper::Create()),
+    _currentMicLevel(0),
+    _newMicLevel(0),
+    _typingStatus(false),
+    _playDelayMS(0),
+    _recDelayMS(0),
+    _clockDrift(0),
+    // Set to the interval in order to log on the first occurrence.
+    high_delay_counter_(kLogHighDelayIntervalFrames) {
+    // valid ID will be set later by SetId, use -1 for now
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created", __FUNCTION__);
+    memset(_recBuffer, 0, kMaxBufferSizeBytes);
+    memset(_playBuffer, 0, kMaxBufferSizeBytes);
+}
+
+// ----------------------------------------------------------------------------
+//  dtor
+// ----------------------------------------------------------------------------
+
+AudioDeviceBuffer::~AudioDeviceBuffer()
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
+    {
+        CriticalSectionScoped lock(&_critSect);
+
+        _recFile.Flush();
+        _recFile.CloseFile();
+        delete &_recFile;
+
+        _playFile.Flush();
+        _playFile.CloseFile();
+        delete &_playFile;
+    }
+
+    delete &_critSect;
+    delete &_critSectCb;
+}
+
+// ----------------------------------------------------------------------------
+//  SetId
+// ----------------------------------------------------------------------------
+
+void AudioDeviceBuffer::SetId(uint32_t id)
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "AudioDeviceBuffer::SetId(id=%d)", id);
+    _id = id;
+}
+
+// ----------------------------------------------------------------------------
+//  RegisterAudioCallback
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::RegisterAudioCallback(AudioTransport* audioCallback)
+{
+    CriticalSectionScoped lock(&_critSectCb);
+    _ptrCbAudioTransport = audioCallback;
+
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  InitPlayout
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::InitPlayout()
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  InitRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::InitRecording()
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordingSampleRate
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz)
+{
+    CriticalSectionScoped lock(&_critSect);
+    _recSampleRate = fsHz;
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetPlayoutSampleRate
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz)
+{
+    CriticalSectionScoped lock(&_critSect);
+    _playSampleRate = fsHz;
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  RecordingSampleRate
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::RecordingSampleRate() const
+{
+    return _recSampleRate;
+}
+
+// ----------------------------------------------------------------------------
+//  PlayoutSampleRate
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::PlayoutSampleRate() const
+{
+    return _playSampleRate;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordingChannels
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::SetRecordingChannels(uint8_t channels)
+{
+    CriticalSectionScoped lock(&_critSect);
+    _recChannels = channels;
+    _recBytesPerSample = 2*channels;  // 16 bits per sample in mono, 32 bits in stereo
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetPlayoutChannels
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::SetPlayoutChannels(uint8_t channels)
+{
+    CriticalSectionScoped lock(&_critSect);
+    _playChannels = channels;
+    // 16 bits per sample in mono, 32 bits in stereo
+    _playBytesPerSample = 2*channels;
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordingChannel
+//
+//  Select which channel to use while recording.
+//  This API requires that stereo is enabled.
+//
+//  Note that, the nChannel parameter in RecordedDataIsAvailable will be
+//  set to 2 even for kChannelLeft and kChannelRight. However, nBytesPerSample
+//  will be 2 instead of 4 four these cases.
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::SetRecordingChannel(const AudioDeviceModule::ChannelType channel)
+{
+    CriticalSectionScoped lock(&_critSect);
+
+    if (_recChannels == 1)
+    {
+        return -1;
+    }
+
+    if (channel == AudioDeviceModule::kChannelBoth)
+    {
+        // two bytes per channel
+        _recBytesPerSample = 4;
+    }
+    else
+    {
+        // only utilize one out of two possible channels (left or right)
+        _recBytesPerSample = 2;
+    }
+    _recChannel = channel;
+
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  RecordingChannel
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& channel) const
+{
+    channel = _recChannel;
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  RecordingChannels
+// ----------------------------------------------------------------------------
+
+uint8_t AudioDeviceBuffer::RecordingChannels() const
+{
+    return _recChannels;
+}
+
+// ----------------------------------------------------------------------------
+//  PlayoutChannels
+// ----------------------------------------------------------------------------
+
+uint8_t AudioDeviceBuffer::PlayoutChannels() const
+{
+    return _playChannels;
+}
+
+// ----------------------------------------------------------------------------
+//  SetCurrentMicLevel
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::SetCurrentMicLevel(uint32_t level)
+{
+    _currentMicLevel = level;
+    return 0;
+}
+
+int32_t AudioDeviceBuffer::SetTypingStatus(bool typingStatus)
+{
+    _typingStatus = typingStatus;
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  NewMicLevel
+// ----------------------------------------------------------------------------
+
+uint32_t AudioDeviceBuffer::NewMicLevel() const
+{
+    return _newMicLevel;
+}
+
+// ----------------------------------------------------------------------------
+//  SetVQEData
+// ----------------------------------------------------------------------------
+
+void AudioDeviceBuffer::SetVQEData(int playDelayMs, int recDelayMs,
+                                   int clockDrift) {
+  if (high_delay_counter_ < kLogHighDelayIntervalFrames) {
+    ++high_delay_counter_;
+  } else {
+    if (playDelayMs + recDelayMs > kHighDelayThresholdMs) {
+      high_delay_counter_ = 0;
+      LOG(LS_WARNING) << "High audio device delay reported (render="
+                      << playDelayMs << " ms, capture=" << recDelayMs << " ms)";
+    }
+  }
+
+  _playDelayMS = playDelayMs;
+  _recDelayMS = recDelayMs;
+  _clockDrift = clockDrift;
+}
+
+// ----------------------------------------------------------------------------
+//  StartInputFileRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::StartInputFileRecording(
+    const char fileName[kAdmMaxFileNameSize])
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+    CriticalSectionScoped lock(&_critSect);
+
+    _recFile.Flush();
+    _recFile.CloseFile();
+
+    return (_recFile.OpenFile(fileName, false, false, false));
+}
+
+// ----------------------------------------------------------------------------
+//  StopInputFileRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::StopInputFileRecording()
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+    CriticalSectionScoped lock(&_critSect);
+
+    _recFile.Flush();
+    _recFile.CloseFile();
+
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  StartOutputFileRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::StartOutputFileRecording(
+    const char fileName[kAdmMaxFileNameSize])
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+    CriticalSectionScoped lock(&_critSect);
+
+    _playFile.Flush();
+    _playFile.CloseFile();
+
+    return (_playFile.OpenFile(fileName, false, false, false));
+}
+
+// ----------------------------------------------------------------------------
+//  StopOutputFileRecording
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::StopOutputFileRecording()
+{
+    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+
+    CriticalSectionScoped lock(&_critSect);
+
+    _playFile.Flush();
+    _playFile.CloseFile();
+
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  SetRecordedBuffer
+//
+//  Store recorded audio buffer in local memory ready for the actual
+//  "delivery" using a callback.
+//
+//  This method can also parse out left or right channel from a stereo
+//  input signal, i.e., emulate mono.
+//
+//  Examples:
+//
+//  16-bit,48kHz mono,  10ms => nSamples=480 => _recSize=2*480=960 bytes
+//  16-bit,48kHz stereo,10ms => nSamples=480 => _recSize=4*480=1920 bytes
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer,
+                                             size_t nSamples)
+{
+    CriticalSectionScoped lock(&_critSect);
+
+    if (_recBytesPerSample == 0)
+    {
+        assert(false);
+        return -1;
+    }
+
+    _recSamples = nSamples;
+    _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
+    if (_recSize > kMaxBufferSizeBytes)
+    {
+        assert(false);
+        return -1;
+    }
+
+    if (_recChannel == AudioDeviceModule::kChannelBoth)
+    {
+        // (default) copy the complete input buffer to the local buffer
+        memcpy(&_recBuffer[0], audioBuffer, _recSize);
+    }
+    else
+    {
+        int16_t* ptr16In = (int16_t*)audioBuffer;
+        int16_t* ptr16Out = (int16_t*)&_recBuffer[0];
+
+        if (AudioDeviceModule::kChannelRight == _recChannel)
+        {
+            ptr16In++;
+        }
+
+        // exctract left or right channel from input buffer to the local buffer
+        for (size_t i = 0; i < _recSamples; i++)
+        {
+            *ptr16Out = *ptr16In;
+            ptr16Out++;
+            ptr16In++;
+            ptr16In++;
+        }
+    }
+
+    if (_recFile.Open())
+    {
+        // write to binary file in mono or stereo (interleaved)
+        _recFile.Write(&_recBuffer[0], _recSize);
+    }
+
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  DeliverRecordedData
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::DeliverRecordedData()
+{
+    CriticalSectionScoped lock(&_critSectCb);
+
+    // Ensure that user has initialized all essential members
+    if ((_recSampleRate == 0)     ||
+        (_recSamples == 0)        ||
+        (_recBytesPerSample == 0) ||
+        (_recChannels == 0))
+    {
+        assert(false);
+        return -1;
+    }
+
+    if (_ptrCbAudioTransport == NULL)
+    {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to deliver recorded data (AudioTransport does not exist)");
+        return 0;
+    }
+
+    int32_t res(0);
+    uint32_t newMicLevel(0);
+    uint32_t totalDelayMS = _playDelayMS +_recDelayMS;
+
+    res = _ptrCbAudioTransport->RecordedDataIsAvailable(&_recBuffer[0],
+                                                        _recSamples,
+                                                        _recBytesPerSample,
+                                                        _recChannels,
+                                                        _recSampleRate,
+                                                        totalDelayMS,
+                                                        _clockDrift,
+                                                        _currentMicLevel,
+                                                        _typingStatus,
+                                                        newMicLevel);
+    if (res != -1)
+    {
+        _newMicLevel = newMicLevel;
+    }
+
+    return 0;
+}
+
+// ----------------------------------------------------------------------------
+//  RequestPlayoutData
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::RequestPlayoutData(size_t nSamples)
+{
+    uint32_t playSampleRate = 0;
+    size_t playBytesPerSample = 0;
+    uint8_t playChannels = 0;
+    {
+        CriticalSectionScoped lock(&_critSect);
+
+        // Store copies under lock and use copies hereafter to avoid race with
+        // setter methods.
+        playSampleRate = _playSampleRate;
+        playBytesPerSample = _playBytesPerSample;
+        playChannels = _playChannels;
+
+        // Ensure that user has initialized all essential members
+        if ((playBytesPerSample == 0) ||
+            (playChannels == 0)       ||
+            (playSampleRate == 0))
+        {
+            assert(false);
+            return -1;
+        }
+
+        _playSamples = nSamples;
+        _playSize = playBytesPerSample * nSamples;  // {2,4}*nSamples
+        if (_playSize > kMaxBufferSizeBytes)
+        {
+            assert(false);
+            return -1;
+        }
+
+        if (nSamples != _playSamples)
+        {
+            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of samples to be played out (%d)", nSamples);
+            return -1;
+        }
+    }
+
+    size_t nSamplesOut(0);
+
+    CriticalSectionScoped lock(&_critSectCb);
+
+    if (_ptrCbAudioTransport == NULL)
+    {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to feed data to playout (AudioTransport does not exist)");
+        return 0;
+    }
+
+    if (_ptrCbAudioTransport)
+    {
+        uint32_t res(0);
+        int64_t elapsed_time_ms = -1;
+        int64_t ntp_time_ms = -1;
+        res = _ptrCbAudioTransport->NeedMorePlayData(_playSamples,
+                                                     playBytesPerSample,
+                                                     playChannels,
+                                                     playSampleRate,
+                                                     &_playBuffer[0],
+                                                     nSamplesOut,
+                                                     &elapsed_time_ms,
+                                                     &ntp_time_ms);
+        if (res != 0)
+        {
+            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "NeedMorePlayData() failed");
+        }
+    }
+
+    return static_cast<int32_t>(nSamplesOut);
+}
+
+// ----------------------------------------------------------------------------
+//  GetPlayoutData
+// ----------------------------------------------------------------------------
+
+int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer)
+{
+    CriticalSectionScoped lock(&_critSect);
+
+    if (_playSize > kMaxBufferSizeBytes)
+    {
+       WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
+                    "_playSize %" PRIuS " exceeds kMaxBufferSizeBytes in "
+                    "AudioDeviceBuffer::GetPlayoutData", _playSize);
+       assert(false);
+       return -1;
+    }
+
+    memcpy(audioBuffer, &_playBuffer[0], _playSize);
+
+    if (_playFile.Open())
+    {
+        // write to binary file in mono or stereo (interleaved)
+        _playFile.Write(&_playBuffer[0], _playSize);
+    }
+
+    return static_cast<int32_t>(_playSamples);
+}
+
+}  // namespace webrtc