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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 |
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106 // Return value : 0 if OK. | 106 // Return value : 0 if OK. |
107 // <0 if NetEq returned an error. | 107 // <0 if NetEq returned an error. |
108 // | 108 // |
109 int AddCodec(int acm_codec_id, | 109 int AddCodec(int acm_codec_id, |
110 uint8_t payload_type, | 110 uint8_t payload_type, |
111 size_t channels, | 111 size_t channels, |
112 int sample_rate_hz, | 112 int sample_rate_hz, |
113 AudioDecoder* audio_decoder, | 113 AudioDecoder* audio_decoder, |
114 const std::string& name); | 114 const std::string& name); |
115 | 115 |
| 116 // Adds a new decoder to the NetEq codec database. Returns true iff |
| 117 // successful. |
| 118 bool AddCodec(int rtp_payload_type, const SdpAudioFormat& audio_format); |
| 119 |
116 // | 120 // |
117 // Sets a minimum delay for packet buffer. The given delay is maintained, | 121 // Sets a minimum delay for packet buffer. The given delay is maintained, |
118 // unless channel condition dictates a higher delay. | 122 // unless channel condition dictates a higher delay. |
119 // | 123 // |
120 // Input: | 124 // Input: |
121 // - delay_ms : minimum delay in milliseconds. | 125 // - delay_ms : minimum delay in milliseconds. |
122 // | 126 // |
123 // Return value : 0 if OK. | 127 // Return value : 0 if OK. |
124 // <0 if NetEq returned an error. | 128 // <0 if NetEq returned an error. |
125 // | 129 // |
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276 Clock* clock_; // TODO(henrik.lundin) Make const if possible. | 280 Clock* clock_; // TODO(henrik.lundin) Make const if possible. |
277 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); | 281 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); |
278 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); | 282 rtc::Optional<int> last_packet_sample_rate_hz_ GUARDED_BY(crit_sect_); |
279 }; | 283 }; |
280 | 284 |
281 } // namespace acm2 | 285 } // namespace acm2 |
282 | 286 |
283 } // namespace webrtc | 287 } // namespace webrtc |
284 | 288 |
285 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ | 289 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |
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