Move FilePlayer and FileRecorder to Voice Engine

webrtc/modules/utility/source/file_recorder.cc

Index: webrtc/modules/utility/source/file_recorder.cc
diff --git a/webrtc/modules/utility/source/file_recorder.cc b/webrtc/modules/utility/source/file_recorder.cc
deleted file mode 100644
index 34bd3945faee0c27ad95d6ba7f279ed141bebef1..0000000000000000000000000000000000000000
--- a/webrtc/modules/utility/source/file_recorder.cc
+++ /dev/null
@@ -1,262 +0,0 @@
-/*
- *  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/utility/include/file_recorder.h"
-
-#include <list>
-
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/common_audio/resampler/include/resampler.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/media_file/media_file.h"
-#include "webrtc/modules/media_file/media_file_defines.h"
-#include "webrtc/modules/utility/source/coder.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/logging.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-namespace {
-
-// The largest decoded frame size in samples (60ms with 32kHz sample rate).
-enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 };
-enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 };
-enum { kMaxAudioBufferQueueLength = 100 };
-
-class CriticalSectionWrapper;
-
-class FileRecorderImpl : public FileRecorder {
- public:
-  FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat);
-  ~FileRecorderImpl() override;
-
-  // FileRecorder functions.
-  int32_t RegisterModuleFileCallback(FileCallback* callback) override;
-  FileFormats RecordingFileFormat() const override;
-  int32_t StartRecordingAudioFile(const char* fileName,
-                                  const CodecInst& codecInst,
-                                  uint32_t notificationTimeMs) override;
-  int32_t StartRecordingAudioFile(OutStream* destStream,
-                                  const CodecInst& codecInst,
-                                  uint32_t notificationTimeMs) override;
-  int32_t StopRecording() override;
-  bool IsRecording() const override;
-  int32_t codec_info(CodecInst* codecInst) const override;
-  int32_t RecordAudioToFile(const AudioFrame& frame) override;
-
- private:
-  int32_t WriteEncodedAudioData(const int8_t* audioBuffer, size_t bufferLength);
-
-  int32_t SetUpAudioEncoder();
-
-  uint32_t _instanceID;
-  FileFormats _fileFormat;
-  MediaFile* _moduleFile;
-
-  CodecInst codec_info_;
-  int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
-  AudioCoder _audioEncoder;
-  Resampler _audioResampler;
-};
-
-FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat)
-    : _instanceID(instanceID),
-      _fileFormat(fileFormat),
-      _moduleFile(MediaFile::CreateMediaFile(_instanceID)),
-      codec_info_(),
-      _audioBuffer(),
-      _audioEncoder(instanceID),
-      _audioResampler() {}
-
-FileRecorderImpl::~FileRecorderImpl() {
-  MediaFile::DestroyMediaFile(_moduleFile);
-}
-
-FileFormats FileRecorderImpl::RecordingFileFormat() const {
-  return _fileFormat;
-}
-
-int32_t FileRecorderImpl::RegisterModuleFileCallback(FileCallback* callback) {
-  if (_moduleFile == NULL) {
-    return -1;
-  }
-  return _moduleFile->SetModuleFileCallback(callback);
-}
-
-int32_t FileRecorderImpl::StartRecordingAudioFile(const char* fileName,
-                                                  const CodecInst& codecInst,
-                                                  uint32_t notificationTimeMs) {
-  if (_moduleFile == NULL) {
-    return -1;
-  }
-  codec_info_ = codecInst;
-  int32_t retVal = 0;
-  retVal = _moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
-                                                codecInst, notificationTimeMs);
-
-  if (retVal == 0) {
-    retVal = SetUpAudioEncoder();
-  }
-  if (retVal != 0) {
-    LOG(LS_WARNING) << "Failed to initialize file " << fileName
-                    << " for recording.";
-
-    if (IsRecording()) {
-      StopRecording();
-    }
-  }
-  return retVal;
-}
-
-int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream* destStream,
-                                                  const CodecInst& codecInst,
-                                                  uint32_t notificationTimeMs) {
-  codec_info_ = codecInst;
-  int32_t retVal = _moduleFile->StartRecordingAudioStream(
-      *destStream, _fileFormat, codecInst, notificationTimeMs);
-
-  if (retVal == 0) {
-    retVal = SetUpAudioEncoder();
-  }
-  if (retVal != 0) {
-    LOG(LS_WARNING) << "Failed to initialize outStream for recording.";
-
-    if (IsRecording()) {
-      StopRecording();
-    }
-  }
-  return retVal;
-}
-
-int32_t FileRecorderImpl::StopRecording() {
-  memset(&codec_info_, 0, sizeof(CodecInst));
-  return _moduleFile->StopRecording();
-}
-
-bool FileRecorderImpl::IsRecording() const {
-  return _moduleFile->IsRecording();
-}
-
-int32_t FileRecorderImpl::RecordAudioToFile(
-    const AudioFrame& incomingAudioFrame) {
-  if (codec_info_.plfreq == 0) {
-    LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not "
-                    << "turned on.";
-    return -1;
-  }
-  AudioFrame tempAudioFrame;
-  tempAudioFrame.samples_per_channel_ = 0;
-  if (incomingAudioFrame.num_channels_ == 2 && !_moduleFile->IsStereo()) {
-    // Recording mono but incoming audio is (interleaved) stereo.
-    tempAudioFrame.num_channels_ = 1;
-    tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
-    tempAudioFrame.samples_per_channel_ =
-        incomingAudioFrame.samples_per_channel_;
-    for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
-      // Sample value is the average of left and right buffer rounded to
-      // closest integer value. Note samples can be either 1 or 2 byte.
-      tempAudioFrame.data_[i] = ((incomingAudioFrame.data_[2 * i] +
-                                  incomingAudioFrame.data_[(2 * i) + 1] + 1) >>
-                                 1);
-    }
-  } else if (incomingAudioFrame.num_channels_ == 1 && _moduleFile->IsStereo()) {
-    // Recording stereo but incoming audio is mono.
-    tempAudioFrame.num_channels_ = 2;
-    tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_;
-    tempAudioFrame.samples_per_channel_ =
-        incomingAudioFrame.samples_per_channel_;
-    for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) {
-      // Duplicate sample to both channels
-      tempAudioFrame.data_[2 * i] = incomingAudioFrame.data_[i];
-      tempAudioFrame.data_[2 * i + 1] = incomingAudioFrame.data_[i];
-    }
-  }
-
-  const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
-  if (tempAudioFrame.samples_per_channel_ != 0) {
-    // If ptrAudioFrame is not empty it contains the audio to be recorded.
-    ptrAudioFrame = &tempAudioFrame;
-  }
-
-  // Encode the audio data before writing to file. Don't encode if the codec
-  // is PCM.
-  // NOTE: stereo recording is only supported for WAV files.
-  // TODO(hellner): WAV expect PCM in little endian byte order. Not
-  // "encoding" with PCM coder should be a problem for big endian systems.
-  size_t encodedLenInBytes = 0;
-  if (_fileFormat == kFileFormatPreencodedFile ||
-      STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
-    if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
-                             &encodedLenInBytes) == -1) {
-      LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname
-                      << " not supported or failed to encode stream.";
-      return -1;
-    }
-  } else {
-    size_t outLen = 0;
-    _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
-                                  codec_info_.plfreq,
-                                  ptrAudioFrame->num_channels_);
-    _audioResampler.Push(
-        ptrAudioFrame->data_,
-        ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_,
-        reinterpret_cast<int16_t*>(_audioBuffer), MAX_AUDIO_BUFFER_IN_BYTES,
-        outLen);
-    encodedLenInBytes = outLen * sizeof(int16_t);
-  }
-
-  // Codec may not be operating at a frame rate of 10 ms. Whenever enough
-  // 10 ms chunks of data has been pushed to the encoder an encoded frame
-  // will be available. Wait until then.
-  if (encodedLenInBytes) {
-    if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1) {
-      return -1;
-    }
-  }
-  return 0;
-}
-
-int32_t FileRecorderImpl::SetUpAudioEncoder() {
-  if (_fileFormat == kFileFormatPreencodedFile ||
-      STR_CASE_CMP(codec_info_.plname, "L16") != 0) {
-    if (_audioEncoder.SetEncodeCodec(codec_info_) == -1) {
-      LOG(LS_ERROR) << "SetUpAudioEncoder() codec " << codec_info_.plname
-                    << " not supported.";
-      return -1;
-    }
-  }
-  return 0;
-}
-
-int32_t FileRecorderImpl::codec_info(CodecInst* codecInst) const {
-  if (codec_info_.plfreq == 0) {
-    return -1;
-  }
-  *codecInst = codec_info_;
-  return 0;
-}
-
-int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
-                                                size_t bufferLength) {
-  return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
-}
-
-}  // namespace
-
-std::unique_ptr<FileRecorder> FileRecorder::CreateFileRecorder(
-    uint32_t instanceID,
-    FileFormats fileFormat) {
-  return std::unique_ptr<FileRecorder>(
-      new FileRecorderImpl(instanceID, fileFormat));
-}
-
-}  // namespace webrtc