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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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130 // Get current receive frequency. | 130 // Get current receive frequency. |
131 int ReceiveFrequency() const override; | 131 int ReceiveFrequency() const override; |
132 | 132 |
133 // Get current playout frequency. | 133 // Get current playout frequency. |
134 int PlayoutFrequency() const override; | 134 int PlayoutFrequency() const override; |
135 | 135 |
136 // Register possible receive codecs, can be called multiple times, | 136 // Register possible receive codecs, can be called multiple times, |
137 // for codecs, CNG, DTMF, RED. | 137 // for codecs, CNG, DTMF, RED. |
138 int RegisterReceiveCodec(const CodecInst& receive_codec) override; | 138 int RegisterReceiveCodec(const CodecInst& receive_codec) override; |
139 | 139 |
| 140 int RegisterExternalReceiveCodec(int rtp_payload_type, |
| 141 AudioDecoder* external_decoder, |
| 142 int sample_rate_hz, |
| 143 int num_channels) override; |
| 144 |
140 // Get current received codec. | 145 // Get current received codec. |
141 int ReceiveCodec(CodecInst* current_codec) const override; | 146 int ReceiveCodec(CodecInst* current_codec) const override; |
142 | 147 |
143 // Incoming packet from network parsed and ready for decode. | 148 // Incoming packet from network parsed and ready for decode. |
144 int IncomingPacket(const uint8_t* incoming_payload, | 149 int IncomingPacket(const uint8_t* incoming_payload, |
145 const size_t payload_length, | 150 const size_t payload_length, |
146 const WebRtcRTPHeader& rtp_info) override; | 151 const WebRtcRTPHeader& rtp_info) override; |
147 | 152 |
148 // Incoming payloads, without rtp-info, the rtp-info will be created in ACM. | 153 // Incoming payloads, without rtp-info, the rtp-info will be created in ACM. |
149 // One usage for this API is when pre-encoded files are pushed in ACM. | 154 // One usage for this API is when pre-encoded files are pushed in ACM. |
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400 int playout_frequency_hz_; | 405 int playout_frequency_hz_; |
401 // TODO(henrik.lundin): All members below this line are temporary and should | 406 // TODO(henrik.lundin): All members below this line are temporary and should |
402 // be removed after refactoring is completed. | 407 // be removed after refactoring is completed. |
403 rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_; | 408 rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_; |
404 CodecInst current_send_codec_; | 409 CodecInst current_send_codec_; |
405 }; | 410 }; |
406 | 411 |
407 } // namespace webrtc | 412 } // namespace webrtc |
408 | 413 |
409 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ | 414 #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_ |
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