Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(412)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
13 13
14 #include "webrtc/base/scoped_ptr.h" 14 #include "webrtc/base/scoped_ptr.h"
15 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" 15 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 16 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
17 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" 17 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 18 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
19 #include "webrtc/typedefs.h" 19 #include "webrtc/typedefs.h"
20 20
21 namespace webrtc { 21 namespace webrtc {
22 22
23 class RtpReceiverImpl : public RtpReceiver { 23 class RtpReceiverImpl : public RtpReceiver {
24 public: 24 public:
25 // Callbacks passed in here may not be NULL (use Null Object callbacks if you 25 // Callbacks passed in here may not be NULL (use Null Object callbacks if you
26 // want callbacks to do nothing). This class takes ownership of the media 26 // want callbacks to do nothing). This class takes ownership of the media
27 // receiver but nothing else. 27 // receiver but nothing else.
28 RtpReceiverImpl(Clock* clock, 28 RtpReceiverImpl(Clock* clock,
29 RtpAudioFeedback* incoming_audio_messages_callback,
29 RtpFeedback* incoming_messages_callback, 30 RtpFeedback* incoming_messages_callback,
30 RTPPayloadRegistry* rtp_payload_registry, 31 RTPPayloadRegistry* rtp_payload_registry,
31 RTPReceiverStrategy* rtp_media_receiver); 32 RTPReceiverStrategy* rtp_media_receiver);
32 33
33 virtual ~RtpReceiverImpl(); 34 virtual ~RtpReceiverImpl();
34 35
35 int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE], 36 int32_t RegisterReceivePayload(const char payload_name[RTP_PAYLOAD_NAME_SIZE],
36 const int8_t payload_type, 37 const int8_t payload_type,
37 const uint32_t frequency, 38 const uint32_t frequency,
38 const size_t channels, 39 const size_t channels,
(...skipping 50 matching lines...) Expand 10 before | Expand all | Expand 10 after
89 uint32_t current_remote_csrc_[kRtpCsrcSize]; 90 uint32_t current_remote_csrc_[kRtpCsrcSize];
90 91
91 uint32_t last_received_timestamp_; 92 uint32_t last_received_timestamp_;
92 int64_t last_received_frame_time_ms_; 93 int64_t last_received_frame_time_ms_;
93 uint16_t last_received_sequence_number_; 94 uint16_t last_received_sequence_number_;
94 95
95 NACKMethod nack_method_; 96 NACKMethod nack_method_;
96 }; 97 };
97 } // namespace webrtc 98 } // namespace webrtc
98 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ 99 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698