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| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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_ACM2_ACM_RECEIVER_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |
| 12 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |
| 13 | 13 |
| 14 #include <map> | 14 #include <map> |
| 15 #include <memory> |
| 15 #include <string> | 16 #include <string> |
| 16 #include <vector> | 17 #include <vector> |
| 17 | 18 |
| 18 #include "webrtc/base/array_view.h" | 19 #include "webrtc/base/array_view.h" |
| 19 #include "webrtc/base/criticalsection.h" | 20 #include "webrtc/base/criticalsection.h" |
| 20 #include "webrtc/base/optional.h" | 21 #include "webrtc/base/optional.h" |
| 21 #include "webrtc/base/scoped_ptr.h" | |
| 22 #include "webrtc/base/thread_annotations.h" | 22 #include "webrtc/base/thread_annotations.h" |
| 23 #include "webrtc/common_audio/vad/include/webrtc_vad.h" | 23 #include "webrtc/common_audio/vad/include/webrtc_vad.h" |
| 24 #include "webrtc/engine_configurations.h" | 24 #include "webrtc/engine_configurations.h" |
| 25 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" | 25 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" |
| 26 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" | 26 #include "webrtc/modules/audio_coding/acm2/call_statistics.h" |
| 27 #include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h" | 27 #include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h" |
| 28 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 28 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
| 29 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" | 29 #include "webrtc/modules/audio_coding/neteq/include/neteq.h" |
| 30 #include "webrtc/modules/include/module_common_types.h" | 30 #include "webrtc/modules/include/module_common_types.h" |
| 31 #include "webrtc/typedefs.h" | 31 #include "webrtc/typedefs.h" |
| (...skipping 249 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 281 | 281 |
| 282 uint32_t NowInTimestamp(int decoder_sampling_rate) const; | 282 uint32_t NowInTimestamp(int decoder_sampling_rate) const; |
| 283 | 283 |
| 284 rtc::CriticalSection crit_sect_; | 284 rtc::CriticalSection crit_sect_; |
| 285 int id_; // TODO(henrik.lundin) Make const. | 285 int id_; // TODO(henrik.lundin) Make const. |
| 286 const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_); | 286 const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_); |
| 287 AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_); | 287 AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_); |
| 288 ACMResampler resampler_ GUARDED_BY(crit_sect_); | 288 ACMResampler resampler_ GUARDED_BY(crit_sect_); |
| 289 // Used in GetAudio, declared as member to avoid allocating every 10ms. | 289 // Used in GetAudio, declared as member to avoid allocating every 10ms. |
| 290 // TODO(henrik.lundin) Stack-allocate in GetAudio instead? | 290 // TODO(henrik.lundin) Stack-allocate in GetAudio instead? |
| 291 rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_); | 291 std::unique_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_); |
| 292 rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_); | 292 std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_); |
| 293 CallStatistics call_stats_ GUARDED_BY(crit_sect_); | 293 CallStatistics call_stats_ GUARDED_BY(crit_sect_); |
| 294 NetEq* neteq_; | 294 NetEq* neteq_; |
| 295 // Decoders map is keyed by payload type | 295 // Decoders map is keyed by payload type |
| 296 std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_); | 296 std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_); |
| 297 bool vad_enabled_; | 297 bool vad_enabled_; |
| 298 Clock* clock_; // TODO(henrik.lundin) Make const if possible. | 298 Clock* clock_; // TODO(henrik.lundin) Make const if possible. |
| 299 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); | 299 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); |
| 300 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); | 300 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); |
| 301 }; | 301 }; |
| 302 | 302 |
| 303 } // namespace acm2 | 303 } // namespace acm2 |
| 304 | 304 |
| 305 } // namespace webrtc | 305 } // namespace webrtc |
| 306 | 306 |
| 307 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ | 307 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |
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