niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame^] | 1 | /* |
| 2 | * Copyright (c) 2011 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 "audio_processing_impl.h" |
| 12 | |
| 13 | #include <cassert> |
| 14 | |
| 15 | #include "module_common_types.h" |
| 16 | |
| 17 | #include "critical_section_wrapper.h" |
| 18 | #include "file_wrapper.h" |
| 19 | |
| 20 | #include "audio_buffer.h" |
| 21 | #include "echo_cancellation_impl.h" |
| 22 | #include "echo_control_mobile_impl.h" |
| 23 | #include "high_pass_filter_impl.h" |
| 24 | #include "gain_control_impl.h" |
| 25 | #include "level_estimator_impl.h" |
| 26 | #include "noise_suppression_impl.h" |
| 27 | #include "processing_component.h" |
| 28 | #include "splitting_filter.h" |
| 29 | #include "voice_detection_impl.h" |
| 30 | |
| 31 | namespace webrtc { |
| 32 | namespace { |
| 33 | |
| 34 | enum Events { |
| 35 | kInitializeEvent, |
| 36 | kRenderEvent, |
| 37 | kCaptureEvent |
| 38 | }; |
| 39 | |
| 40 | const char kMagicNumber[] = "#!vqetrace1.2"; |
| 41 | } // namespace |
| 42 | |
| 43 | AudioProcessing* AudioProcessing::Create(int id) { |
| 44 | /*WEBRTC_TRACE(webrtc::kTraceModuleCall, |
| 45 | webrtc::kTraceAudioProcessing, |
| 46 | id, |
| 47 | "AudioProcessing::Create()");*/ |
| 48 | |
| 49 | AudioProcessingImpl* apm = new AudioProcessingImpl(id); |
| 50 | if (apm->Initialize() != kNoError) { |
| 51 | delete apm; |
| 52 | apm = NULL; |
| 53 | } |
| 54 | |
| 55 | return apm; |
| 56 | } |
| 57 | |
| 58 | void AudioProcessing::Destroy(AudioProcessing* apm) { |
| 59 | delete static_cast<AudioProcessingImpl*>(apm); |
| 60 | } |
| 61 | |
| 62 | AudioProcessingImpl::AudioProcessingImpl(int id) |
| 63 | : id_(id), |
| 64 | echo_cancellation_(NULL), |
| 65 | echo_control_mobile_(NULL), |
| 66 | gain_control_(NULL), |
| 67 | high_pass_filter_(NULL), |
| 68 | level_estimator_(NULL), |
| 69 | noise_suppression_(NULL), |
| 70 | voice_detection_(NULL), |
| 71 | debug_file_(FileWrapper::Create()), |
| 72 | crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| 73 | render_audio_(NULL), |
| 74 | capture_audio_(NULL), |
| 75 | sample_rate_hz_(kSampleRate16kHz), |
| 76 | split_sample_rate_hz_(kSampleRate16kHz), |
| 77 | samples_per_channel_(sample_rate_hz_ / 100), |
| 78 | stream_delay_ms_(0), |
| 79 | was_stream_delay_set_(false), |
| 80 | num_render_input_channels_(1), |
| 81 | num_capture_input_channels_(1), |
| 82 | num_capture_output_channels_(1) { |
| 83 | |
| 84 | echo_cancellation_ = new EchoCancellationImpl(this); |
| 85 | component_list_.push_back(echo_cancellation_); |
| 86 | |
| 87 | echo_control_mobile_ = new EchoControlMobileImpl(this); |
| 88 | component_list_.push_back(echo_control_mobile_); |
| 89 | |
| 90 | gain_control_ = new GainControlImpl(this); |
| 91 | component_list_.push_back(gain_control_); |
| 92 | |
| 93 | high_pass_filter_ = new HighPassFilterImpl(this); |
| 94 | component_list_.push_back(high_pass_filter_); |
| 95 | |
| 96 | level_estimator_ = new LevelEstimatorImpl(this); |
| 97 | component_list_.push_back(level_estimator_); |
| 98 | |
| 99 | noise_suppression_ = new NoiseSuppressionImpl(this); |
| 100 | component_list_.push_back(noise_suppression_); |
| 101 | |
| 102 | voice_detection_ = new VoiceDetectionImpl(this); |
| 103 | component_list_.push_back(voice_detection_); |
| 104 | } |
| 105 | |
| 106 | AudioProcessingImpl::~AudioProcessingImpl() { |
| 107 | while (!component_list_.empty()) { |
| 108 | ProcessingComponent* component = component_list_.front(); |
| 109 | component->Destroy(); |
| 110 | delete component; |
| 111 | component_list_.pop_front(); |
| 112 | } |
| 113 | |
| 114 | if (debug_file_->Open()) { |
| 115 | debug_file_->CloseFile(); |
| 116 | } |
| 117 | delete debug_file_; |
| 118 | debug_file_ = NULL; |
| 119 | |
| 120 | delete crit_; |
| 121 | crit_ = NULL; |
| 122 | |
| 123 | if (render_audio_ != NULL) { |
| 124 | delete render_audio_; |
| 125 | render_audio_ = NULL; |
| 126 | } |
| 127 | |
| 128 | if (capture_audio_ != NULL) { |
| 129 | delete capture_audio_; |
| 130 | capture_audio_ = NULL; |
| 131 | } |
| 132 | } |
| 133 | |
| 134 | CriticalSectionWrapper* AudioProcessingImpl::crit() const { |
| 135 | return crit_; |
| 136 | } |
| 137 | |
| 138 | int AudioProcessingImpl::split_sample_rate_hz() const { |
| 139 | return split_sample_rate_hz_; |
| 140 | } |
| 141 | |
| 142 | int AudioProcessingImpl::Initialize() { |
| 143 | CriticalSectionScoped crit_scoped(*crit_); |
| 144 | return InitializeLocked(); |
| 145 | } |
| 146 | |
| 147 | int AudioProcessingImpl::InitializeLocked() { |
| 148 | if (render_audio_ != NULL) { |
| 149 | delete render_audio_; |
| 150 | render_audio_ = NULL; |
| 151 | } |
| 152 | |
| 153 | if (capture_audio_ != NULL) { |
| 154 | delete capture_audio_; |
| 155 | capture_audio_ = NULL; |
| 156 | } |
| 157 | |
| 158 | render_audio_ = new AudioBuffer(num_render_input_channels_, |
| 159 | samples_per_channel_); |
| 160 | capture_audio_ = new AudioBuffer(num_capture_input_channels_, |
| 161 | samples_per_channel_); |
| 162 | |
| 163 | was_stream_delay_set_ = false; |
| 164 | |
| 165 | // Initialize all components. |
| 166 | std::list<ProcessingComponent*>::iterator it; |
| 167 | for (it = component_list_.begin(); it != component_list_.end(); it++) { |
| 168 | int err = (*it)->Initialize(); |
| 169 | if (err != kNoError) { |
| 170 | return err; |
| 171 | } |
| 172 | } |
| 173 | |
| 174 | return kNoError; |
| 175 | } |
| 176 | |
| 177 | int AudioProcessingImpl::set_sample_rate_hz(int rate) { |
| 178 | CriticalSectionScoped crit_scoped(*crit_); |
| 179 | if (rate != kSampleRate8kHz && |
| 180 | rate != kSampleRate16kHz && |
| 181 | rate != kSampleRate32kHz) { |
| 182 | return kBadParameterError; |
| 183 | } |
| 184 | |
| 185 | sample_rate_hz_ = rate; |
| 186 | samples_per_channel_ = rate / 100; |
| 187 | |
| 188 | if (sample_rate_hz_ == kSampleRate32kHz) { |
| 189 | split_sample_rate_hz_ = kSampleRate16kHz; |
| 190 | } else { |
| 191 | split_sample_rate_hz_ = sample_rate_hz_; |
| 192 | } |
| 193 | |
| 194 | return InitializeLocked(); |
| 195 | } |
| 196 | |
| 197 | int AudioProcessingImpl::sample_rate_hz() const { |
| 198 | return sample_rate_hz_; |
| 199 | } |
| 200 | |
| 201 | int AudioProcessingImpl::set_num_reverse_channels(int channels) { |
| 202 | CriticalSectionScoped crit_scoped(*crit_); |
| 203 | // Only stereo supported currently. |
| 204 | if (channels > 2 || channels < 1) { |
| 205 | return kBadParameterError; |
| 206 | } |
| 207 | |
| 208 | num_render_input_channels_ = channels; |
| 209 | |
| 210 | return InitializeLocked(); |
| 211 | } |
| 212 | |
| 213 | int AudioProcessingImpl::num_reverse_channels() const { |
| 214 | return num_render_input_channels_; |
| 215 | } |
| 216 | |
| 217 | int AudioProcessingImpl::set_num_channels( |
| 218 | int input_channels, |
| 219 | int output_channels) { |
| 220 | CriticalSectionScoped crit_scoped(*crit_); |
| 221 | if (output_channels > input_channels) { |
| 222 | return kBadParameterError; |
| 223 | } |
| 224 | |
| 225 | // Only stereo supported currently. |
| 226 | if (input_channels > 2 || input_channels < 1) { |
| 227 | return kBadParameterError; |
| 228 | } |
| 229 | |
| 230 | if (output_channels > 2 || output_channels < 1) { |
| 231 | return kBadParameterError; |
| 232 | } |
| 233 | |
| 234 | num_capture_input_channels_ = input_channels; |
| 235 | num_capture_output_channels_ = output_channels; |
| 236 | |
| 237 | return InitializeLocked(); |
| 238 | } |
| 239 | |
| 240 | int AudioProcessingImpl::num_input_channels() const { |
| 241 | return num_capture_input_channels_; |
| 242 | } |
| 243 | |
| 244 | int AudioProcessingImpl::num_output_channels() const { |
| 245 | return num_capture_output_channels_; |
| 246 | } |
| 247 | |
| 248 | int AudioProcessingImpl::ProcessStream(AudioFrame* frame) { |
| 249 | CriticalSectionScoped crit_scoped(*crit_); |
| 250 | int err = kNoError; |
| 251 | |
| 252 | if (frame == NULL) { |
| 253 | return kNullPointerError; |
| 254 | } |
| 255 | |
| 256 | if (frame->_frequencyInHz != |
| 257 | static_cast<WebRtc_UWord32>(sample_rate_hz_)) { |
| 258 | return kBadSampleRateError; |
| 259 | } |
| 260 | |
| 261 | if (frame->_audioChannel != num_capture_input_channels_) { |
| 262 | return kBadNumberChannelsError; |
| 263 | } |
| 264 | |
| 265 | if (frame->_payloadDataLengthInSamples != samples_per_channel_) { |
| 266 | return kBadDataLengthError; |
| 267 | } |
| 268 | |
| 269 | if (debug_file_->Open()) { |
| 270 | WebRtc_UWord8 event = kCaptureEvent; |
| 271 | if (!debug_file_->Write(&event, sizeof(event))) { |
| 272 | return kFileError; |
| 273 | } |
| 274 | |
| 275 | if (!debug_file_->Write(&frame->_frequencyInHz, |
| 276 | sizeof(frame->_frequencyInHz))) { |
| 277 | return kFileError; |
| 278 | } |
| 279 | |
| 280 | if (!debug_file_->Write(&frame->_audioChannel, |
| 281 | sizeof(frame->_audioChannel))) { |
| 282 | return kFileError; |
| 283 | } |
| 284 | |
| 285 | if (!debug_file_->Write(&frame->_payloadDataLengthInSamples, |
| 286 | sizeof(frame->_payloadDataLengthInSamples))) { |
| 287 | return kFileError; |
| 288 | } |
| 289 | |
| 290 | if (!debug_file_->Write(frame->_payloadData, |
| 291 | sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples * |
| 292 | frame->_audioChannel)) { |
| 293 | return kFileError; |
| 294 | } |
| 295 | } |
| 296 | |
| 297 | capture_audio_->DeinterleaveFrom(frame); |
| 298 | |
| 299 | // TODO(ajm): experiment with mixing and AEC placement. |
| 300 | if (num_capture_output_channels_ < num_capture_input_channels_) { |
| 301 | capture_audio_->Mix(num_capture_output_channels_); |
| 302 | |
| 303 | frame->_audioChannel = num_capture_output_channels_; |
| 304 | } |
| 305 | |
| 306 | if (sample_rate_hz_ == kSampleRate32kHz) { |
| 307 | for (int i = 0; i < num_capture_input_channels_; i++) { |
| 308 | // Split into a low and high band. |
| 309 | SplittingFilterAnalysis(capture_audio_->data(i), |
| 310 | capture_audio_->low_pass_split_data(i), |
| 311 | capture_audio_->high_pass_split_data(i), |
| 312 | capture_audio_->analysis_filter_state1(i), |
| 313 | capture_audio_->analysis_filter_state2(i)); |
| 314 | } |
| 315 | } |
| 316 | |
| 317 | err = high_pass_filter_->ProcessCaptureAudio(capture_audio_); |
| 318 | if (err != kNoError) { |
| 319 | return err; |
| 320 | } |
| 321 | |
| 322 | err = gain_control_->AnalyzeCaptureAudio(capture_audio_); |
| 323 | if (err != kNoError) { |
| 324 | return err; |
| 325 | } |
| 326 | |
| 327 | err = echo_cancellation_->ProcessCaptureAudio(capture_audio_); |
| 328 | if (err != kNoError) { |
| 329 | return err; |
| 330 | } |
| 331 | |
| 332 | if (echo_control_mobile_->is_enabled() && |
| 333 | noise_suppression_->is_enabled()) { |
| 334 | capture_audio_->CopyLowPassToReference(); |
| 335 | } |
| 336 | |
| 337 | err = noise_suppression_->ProcessCaptureAudio(capture_audio_); |
| 338 | if (err != kNoError) { |
| 339 | return err; |
| 340 | } |
| 341 | |
| 342 | err = echo_control_mobile_->ProcessCaptureAudio(capture_audio_); |
| 343 | if (err != kNoError) { |
| 344 | return err; |
| 345 | } |
| 346 | |
| 347 | err = voice_detection_->ProcessCaptureAudio(capture_audio_); |
| 348 | if (err != kNoError) { |
| 349 | return err; |
| 350 | } |
| 351 | |
| 352 | err = gain_control_->ProcessCaptureAudio(capture_audio_); |
| 353 | if (err != kNoError) { |
| 354 | return err; |
| 355 | } |
| 356 | |
| 357 | //err = level_estimator_->ProcessCaptureAudio(capture_audio_); |
| 358 | //if (err != kNoError) { |
| 359 | // return err; |
| 360 | //} |
| 361 | |
| 362 | if (sample_rate_hz_ == kSampleRate32kHz) { |
| 363 | for (int i = 0; i < num_capture_output_channels_; i++) { |
| 364 | // Recombine low and high bands. |
| 365 | SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i), |
| 366 | capture_audio_->high_pass_split_data(i), |
| 367 | capture_audio_->data(i), |
| 368 | capture_audio_->synthesis_filter_state1(i), |
| 369 | capture_audio_->synthesis_filter_state2(i)); |
| 370 | } |
| 371 | } |
| 372 | |
| 373 | capture_audio_->InterleaveTo(frame); |
| 374 | |
| 375 | return kNoError; |
| 376 | } |
| 377 | |
| 378 | int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) { |
| 379 | CriticalSectionScoped crit_scoped(*crit_); |
| 380 | int err = kNoError; |
| 381 | |
| 382 | if (frame == NULL) { |
| 383 | return kNullPointerError; |
| 384 | } |
| 385 | |
| 386 | if (frame->_frequencyInHz != |
| 387 | static_cast<WebRtc_UWord32>(sample_rate_hz_)) { |
| 388 | return kBadSampleRateError; |
| 389 | } |
| 390 | |
| 391 | if (frame->_audioChannel != num_render_input_channels_) { |
| 392 | return kBadNumberChannelsError; |
| 393 | } |
| 394 | |
| 395 | if (frame->_payloadDataLengthInSamples != samples_per_channel_) { |
| 396 | return kBadDataLengthError; |
| 397 | } |
| 398 | |
| 399 | if (debug_file_->Open()) { |
| 400 | WebRtc_UWord8 event = kRenderEvent; |
| 401 | if (!debug_file_->Write(&event, sizeof(event))) { |
| 402 | return kFileError; |
| 403 | } |
| 404 | |
| 405 | if (!debug_file_->Write(&frame->_frequencyInHz, |
| 406 | sizeof(frame->_frequencyInHz))) { |
| 407 | return kFileError; |
| 408 | } |
| 409 | |
| 410 | if (!debug_file_->Write(&frame->_audioChannel, |
| 411 | sizeof(frame->_audioChannel))) { |
| 412 | return kFileError; |
| 413 | } |
| 414 | |
| 415 | if (!debug_file_->Write(&frame->_payloadDataLengthInSamples, |
| 416 | sizeof(frame->_payloadDataLengthInSamples))) { |
| 417 | return kFileError; |
| 418 | } |
| 419 | |
| 420 | if (!debug_file_->Write(frame->_payloadData, |
| 421 | sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples * |
| 422 | frame->_audioChannel)) { |
| 423 | return kFileError; |
| 424 | } |
| 425 | } |
| 426 | |
| 427 | render_audio_->DeinterleaveFrom(frame); |
| 428 | |
| 429 | // TODO(ajm): turn the splitting filter into a component? |
| 430 | if (sample_rate_hz_ == kSampleRate32kHz) { |
| 431 | for (int i = 0; i < num_render_input_channels_; i++) { |
| 432 | // Split into low and high band. |
| 433 | SplittingFilterAnalysis(render_audio_->data(i), |
| 434 | render_audio_->low_pass_split_data(i), |
| 435 | render_audio_->high_pass_split_data(i), |
| 436 | render_audio_->analysis_filter_state1(i), |
| 437 | render_audio_->analysis_filter_state2(i)); |
| 438 | } |
| 439 | } |
| 440 | |
| 441 | // TODO(ajm): warnings possible from components? |
| 442 | err = echo_cancellation_->ProcessRenderAudio(render_audio_); |
| 443 | if (err != kNoError) { |
| 444 | return err; |
| 445 | } |
| 446 | |
| 447 | err = echo_control_mobile_->ProcessRenderAudio(render_audio_); |
| 448 | if (err != kNoError) { |
| 449 | return err; |
| 450 | } |
| 451 | |
| 452 | err = gain_control_->ProcessRenderAudio(render_audio_); |
| 453 | if (err != kNoError) { |
| 454 | return err; |
| 455 | } |
| 456 | |
| 457 | //err = level_estimator_->AnalyzeReverseStream(render_audio_); |
| 458 | //if (err != kNoError) { |
| 459 | // return err; |
| 460 | //} |
| 461 | |
| 462 | was_stream_delay_set_ = false; |
| 463 | return err; // TODO(ajm): this is for returning warnings; necessary? |
| 464 | } |
| 465 | |
| 466 | int AudioProcessingImpl::set_stream_delay_ms(int delay) { |
| 467 | was_stream_delay_set_ = true; |
| 468 | if (delay < 0) { |
| 469 | return kBadParameterError; |
| 470 | } |
| 471 | |
| 472 | // TODO(ajm): the max is rather arbitrarily chosen; investigate. |
| 473 | if (delay > 500) { |
| 474 | stream_delay_ms_ = 500; |
| 475 | return kBadStreamParameterWarning; |
| 476 | } |
| 477 | |
| 478 | stream_delay_ms_ = delay; |
| 479 | return kNoError; |
| 480 | } |
| 481 | |
| 482 | int AudioProcessingImpl::stream_delay_ms() const { |
| 483 | return stream_delay_ms_; |
| 484 | } |
| 485 | |
| 486 | bool AudioProcessingImpl::was_stream_delay_set() const { |
| 487 | return was_stream_delay_set_; |
| 488 | } |
| 489 | |
| 490 | int AudioProcessingImpl::StartDebugRecording( |
| 491 | const char filename[AudioProcessing::kMaxFilenameSize]) { |
| 492 | CriticalSectionScoped crit_scoped(*crit_); |
| 493 | assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize); |
| 494 | |
| 495 | if (filename == NULL) { |
| 496 | return kNullPointerError; |
| 497 | } |
| 498 | |
| 499 | // Stop any ongoing recording. |
| 500 | if (debug_file_->Open()) { |
| 501 | if (debug_file_->CloseFile() == -1) { |
| 502 | return kFileError; |
| 503 | } |
| 504 | } |
| 505 | |
| 506 | if (debug_file_->OpenFile(filename, false) == -1) { |
| 507 | debug_file_->CloseFile(); |
| 508 | return kFileError; |
| 509 | } |
| 510 | |
| 511 | if (debug_file_->WriteText("%s\n", kMagicNumber) == -1) { |
| 512 | debug_file_->CloseFile(); |
| 513 | return kFileError; |
| 514 | } |
| 515 | |
| 516 | // TODO(ajm): should we do this? If so, we need the number of channels etc. |
| 517 | // Record the default sample rate. |
| 518 | WebRtc_UWord8 event = kInitializeEvent; |
| 519 | if (!debug_file_->Write(&event, sizeof(event))) { |
| 520 | return kFileError; |
| 521 | } |
| 522 | |
| 523 | if (!debug_file_->Write(&sample_rate_hz_, sizeof(sample_rate_hz_))) { |
| 524 | return kFileError; |
| 525 | } |
| 526 | |
| 527 | return kNoError; |
| 528 | } |
| 529 | |
| 530 | int AudioProcessingImpl::StopDebugRecording() { |
| 531 | CriticalSectionScoped crit_scoped(*crit_); |
| 532 | // We just return if recording hasn't started. |
| 533 | if (debug_file_->Open()) { |
| 534 | if (debug_file_->CloseFile() == -1) { |
| 535 | return kFileError; |
| 536 | } |
| 537 | } |
| 538 | |
| 539 | return kNoError; |
| 540 | } |
| 541 | |
| 542 | EchoCancellation* AudioProcessingImpl::echo_cancellation() const { |
| 543 | return echo_cancellation_; |
| 544 | } |
| 545 | |
| 546 | EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const { |
| 547 | return echo_control_mobile_; |
| 548 | } |
| 549 | |
| 550 | GainControl* AudioProcessingImpl::gain_control() const { |
| 551 | return gain_control_; |
| 552 | } |
| 553 | |
| 554 | HighPassFilter* AudioProcessingImpl::high_pass_filter() const { |
| 555 | return high_pass_filter_; |
| 556 | } |
| 557 | |
| 558 | LevelEstimator* AudioProcessingImpl::level_estimator() const { |
| 559 | return level_estimator_; |
| 560 | } |
| 561 | |
| 562 | NoiseSuppression* AudioProcessingImpl::noise_suppression() const { |
| 563 | return noise_suppression_; |
| 564 | } |
| 565 | |
| 566 | VoiceDetection* AudioProcessingImpl::voice_detection() const { |
| 567 | return voice_detection_; |
| 568 | } |
| 569 | |
| 570 | WebRtc_Word32 AudioProcessingImpl::Version(WebRtc_Word8* version, |
| 571 | WebRtc_UWord32& bytes_remaining, WebRtc_UWord32& position) const { |
| 572 | if (version == NULL) { |
| 573 | /*WEBRTC_TRACE(webrtc::kTraceError, |
| 574 | webrtc::kTraceAudioProcessing, |
| 575 | -1, |
| 576 | "Null version pointer");*/ |
| 577 | return kNullPointerError; |
| 578 | } |
| 579 | memset(&version[position], 0, bytes_remaining); |
| 580 | |
| 581 | WebRtc_Word8 my_version[] = "AudioProcessing 1.0.0"; |
| 582 | // Includes null termination. |
| 583 | WebRtc_UWord32 length = static_cast<WebRtc_UWord32>(strlen(my_version)); |
| 584 | if (bytes_remaining < length) { |
| 585 | /*WEBRTC_TRACE(webrtc::kTraceError, |
| 586 | webrtc::kTraceAudioProcessing, |
| 587 | -1, |
| 588 | "Buffer of insufficient length");*/ |
| 589 | return kBadParameterError; |
| 590 | } |
| 591 | memcpy(&version[position], my_version, length); |
| 592 | bytes_remaining -= length; |
| 593 | position += length; |
| 594 | |
| 595 | std::list<ProcessingComponent*>::const_iterator it; |
| 596 | for (it = component_list_.begin(); it != component_list_.end(); it++) { |
| 597 | char component_version[256]; |
| 598 | strcpy(component_version, "\n"); |
| 599 | int err = (*it)->get_version(&component_version[1], |
| 600 | sizeof(component_version) - 1); |
| 601 | if (err != kNoError) { |
| 602 | return err; |
| 603 | } |
| 604 | if (strncmp(&component_version[1], "\0", 1) == 0) { |
| 605 | // Assume empty if first byte is NULL. |
| 606 | continue; |
| 607 | } |
| 608 | |
| 609 | length = static_cast<WebRtc_UWord32>(strlen(component_version)); |
| 610 | if (bytes_remaining < length) { |
| 611 | /*WEBRTC_TRACE(webrtc::kTraceError, |
| 612 | webrtc::kTraceAudioProcessing, |
| 613 | -1, |
| 614 | "Buffer of insufficient length");*/ |
| 615 | return kBadParameterError; |
| 616 | } |
| 617 | memcpy(&version[position], component_version, length); |
| 618 | bytes_remaining -= length; |
| 619 | position += length; |
| 620 | } |
| 621 | |
| 622 | return kNoError; |
| 623 | } |
| 624 | |
| 625 | WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) { |
| 626 | CriticalSectionScoped crit_scoped(*crit_); |
| 627 | /*WEBRTC_TRACE(webrtc::kTraceModuleCall, |
| 628 | webrtc::kTraceAudioProcessing, |
| 629 | id_, |
| 630 | "ChangeUniqueId(new id = %d)", |
| 631 | id);*/ |
| 632 | id_ = id; |
| 633 | |
| 634 | return kNoError; |
| 635 | } |
| 636 | } // namespace webrtc |