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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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322 AudioFrame(); | 322 AudioFrame(); |
323 | 323 |
324 // Resets all members to their default state. | 324 // Resets all members to their default state. |
325 void Reset(); | 325 void Reset(); |
326 // Same as Reset(), but leaves mute state unchanged. Muting a frame requires | 326 // Same as Reset(), but leaves mute state unchanged. Muting a frame requires |
327 // the buffer to be zeroed on the next call to mutable_data(). Callers | 327 // the buffer to be zeroed on the next call to mutable_data(). Callers |
328 // intending to write to the buffer immediately after Reset() can instead use | 328 // intending to write to the buffer immediately after Reset() can instead use |
329 // ResetWithoutMuting() to skip this wasteful zeroing. | 329 // ResetWithoutMuting() to skip this wasteful zeroing. |
330 void ResetWithoutMuting(); | 330 void ResetWithoutMuting(); |
331 | 331 |
332 void UpdateFrame(int id, uint32_t timestamp, const int16_t* data, | 332 // TODO(solenberg): Remove once downstream users of AudioFrame have updated. |
| 333 RTC_DEPRECATED |
| 334 void UpdateFrame(int id, uint32_t timestamp, const int16_t* data, |
| 335 size_t samples_per_channel, int sample_rate_hz, |
| 336 SpeechType speech_type, VADActivity vad_activity, |
| 337 size_t num_channels = 1) { |
| 338 RTC_UNUSED(id); |
| 339 UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz, |
| 340 speech_type, vad_activity, num_channels); |
| 341 } |
| 342 |
| 343 void UpdateFrame(uint32_t timestamp, const int16_t* data, |
333 size_t samples_per_channel, int sample_rate_hz, | 344 size_t samples_per_channel, int sample_rate_hz, |
334 SpeechType speech_type, VADActivity vad_activity, | 345 SpeechType speech_type, VADActivity vad_activity, |
335 size_t num_channels = 1); | 346 size_t num_channels = 1); |
336 | 347 |
337 void CopyFrom(const AudioFrame& src); | 348 void CopyFrom(const AudioFrame& src); |
338 | 349 |
339 // data() returns a zeroed static buffer if the frame is muted. | 350 // data() returns a zeroed static buffer if the frame is muted. |
340 // mutable_frame() always returns a non-static buffer; the first call to | 351 // mutable_frame() always returns a non-static buffer; the first call to |
341 // mutable_frame() zeros the non-static buffer and marks the frame unmuted. | 352 // mutable_frame() zeros the non-static buffer and marks the frame unmuted. |
342 const int16_t* data() const; | 353 const int16_t* data() const; |
343 int16_t* mutable_data(); | 354 int16_t* mutable_data(); |
344 | 355 |
345 // Prefer to mute frames using AudioFrameOperations::Mute. | 356 // Prefer to mute frames using AudioFrameOperations::Mute. |
346 void Mute(); | 357 void Mute(); |
347 // Frame is muted by default. | 358 // Frame is muted by default. |
348 bool muted() const; | 359 bool muted() const; |
349 | 360 |
350 // These methods are deprecated. Use the functions in | 361 // These methods are deprecated. Use the functions in |
351 // webrtc/audio/utility instead. These methods will exists for a | 362 // webrtc/audio/utility instead. These methods will exists for a |
352 // short period of time until webrtc clients have updated. See | 363 // short period of time until webrtc clients have updated. See |
353 // webrtc:6548 for details. | 364 // webrtc:6548 for details. |
354 RTC_DEPRECATED AudioFrame& operator>>=(const int rhs); | 365 RTC_DEPRECATED AudioFrame& operator>>=(const int rhs); |
355 RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs); | 366 RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs); |
356 | 367 |
357 int id_; | |
358 // RTP timestamp of the first sample in the AudioFrame. | 368 // RTP timestamp of the first sample in the AudioFrame. |
359 uint32_t timestamp_ = 0; | 369 uint32_t timestamp_ = 0; |
360 // Time since the first frame in milliseconds. | 370 // Time since the first frame in milliseconds. |
361 // -1 represents an uninitialized value. | 371 // -1 represents an uninitialized value. |
362 int64_t elapsed_time_ms_ = -1; | 372 int64_t elapsed_time_ms_ = -1; |
363 // NTP time of the estimated capture time in local timebase in milliseconds. | 373 // NTP time of the estimated capture time in local timebase in milliseconds. |
364 // -1 represents an uninitialized value. | 374 // -1 represents an uninitialized value. |
365 int64_t ntp_time_ms_ = -1; | 375 int64_t ntp_time_ms_ = -1; |
366 size_t samples_per_channel_ = 0; | 376 size_t samples_per_channel_ = 0; |
367 int sample_rate_hz_ = 0; | 377 int sample_rate_hz_ = 0; |
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389 // Visual Studio doesn't like this in the class definition. | 399 // Visual Studio doesn't like this in the class definition. |
390 static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes"); | 400 static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes"); |
391 } | 401 } |
392 | 402 |
393 inline void AudioFrame::Reset() { | 403 inline void AudioFrame::Reset() { |
394 ResetWithoutMuting(); | 404 ResetWithoutMuting(); |
395 muted_ = true; | 405 muted_ = true; |
396 } | 406 } |
397 | 407 |
398 inline void AudioFrame::ResetWithoutMuting() { | 408 inline void AudioFrame::ResetWithoutMuting() { |
399 id_ = -1; | |
400 // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize | 409 // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize |
401 // to an invalid value, or add a new member to indicate invalidity. | 410 // to an invalid value, or add a new member to indicate invalidity. |
402 timestamp_ = 0; | 411 timestamp_ = 0; |
403 elapsed_time_ms_ = -1; | 412 elapsed_time_ms_ = -1; |
404 ntp_time_ms_ = -1; | 413 ntp_time_ms_ = -1; |
405 samples_per_channel_ = 0; | 414 samples_per_channel_ = 0; |
406 sample_rate_hz_ = 0; | 415 sample_rate_hz_ = 0; |
407 num_channels_ = 0; | 416 num_channels_ = 0; |
408 speech_type_ = kUndefined; | 417 speech_type_ = kUndefined; |
409 vad_activity_ = kVadUnknown; | 418 vad_activity_ = kVadUnknown; |
410 } | 419 } |
411 | 420 |
412 inline void AudioFrame::UpdateFrame(int id, | 421 inline void AudioFrame::UpdateFrame(uint32_t timestamp, |
413 uint32_t timestamp, | |
414 const int16_t* data, | 422 const int16_t* data, |
415 size_t samples_per_channel, | 423 size_t samples_per_channel, |
416 int sample_rate_hz, | 424 int sample_rate_hz, |
417 SpeechType speech_type, | 425 SpeechType speech_type, |
418 VADActivity vad_activity, | 426 VADActivity vad_activity, |
419 size_t num_channels) { | 427 size_t num_channels) { |
420 id_ = id; | |
421 timestamp_ = timestamp; | 428 timestamp_ = timestamp; |
422 samples_per_channel_ = samples_per_channel; | 429 samples_per_channel_ = samples_per_channel; |
423 sample_rate_hz_ = sample_rate_hz; | 430 sample_rate_hz_ = sample_rate_hz; |
424 speech_type_ = speech_type; | 431 speech_type_ = speech_type; |
425 vad_activity_ = vad_activity; | 432 vad_activity_ = vad_activity; |
426 num_channels_ = num_channels; | 433 num_channels_ = num_channels; |
427 | 434 |
428 const size_t length = samples_per_channel * num_channels; | 435 const size_t length = samples_per_channel * num_channels; |
429 assert(length <= kMaxDataSizeSamples); | 436 assert(length <= kMaxDataSizeSamples); |
430 if (data != nullptr) { | 437 if (data != nullptr) { |
431 memcpy(data_, data, sizeof(int16_t) * length); | 438 memcpy(data_, data, sizeof(int16_t) * length); |
432 muted_ = false; | 439 muted_ = false; |
433 } else { | 440 } else { |
434 muted_ = true; | 441 muted_ = true; |
435 } | 442 } |
436 } | 443 } |
437 | 444 |
438 inline void AudioFrame::CopyFrom(const AudioFrame& src) { | 445 inline void AudioFrame::CopyFrom(const AudioFrame& src) { |
439 if (this == &src) return; | 446 if (this == &src) return; |
440 | 447 |
441 id_ = src.id_; | |
442 timestamp_ = src.timestamp_; | 448 timestamp_ = src.timestamp_; |
443 elapsed_time_ms_ = src.elapsed_time_ms_; | 449 elapsed_time_ms_ = src.elapsed_time_ms_; |
444 ntp_time_ms_ = src.ntp_time_ms_; | 450 ntp_time_ms_ = src.ntp_time_ms_; |
445 muted_ = src.muted(); | 451 muted_ = src.muted(); |
446 samples_per_channel_ = src.samples_per_channel_; | 452 samples_per_channel_ = src.samples_per_channel_; |
447 sample_rate_hz_ = src.sample_rate_hz_; | 453 sample_rate_hz_ = src.sample_rate_hz_; |
448 speech_type_ = src.speech_type_; | 454 speech_type_ = src.speech_type_; |
449 vad_activity_ = src.vad_activity_; | 455 vad_activity_ = src.vad_activity_; |
450 num_channels_ = src.num_channels_; | 456 num_channels_ = src.num_channels_; |
451 | 457 |
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638 static constexpr int kNotAProbe = -1; | 644 static constexpr int kNotAProbe = -1; |
639 int send_bitrate_bps = -1; | 645 int send_bitrate_bps = -1; |
640 int probe_cluster_id = kNotAProbe; | 646 int probe_cluster_id = kNotAProbe; |
641 int probe_cluster_min_probes = -1; | 647 int probe_cluster_min_probes = -1; |
642 int probe_cluster_min_bytes = -1; | 648 int probe_cluster_min_bytes = -1; |
643 }; | 649 }; |
644 | 650 |
645 } // namespace webrtc | 651 } // namespace webrtc |
646 | 652 |
647 #endif // MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ | 653 #endif // MODULES_INCLUDE_MODULE_COMMON_TYPES_H_ |
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