webrtc_m130/webrtc/modules/audio_device/audio_device_buffer.cc

/*
 *  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(size_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(size_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
// ----------------------------------------------------------------------------

size_t AudioDeviceBuffer::RecordingChannels() const {
  return _recChannels;
}

// ----------------------------------------------------------------------------
//  PlayoutChannels
// ----------------------------------------------------------------------------

size_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) ? 0 : -1;
}

// ----------------------------------------------------------------------------
//  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) ? 0 : -1;
}

// ----------------------------------------------------------------------------
//  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.is_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;
  size_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.is_open()) {
    // write to binary file in mono or stereo (interleaved)
    _playFile.Write(&_playBuffer[0], _playSize);
  }

  return static_cast<int32_t>(_playSamples);
}

}  // namespace webrtc