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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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