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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 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ |
13 | 13 |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
16 #include "webrtc/base/scoped_ptr.h" | 16 #include "webrtc/base/scoped_ptr.h" |
17 #include "webrtc/base/thread_annotations.h" | 17 #include "webrtc/base/thread_annotations.h" |
18 #include "webrtc/common_types.h" | 18 #include "webrtc/common_types.h" |
19 #include "webrtc/engine_configurations.h" | 19 #include "webrtc/engine_configurations.h" |
20 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" | 20 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h" |
21 #include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" | 21 #include "webrtc/modules/audio_coding/main/acm2/acm_receiver.h" |
22 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" | 22 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h" |
23 #include "webrtc/modules/audio_coding/main/acm2/codec_manager.h" | 23 #include "webrtc/modules/audio_coding/main/acm2/codec_manager.h" |
| 24 #include "webrtc/modules/audio_coding/main/acm2/delayed_logger.h" |
24 | 25 |
25 namespace webrtc { | 26 namespace webrtc { |
26 | 27 |
27 class CriticalSectionWrapper; | 28 class CriticalSectionWrapper; |
28 class AudioCodingImpl; | 29 class AudioCodingImpl; |
29 | 30 |
30 namespace acm2 { | 31 namespace acm2 { |
31 | 32 |
32 class ACMDTMFDetection; | 33 class ACMDTMFDetection; |
33 | 34 |
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278 // Change required states after starting to receive the codec corresponding | 279 // Change required states after starting to receive the codec corresponding |
279 // to |index|. | 280 // to |index|. |
280 int UpdateUponReceivingCodec(int index); | 281 int UpdateUponReceivingCodec(int index); |
281 | 282 |
282 CriticalSectionWrapper* acm_crit_sect_; | 283 CriticalSectionWrapper* acm_crit_sect_; |
283 int id_; // TODO(henrik.lundin) Make const. | 284 int id_; // TODO(henrik.lundin) Make const. |
284 uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_); | 285 uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_); |
285 uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_); | 286 uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_); |
286 ACMResampler resampler_ GUARDED_BY(acm_crit_sect_); | 287 ACMResampler resampler_ GUARDED_BY(acm_crit_sect_); |
287 AcmReceiver receiver_; // AcmReceiver has it's own internal lock. | 288 AcmReceiver receiver_; // AcmReceiver has it's own internal lock. |
| 289 DelayedLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_); |
288 CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_); | 290 CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_); |
289 | 291 |
290 // This is to keep track of CN instances where we can send DTMFs. | 292 // This is to keep track of CN instances where we can send DTMFs. |
291 uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_); | 293 uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_); |
292 | 294 |
293 // Used when payloads are pushed into ACM without any RTP info | 295 // Used when payloads are pushed into ACM without any RTP info |
294 // One example is when pre-encoded bit-stream is pushed from | 296 // One example is when pre-encoded bit-stream is pushed from |
295 // a file. | 297 // a file. |
296 // IMPORTANT: this variable is only used in IncomingPayload(), therefore, | 298 // IMPORTANT: this variable is only used in IncomingPayload(), therefore, |
297 // no lock acquired when interacting with this variable. If it is going to | 299 // no lock acquired when interacting with this variable. If it is going to |
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383 int playout_frequency_hz_; | 385 int playout_frequency_hz_; |
384 // TODO(henrik.lundin): All members below this line are temporary and should | 386 // TODO(henrik.lundin): All members below this line are temporary and should |
385 // be removed after refactoring is completed. | 387 // be removed after refactoring is completed. |
386 rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_; | 388 rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_; |
387 CodecInst current_send_codec_; | 389 CodecInst current_send_codec_; |
388 }; | 390 }; |
389 | 391 |
390 } // namespace webrtc | 392 } // namespace webrtc |
391 | 393 |
392 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ | 394 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ |
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