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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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143 | 143 |
144 // Minimum playout delay. | 144 // Minimum playout delay. |
145 int SetMinimumPlayoutDelay(int time_ms) override; | 145 int SetMinimumPlayoutDelay(int time_ms) override; |
146 | 146 |
147 // Maximum playout delay. | 147 // Maximum playout delay. |
148 int SetMaximumPlayoutDelay(int time_ms) override; | 148 int SetMaximumPlayoutDelay(int time_ms) override; |
149 | 149 |
150 // Smallest latency NetEq will maintain. | 150 // Smallest latency NetEq will maintain. |
151 int LeastRequiredDelayMs() const override; | 151 int LeastRequiredDelayMs() const override; |
152 | 152 |
153 // Impose an initial delay on playout. ACM plays silence until |delay_ms| | |
154 // audio is accumulated in NetEq buffer, then starts decoding payloads. | |
155 int SetInitialPlayoutDelay(int delay_ms) override; | |
156 | |
157 // Get playout timestamp. | 153 // Get playout timestamp. |
158 int PlayoutTimestamp(uint32_t* timestamp) override; | 154 int PlayoutTimestamp(uint32_t* timestamp) override; |
159 | 155 |
160 // Get 10 milliseconds of raw audio data to play out, and | 156 // Get 10 milliseconds of raw audio data to play out, and |
161 // automatic resample to the requested frequency if > 0. | 157 // automatic resample to the requested frequency if > 0. |
162 int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override; | 158 int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override; |
163 | 159 |
164 ///////////////////////////////////////// | 160 ///////////////////////////////////////// |
165 // Statistics | 161 // Statistics |
166 // | 162 // |
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276 const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_; | 272 const rtc::scoped_ptr<CriticalSectionWrapper> callback_crit_sect_; |
277 AudioPacketizationCallback* packetization_callback_ | 273 AudioPacketizationCallback* packetization_callback_ |
278 GUARDED_BY(callback_crit_sect_); | 274 GUARDED_BY(callback_crit_sect_); |
279 ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_); | 275 ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_); |
280 }; | 276 }; |
281 | 277 |
282 } // namespace acm2 | 278 } // namespace acm2 |
283 } // namespace webrtc | 279 } // namespace webrtc |
284 | 280 |
285 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ | 281 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ |
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