OLD | NEW |
| (Empty) |
1 /* | |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #include "webrtc/modules/utility/include/file_recorder.h" | |
12 | |
13 #include <list> | |
14 | |
15 #include "webrtc/base/platform_thread.h" | |
16 #include "webrtc/common_audio/resampler/include/resampler.h" | |
17 #include "webrtc/common_types.h" | |
18 #include "webrtc/engine_configurations.h" | |
19 #include "webrtc/modules/include/module_common_types.h" | |
20 #include "webrtc/modules/media_file/media_file.h" | |
21 #include "webrtc/modules/media_file/media_file_defines.h" | |
22 #include "webrtc/modules/utility/source/coder.h" | |
23 #include "webrtc/system_wrappers/include/event_wrapper.h" | |
24 #include "webrtc/system_wrappers/include/logging.h" | |
25 #include "webrtc/typedefs.h" | |
26 | |
27 namespace webrtc { | |
28 | |
29 namespace { | |
30 | |
31 // The largest decoded frame size in samples (60ms with 32kHz sample rate). | |
32 enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60 * 32 }; | |
33 enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES * 2 }; | |
34 enum { kMaxAudioBufferQueueLength = 100 }; | |
35 | |
36 class CriticalSectionWrapper; | |
37 | |
38 class FileRecorderImpl : public FileRecorder { | |
39 public: | |
40 FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat); | |
41 ~FileRecorderImpl() override; | |
42 | |
43 // FileRecorder functions. | |
44 int32_t RegisterModuleFileCallback(FileCallback* callback) override; | |
45 FileFormats RecordingFileFormat() const override; | |
46 int32_t StartRecordingAudioFile(const char* fileName, | |
47 const CodecInst& codecInst, | |
48 uint32_t notificationTimeMs) override; | |
49 int32_t StartRecordingAudioFile(OutStream* destStream, | |
50 const CodecInst& codecInst, | |
51 uint32_t notificationTimeMs) override; | |
52 int32_t StopRecording() override; | |
53 bool IsRecording() const override; | |
54 int32_t codec_info(CodecInst* codecInst) const override; | |
55 int32_t RecordAudioToFile(const AudioFrame& frame) override; | |
56 | |
57 private: | |
58 int32_t WriteEncodedAudioData(const int8_t* audioBuffer, size_t bufferLength); | |
59 | |
60 int32_t SetUpAudioEncoder(); | |
61 | |
62 uint32_t _instanceID; | |
63 FileFormats _fileFormat; | |
64 MediaFile* _moduleFile; | |
65 | |
66 CodecInst codec_info_; | |
67 int8_t _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES]; | |
68 AudioCoder _audioEncoder; | |
69 Resampler _audioResampler; | |
70 }; | |
71 | |
72 FileRecorderImpl::FileRecorderImpl(uint32_t instanceID, FileFormats fileFormat) | |
73 : _instanceID(instanceID), | |
74 _fileFormat(fileFormat), | |
75 _moduleFile(MediaFile::CreateMediaFile(_instanceID)), | |
76 codec_info_(), | |
77 _audioBuffer(), | |
78 _audioEncoder(instanceID), | |
79 _audioResampler() {} | |
80 | |
81 FileRecorderImpl::~FileRecorderImpl() { | |
82 MediaFile::DestroyMediaFile(_moduleFile); | |
83 } | |
84 | |
85 FileFormats FileRecorderImpl::RecordingFileFormat() const { | |
86 return _fileFormat; | |
87 } | |
88 | |
89 int32_t FileRecorderImpl::RegisterModuleFileCallback(FileCallback* callback) { | |
90 if (_moduleFile == NULL) { | |
91 return -1; | |
92 } | |
93 return _moduleFile->SetModuleFileCallback(callback); | |
94 } | |
95 | |
96 int32_t FileRecorderImpl::StartRecordingAudioFile(const char* fileName, | |
97 const CodecInst& codecInst, | |
98 uint32_t notificationTimeMs) { | |
99 if (_moduleFile == NULL) { | |
100 return -1; | |
101 } | |
102 codec_info_ = codecInst; | |
103 int32_t retVal = 0; | |
104 retVal = _moduleFile->StartRecordingAudioFile(fileName, _fileFormat, | |
105 codecInst, notificationTimeMs); | |
106 | |
107 if (retVal == 0) { | |
108 retVal = SetUpAudioEncoder(); | |
109 } | |
110 if (retVal != 0) { | |
111 LOG(LS_WARNING) << "Failed to initialize file " << fileName | |
112 << " for recording."; | |
113 | |
114 if (IsRecording()) { | |
115 StopRecording(); | |
116 } | |
117 } | |
118 return retVal; | |
119 } | |
120 | |
121 int32_t FileRecorderImpl::StartRecordingAudioFile(OutStream* destStream, | |
122 const CodecInst& codecInst, | |
123 uint32_t notificationTimeMs) { | |
124 codec_info_ = codecInst; | |
125 int32_t retVal = _moduleFile->StartRecordingAudioStream( | |
126 *destStream, _fileFormat, codecInst, notificationTimeMs); | |
127 | |
128 if (retVal == 0) { | |
129 retVal = SetUpAudioEncoder(); | |
130 } | |
131 if (retVal != 0) { | |
132 LOG(LS_WARNING) << "Failed to initialize outStream for recording."; | |
133 | |
134 if (IsRecording()) { | |
135 StopRecording(); | |
136 } | |
137 } | |
138 return retVal; | |
139 } | |
140 | |
141 int32_t FileRecorderImpl::StopRecording() { | |
142 memset(&codec_info_, 0, sizeof(CodecInst)); | |
143 return _moduleFile->StopRecording(); | |
144 } | |
145 | |
146 bool FileRecorderImpl::IsRecording() const { | |
147 return _moduleFile->IsRecording(); | |
148 } | |
149 | |
150 int32_t FileRecorderImpl::RecordAudioToFile( | |
151 const AudioFrame& incomingAudioFrame) { | |
152 if (codec_info_.plfreq == 0) { | |
153 LOG(LS_WARNING) << "RecordAudioToFile() recording audio is not " | |
154 << "turned on."; | |
155 return -1; | |
156 } | |
157 AudioFrame tempAudioFrame; | |
158 tempAudioFrame.samples_per_channel_ = 0; | |
159 if (incomingAudioFrame.num_channels_ == 2 && !_moduleFile->IsStereo()) { | |
160 // Recording mono but incoming audio is (interleaved) stereo. | |
161 tempAudioFrame.num_channels_ = 1; | |
162 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; | |
163 tempAudioFrame.samples_per_channel_ = | |
164 incomingAudioFrame.samples_per_channel_; | |
165 for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) { | |
166 // Sample value is the average of left and right buffer rounded to | |
167 // closest integer value. Note samples can be either 1 or 2 byte. | |
168 tempAudioFrame.data_[i] = ((incomingAudioFrame.data_[2 * i] + | |
169 incomingAudioFrame.data_[(2 * i) + 1] + 1) >> | |
170 1); | |
171 } | |
172 } else if (incomingAudioFrame.num_channels_ == 1 && _moduleFile->IsStereo()) { | |
173 // Recording stereo but incoming audio is mono. | |
174 tempAudioFrame.num_channels_ = 2; | |
175 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; | |
176 tempAudioFrame.samples_per_channel_ = | |
177 incomingAudioFrame.samples_per_channel_; | |
178 for (size_t i = 0; i < (incomingAudioFrame.samples_per_channel_); i++) { | |
179 // Duplicate sample to both channels | |
180 tempAudioFrame.data_[2 * i] = incomingAudioFrame.data_[i]; | |
181 tempAudioFrame.data_[2 * i + 1] = incomingAudioFrame.data_[i]; | |
182 } | |
183 } | |
184 | |
185 const AudioFrame* ptrAudioFrame = &incomingAudioFrame; | |
186 if (tempAudioFrame.samples_per_channel_ != 0) { | |
187 // If ptrAudioFrame is not empty it contains the audio to be recorded. | |
188 ptrAudioFrame = &tempAudioFrame; | |
189 } | |
190 | |
191 // Encode the audio data before writing to file. Don't encode if the codec | |
192 // is PCM. | |
193 // NOTE: stereo recording is only supported for WAV files. | |
194 // TODO(hellner): WAV expect PCM in little endian byte order. Not | |
195 // "encoding" with PCM coder should be a problem for big endian systems. | |
196 size_t encodedLenInBytes = 0; | |
197 if (_fileFormat == kFileFormatPreencodedFile || | |
198 STR_CASE_CMP(codec_info_.plname, "L16") != 0) { | |
199 if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer, | |
200 &encodedLenInBytes) == -1) { | |
201 LOG(LS_WARNING) << "RecordAudioToFile() codec " << codec_info_.plname | |
202 << " not supported or failed to encode stream."; | |
203 return -1; | |
204 } | |
205 } else { | |
206 size_t outLen = 0; | |
207 _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, | |
208 codec_info_.plfreq, | |
209 ptrAudioFrame->num_channels_); | |
210 _audioResampler.Push( | |
211 ptrAudioFrame->data_, | |
212 ptrAudioFrame->samples_per_channel_ * ptrAudioFrame->num_channels_, | |
213 reinterpret_cast<int16_t*>(_audioBuffer), MAX_AUDIO_BUFFER_IN_BYTES, | |
214 outLen); | |
215 encodedLenInBytes = outLen * sizeof(int16_t); | |
216 } | |
217 | |
218 // Codec may not be operating at a frame rate of 10 ms. Whenever enough | |
219 // 10 ms chunks of data has been pushed to the encoder an encoded frame | |
220 // will be available. Wait until then. | |
221 if (encodedLenInBytes) { | |
222 if (WriteEncodedAudioData(_audioBuffer, encodedLenInBytes) == -1) { | |
223 return -1; | |
224 } | |
225 } | |
226 return 0; | |
227 } | |
228 | |
229 int32_t FileRecorderImpl::SetUpAudioEncoder() { | |
230 if (_fileFormat == kFileFormatPreencodedFile || | |
231 STR_CASE_CMP(codec_info_.plname, "L16") != 0) { | |
232 if (_audioEncoder.SetEncodeCodec(codec_info_) == -1) { | |
233 LOG(LS_ERROR) << "SetUpAudioEncoder() codec " << codec_info_.plname | |
234 << " not supported."; | |
235 return -1; | |
236 } | |
237 } | |
238 return 0; | |
239 } | |
240 | |
241 int32_t FileRecorderImpl::codec_info(CodecInst* codecInst) const { | |
242 if (codec_info_.plfreq == 0) { | |
243 return -1; | |
244 } | |
245 *codecInst = codec_info_; | |
246 return 0; | |
247 } | |
248 | |
249 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer, | |
250 size_t bufferLength) { | |
251 return _moduleFile->IncomingAudioData(audioBuffer, bufferLength); | |
252 } | |
253 | |
254 } // namespace | |
255 | |
256 std::unique_ptr<FileRecorder> FileRecorder::CreateFileRecorder( | |
257 uint32_t instanceID, | |
258 FileFormats fileFormat) { | |
259 return std::unique_ptr<FileRecorder>( | |
260 new FileRecorderImpl(instanceID, fileFormat)); | |
261 } | |
262 | |
263 } // namespace webrtc | |
OLD | NEW |