| Index: webrtc/voice_engine/file_recorder.cc
|
| diff --git a/webrtc/voice_engine/file_recorder.cc b/webrtc/voice_engine/file_recorder.cc
|
| deleted file mode 100644
|
| index 1d35be9be40fbf103d146c70aafbe482ad14014b..0000000000000000000000000000000000000000
|
| --- a/webrtc/voice_engine/file_recorder.cc
|
| +++ /dev/null
|
| @@ -1,260 +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/voice_engine/file_recorder.h"
|
| -
|
| -#include <list>
|
| -
|
| -#include "webrtc/audio/utility/audio_frame_operations.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/rtc_base/logging.h"
|
| -#include "webrtc/rtc_base/platform_thread.h"
|
| -#include "webrtc/system_wrappers/include/event_wrapper.h"
|
| -#include "webrtc/typedefs.h"
|
| -#include "webrtc/voice_engine/coder.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 FileRecorderImpl : public FileRecorder {
|
| - public:
|
| - FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat);
|
| - ~FileRecorderImpl() override;
|
| -
|
| - // FileRecorder functions.
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| - int32_t RegisterModuleFileCallback(FileCallback* callback) override;
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| - 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:
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| - int32_t WriteEncodedAudioData(const int8_t* audioBuffer, size_t bufferLength);
|
| -
|
| - int32_t SetUpAudioEncoder();
|
| -
|
| - uint32_t _instanceID;
|
| - FileFormats _fileFormat;
|
| - MediaFile* _moduleFile;
|
| -
|
| - CodecInst codec_info_;
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| - int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
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| - AudioCoder _audioEncoder;
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| - Resampler _audioResampler;
|
| -};
|
| -
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| -FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat)
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| - : _instanceID(instanceID),
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| - _fileFormat(fileFormat),
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| - _moduleFile(MediaFile::CreateMediaFile(_instanceID)),
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| - codec_info_(),
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| - _audioBuffer(),
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| - _audioEncoder(instanceID),
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| - _audioResampler() {}
|
| -
|
| -FileRecorderImpl::~FileRecorderImpl() {
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| - 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,
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| - const CodecInst& codecInst,
|
| - uint32_t notificationTimeMs) {
|
| - if (_moduleFile == NULL) {
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| - return -1;
|
| - }
|
| - codec_info_ = codecInst;
|
| - int32_t retVal = 0;
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| - retVal = _moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
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| - codecInst, notificationTimeMs);
|
| -
|
| - if (retVal == 0) {
|
| - retVal = SetUpAudioEncoder();
|
| - }
|
| - if (retVal != 0) {
|
| - LOG(LS_WARNING) << "Failed to initialize file " << fileName
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| - << " for recording.";
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| -
|
| - if (IsRecording()) {
|
| - StopRecording();
|
| - }
|
| - }
|
| - return retVal;
|
| -}
|
| -
|
| -int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream* destStream,
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| - const CodecInst& codecInst,
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| - uint32_t notificationTimeMs) {
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| - codec_info_ = codecInst;
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| - int32_t retVal = _moduleFile->StartRecordingAudioStream(
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| - *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) {
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| - 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_;
|
| - if (!incomingAudioFrame.muted()) {
|
| - AudioFrameOperations::StereoToMono(
|
| - incomingAudioFrame.data(), incomingAudioFrame.samples_per_channel_,
|
| - tempAudioFrame.mutable_data());
|
| - }
|
| - } 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_;
|
| - if (!incomingAudioFrame.muted()) {
|
| - AudioFrameOperations::MonoToStereo(
|
| - incomingAudioFrame.data(), incomingAudioFrame.samples_per_channel_,
|
| - tempAudioFrame.mutable_data());
|
| - }
|
| - }
|
| -
|
| - 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_);
|
| - // TODO(yujo): skip resample if frame is muted.
|
| - _audioResampler.Push(
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| - ptrAudioFrame->data(),
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| - 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
|
|
|