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Side by Side Diff: webrtc/modules/video_coding/frame_object.cc

Issue 1969403007: FrameBuffer for the new jitter buffer. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Unittest fix and feedback fixes. Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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/video_coding/frame_object.h" 11 #include "webrtc/modules/video_coding/frame_object.h"
12 #include "webrtc/base/criticalsection.h" 12 #include "webrtc/base/criticalsection.h"
13 #include "webrtc/modules/video_coding/packet_buffer.h" 13 #include "webrtc/modules/video_coding/packet_buffer.h"
14 14
15 namespace webrtc { 15 namespace webrtc {
16 namespace video_coding { 16 namespace video_coding {
17 17
18 FrameObject::FrameObject() 18 FrameObject::FrameObject()
19 : picture_id(0), 19 : picture_id(0),
20 spatial_layer(0), 20 spatial_layer(0),
21 timestamp(0),
21 num_references(0), 22 num_references(0),
22 inter_layer_predicted(false) {} 23 inter_layer_predicted(false) {}
23 24
24 RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer, 25 RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
25 uint16_t first_packet, 26 uint16_t first_packet,
26 uint16_t last_packet) 27 uint16_t last_packet)
27 : packet_buffer_(packet_buffer), 28 : packet_buffer_(packet_buffer),
28 first_packet_(first_packet), 29 first_packet_(first_packet),
29 last_packet_(last_packet) {} 30 last_packet_(last_packet) {}
30 31
31 RtpFrameObject::~RtpFrameObject() { 32 RtpFrameObject::~RtpFrameObject() {
32 packet_buffer_->ReturnFrame(this); 33 packet_buffer_->ReturnFrame(this);
33 } 34 }
34 35
35 uint16_t RtpFrameObject::first_seq_num() const { 36 uint16_t RtpFrameObject::first_seq_num() const {
36 return first_packet_; 37 return first_packet_;
37 } 38 }
38 39
39 uint16_t RtpFrameObject::last_seq_num() const { 40 uint16_t RtpFrameObject::last_seq_num() const {
40 return last_packet_; 41 return last_packet_;
41 } 42 }
42 43
43 bool RtpFrameObject::GetBitstream(uint8_t* destination) const { 44 bool RtpFrameObject::GetBitstream(uint8_t* destination) const {
44 return packet_buffer_->GetBitstream(*this, destination); 45 return packet_buffer_->GetBitstream(*this, destination);
45 } 46 }
46 47
47 } // namespace video_coding 48 } // namespace video_coding
48 } // namespace webrtc 49 } // namespace webrtc
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