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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h

Issue 1812453002: Revert of Clean away use of RtpAudioFeedback interface from RTP/RTCP receiver code. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@voe_dtmf_7
Patch Set: Created 4 years, 9 months ago
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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 explicit RTPReceiverAudio(RtpData* data_callback); 31 RTPReceiverAudio(RtpData* data_callback,
32 RtpAudioFeedback* incoming_messages_callback);
32 virtual ~RTPReceiverAudio() {} 33 virtual ~RTPReceiverAudio() {}
33 34
34 // The following three methods implement the TelephoneEventHandler interface. 35 // The following three methods implement the TelephoneEventHandler interface.
35 // Forward DTMFs to decoder for playout. 36 // Forward DTMFs to decoder for playout.
36 void SetTelephoneEventForwardToDecoder(bool forward_to_decoder); 37 void SetTelephoneEventForwardToDecoder(bool forward_to_decoder);
37 38
38 // Is forwarding of outband telephone events turned on/off? 39 // Is forwarding of outband telephone events turned on/off?
39 bool TelephoneEventForwardToDecoder() const; 40 bool TelephoneEventForwardToDecoder() const;
40 41
41 // Is TelephoneEvent configured with payload type payload_type 42 // Is TelephoneEvent configured with payload type payload_type
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111 int8_t cng_fb_payload_type_; 112 int8_t cng_fb_payload_type_;
112 int8_t cng_payload_type_; 113 int8_t cng_payload_type_;
113 114
114 // G722 is special since it use the wrong number of RTP samples in timestamp 115 // G722 is special since it use the wrong number of RTP samples in timestamp
115 // VS. number of samples in the frame 116 // VS. number of samples in the frame
116 int8_t g722_payload_type_; 117 int8_t g722_payload_type_;
117 bool last_received_g722_; 118 bool last_received_g722_;
118 119
119 uint8_t num_energy_; 120 uint8_t num_energy_;
120 uint8_t current_remote_energy_[kRtpCsrcSize]; 121 uint8_t current_remote_energy_[kRtpCsrcSize];
122
123 RtpAudioFeedback* cb_audio_feedback_;
121 }; 124 };
122 } // namespace webrtc 125 } // namespace webrtc
123 126
124 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_ 127 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
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