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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

Issue 1609563003: Remove two dead 'using' instances. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 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
11 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h" 11 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
12 12
13 #include <assert.h> 13 #include <assert.h>
14 #include <math.h> 14 #include <math.h>
15 #include <stdlib.h> 15 #include <stdlib.h>
16 #include <string.h> 16 #include <string.h>
17 17
18 #include "webrtc/base/logging.h" 18 #include "webrtc/base/logging.h"
19 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" 19 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
21 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" 21 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
22 22
23 namespace webrtc { 23 namespace webrtc {
24 24
25 using RtpUtility::Payload; 25 using RtpUtility::Payload;
26 using RtpUtility::StringCompare;
27 26
28 RtpReceiver* RtpReceiver::CreateVideoReceiver( 27 RtpReceiver* RtpReceiver::CreateVideoReceiver(
29 Clock* clock, 28 Clock* clock,
30 RtpData* incoming_payload_callback, 29 RtpData* incoming_payload_callback,
31 RtpFeedback* incoming_messages_callback, 30 RtpFeedback* incoming_messages_callback,
32 RTPPayloadRegistry* rtp_payload_registry) { 31 RTPPayloadRegistry* rtp_payload_registry) {
33 if (!incoming_payload_callback) 32 if (!incoming_payload_callback)
34 incoming_payload_callback = NullObjectRtpData(); 33 incoming_payload_callback = NullObjectRtpData();
35 if (!incoming_messages_callback) 34 if (!incoming_messages_callback)
36 incoming_messages_callback = NullObjectRtpFeedback(); 35 incoming_messages_callback = NullObjectRtpFeedback();
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479 // implementations might have CSRC 0 as a valid value. 478 // implementations might have CSRC 0 as a valid value.
480 if (num_csrcs_diff > 0) { 479 if (num_csrcs_diff > 0) {
481 cb_rtp_feedback_->OnIncomingCSRCChanged(0, true); 480 cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
482 } else if (num_csrcs_diff < 0) { 481 } else if (num_csrcs_diff < 0) {
483 cb_rtp_feedback_->OnIncomingCSRCChanged(0, false); 482 cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
484 } 483 }
485 } 484 }
486 } 485 }
487 486
488 } // namespace webrtc 487 } // namespace webrtc
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