OLD | NEW |
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 <string> |
15 #include <vector> | 16 #include <vector> |
16 | 17 |
17 #include "webrtc/base/array_view.h" | 18 #include "webrtc/base/array_view.h" |
18 #include "webrtc/base/optional.h" | 19 #include "webrtc/base/optional.h" |
19 #include "webrtc/base/scoped_ptr.h" | 20 #include "webrtc/base/scoped_ptr.h" |
20 #include "webrtc/base/thread_annotations.h" | 21 #include "webrtc/base/thread_annotations.h" |
21 #include "webrtc/common_audio/vad/include/webrtc_vad.h" | 22 #include "webrtc/common_audio/vad/include/webrtc_vad.h" |
22 #include "webrtc/engine_configurations.h" | 23 #include "webrtc/engine_configurations.h" |
23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 24 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
24 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" | 25 #include "webrtc/modules/audio_coding/acm2/acm_resampler.h" |
(...skipping 85 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
110 // (e.g. iSAC, where the decoder needs to be paired | 111 // (e.g. iSAC, where the decoder needs to be paired |
111 // with an encoder). | 112 // with an encoder). |
112 // | 113 // |
113 // Return value : 0 if OK. | 114 // Return value : 0 if OK. |
114 // <0 if NetEq returned an error. | 115 // <0 if NetEq returned an error. |
115 // | 116 // |
116 int AddCodec(int acm_codec_id, | 117 int AddCodec(int acm_codec_id, |
117 uint8_t payload_type, | 118 uint8_t payload_type, |
118 int channels, | 119 int channels, |
119 int sample_rate_hz, | 120 int sample_rate_hz, |
120 AudioDecoder* audio_decoder); | 121 AudioDecoder* audio_decoder, |
| 122 const std::string& name); |
121 | 123 |
122 // | 124 // |
123 // Sets a minimum delay for packet buffer. The given delay is maintained, | 125 // Sets a minimum delay for packet buffer. The given delay is maintained, |
124 // unless channel condition dictates a higher delay. | 126 // unless channel condition dictates a higher delay. |
125 // | 127 // |
126 // Input: | 128 // Input: |
127 // - delay_ms : minimum delay in milliseconds. | 129 // - delay_ms : minimum delay in milliseconds. |
128 // | 130 // |
129 // Return value : 0 if OK. | 131 // Return value : 0 if OK. |
130 // <0 if NetEq returned an error. | 132 // <0 if NetEq returned an error. |
(...skipping 165 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
296 Clock* clock_; // TODO(henrik.lundin) Make const if possible. | 298 Clock* clock_; // TODO(henrik.lundin) Make const if possible. |
297 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); | 299 bool resampled_last_output_frame_ GUARDED_BY(crit_sect_); |
298 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_); |
299 }; | 301 }; |
300 | 302 |
301 } // namespace acm2 | 303 } // namespace acm2 |
302 | 304 |
303 } // namespace webrtc | 305 } // namespace webrtc |
304 | 306 |
305 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ | 307 #endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_ |
OLD | NEW |