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Side by Side Diff: webrtc/modules/video_coding/main/source/jitter_buffer.cc

Issue 1417683006: modules: more interface -> include renames (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix last incorrectly wrapped pragma message Created 5 years, 1 month 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 #include "webrtc/modules/video_coding/main/source/jitter_buffer.h" 10 #include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
11 11
12 #include <assert.h> 12 #include <assert.h>
13 13
14 #include <algorithm> 14 #include <algorithm>
15 #include <utility> 15 #include <utility>
16 16
17 #include "webrtc/base/checks.h" 17 #include "webrtc/base/checks.h"
18 #include "webrtc/base/trace_event.h" 18 #include "webrtc/base/trace_event.h"
19 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
20 #include "webrtc/modules/video_coding/main/interface/video_coding.h" 20 #include "webrtc/modules/video_coding/main/interface/video_coding.h"
21 #include "webrtc/modules/video_coding/main/source/frame_buffer.h" 21 #include "webrtc/modules/video_coding/main/source/frame_buffer.h"
22 #include "webrtc/modules/video_coding/main/source/inter_frame_delay.h" 22 #include "webrtc/modules/video_coding/main/source/inter_frame_delay.h"
23 #include "webrtc/modules/video_coding/main/source/internal_defines.h" 23 #include "webrtc/modules/video_coding/main/source/internal_defines.h"
24 #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h" 24 #include "webrtc/modules/video_coding/main/source/jitter_buffer_common.h"
25 #include "webrtc/modules/video_coding/main/source/jitter_estimator.h" 25 #include "webrtc/modules/video_coding/main/source/jitter_estimator.h"
26 #include "webrtc/modules/video_coding/main/source/packet.h" 26 #include "webrtc/modules/video_coding/main/source/packet.h"
27 #include "webrtc/system_wrappers/include/clock.h" 27 #include "webrtc/system_wrappers/include/clock.h"
28 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" 28 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
29 #include "webrtc/system_wrappers/include/event_wrapper.h" 29 #include "webrtc/system_wrappers/include/event_wrapper.h"
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1330 } 1330 }
1331 // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in 1331 // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
1332 // that case we don't wait for retransmissions. 1332 // that case we don't wait for retransmissions.
1333 if (high_rtt_nack_threshold_ms_ >= 0 && 1333 if (high_rtt_nack_threshold_ms_ >= 0 &&
1334 rtt_ms_ >= high_rtt_nack_threshold_ms_) { 1334 rtt_ms_ >= high_rtt_nack_threshold_ms_) {
1335 return false; 1335 return false;
1336 } 1336 }
1337 return true; 1337 return true;
1338 } 1338 }
1339 } // namespace webrtc 1339 } // namespace webrtc
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