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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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