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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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| 21 #include "webrtc/typedefs.h" | 21 #include "webrtc/typedefs.h" |
| 22 | 22 |
| 23 namespace webrtc { | 23 namespace webrtc { |
| 24 | 24 |
| 25 class CriticalSectionWrapper; | 25 class CriticalSectionWrapper; |
| 26 | 26 |
| 27 // Handles audio RTP packets. This class is thread-safe. | 27 // Handles audio RTP packets. This class is thread-safe. |
| 28 class RTPReceiverAudio : public RTPReceiverStrategy, | 28 class RTPReceiverAudio : public RTPReceiverStrategy, |
| 29 public TelephoneEventHandler { | 29 public TelephoneEventHandler { |
| 30 public: | 30 public: |
| 31 RTPReceiverAudio(RtpData* data_callback, | 31 explicit RTPReceiverAudio(RtpData* data_callback); |
| 32 RtpAudioFeedback* incoming_messages_callback); | |
| 33 virtual ~RTPReceiverAudio() {} | 32 virtual ~RTPReceiverAudio() {} |
| 34 | 33 |
| 35 // The following three methods implement the TelephoneEventHandler interface. | 34 // The following three methods implement the TelephoneEventHandler interface. |
| 36 // Forward DTMFs to decoder for playout. | 35 // Forward DTMFs to decoder for playout. |
| 37 void SetTelephoneEventForwardToDecoder(bool forward_to_decoder); | 36 void SetTelephoneEventForwardToDecoder(bool forward_to_decoder); |
| 38 | 37 |
| 39 // Is forwarding of outband telephone events turned on/off? | 38 // Is forwarding of outband telephone events turned on/off? |
| 40 bool TelephoneEventForwardToDecoder() const; | 39 bool TelephoneEventForwardToDecoder() const; |
| 41 | 40 |
| 42 // Is TelephoneEvent configured with payload type payload_type | 41 // Is TelephoneEvent configured with payload type payload_type |
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| 112 int8_t cng_fb_payload_type_; | 111 int8_t cng_fb_payload_type_; |
| 113 int8_t cng_payload_type_; | 112 int8_t cng_payload_type_; |
| 114 | 113 |
| 115 // G722 is special since it use the wrong number of RTP samples in timestamp | 114 // G722 is special since it use the wrong number of RTP samples in timestamp |
| 116 // VS. number of samples in the frame | 115 // VS. number of samples in the frame |
| 117 int8_t g722_payload_type_; | 116 int8_t g722_payload_type_; |
| 118 bool last_received_g722_; | 117 bool last_received_g722_; |
| 119 | 118 |
| 120 uint8_t num_energy_; | 119 uint8_t num_energy_; |
| 121 uint8_t current_remote_energy_[kRtpCsrcSize]; | 120 uint8_t current_remote_energy_[kRtpCsrcSize]; |
| 122 | |
| 123 RtpAudioFeedback* cb_audio_feedback_; | |
| 124 }; | 121 }; |
| 125 } // namespace webrtc | 122 } // namespace webrtc |
| 126 | 123 |
| 127 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ | 124 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ |
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