1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
12
13 #include <assert.h>
14 #include <string.h>
15
16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/logging.h"
18 #include "webrtc/base/trace_event.h"
19 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
20 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
21 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
22 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
23 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
24 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
25
26 namespace webrtc {
27
CreateVideoStrategy(RtpData * data_callback)28 RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy(
29 RtpData* data_callback) {
30 return new RTPReceiverVideo(data_callback);
31 }
32
RTPReceiverVideo(RtpData * data_callback)33 RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
34 : RTPReceiverStrategy(data_callback) {
35 }
36
~RTPReceiverVideo()37 RTPReceiverVideo::~RTPReceiverVideo() {
38 }
39
ShouldReportCsrcChanges(uint8_t payload_type) const40 bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const {
41 // Always do this for video packets.
42 return true;
43 }
44
OnNewPayloadTypeCreated(const char payload_name[RTP_PAYLOAD_NAME_SIZE],int8_t payload_type,uint32_t frequency)45 int32_t RTPReceiverVideo::OnNewPayloadTypeCreated(
46 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
47 int8_t payload_type,
48 uint32_t frequency) {
49 return 0;
50 }
51
ParseRtpPacket(WebRtcRTPHeader * rtp_header,const PayloadUnion & specific_payload,bool is_red,const uint8_t * payload,size_t payload_length,int64_t timestamp_ms,bool is_first_packet)52 int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
53 const PayloadUnion& specific_payload,
54 bool is_red,
55 const uint8_t* payload,
56 size_t payload_length,
57 int64_t timestamp_ms,
58 bool is_first_packet) {
59 TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
60 "seqnum", rtp_header->header.sequenceNumber, "timestamp",
61 rtp_header->header.timestamp);
62 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
63
64 RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
65 const size_t payload_data_length =
66 payload_length - rtp_header->header.paddingLength;
67
68 if (payload == NULL || payload_data_length == 0) {
69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
70 : -1;
71 }
72
73 // We are not allowed to hold a critical section when calling below functions.
74 rtc::scoped_ptr<RtpDepacketizer> depacketizer(
75 RtpDepacketizer::Create(rtp_header->type.Video.codec));
76 if (depacketizer.get() == NULL) {
77 LOG(LS_ERROR) << "Failed to create depacketizer.";
78 return -1;
79 }
80
81 rtp_header->type.Video.isFirstPacket = is_first_packet;
82 RtpDepacketizer::ParsedPayload parsed_payload;
83 if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
84 return -1;
85
86 rtp_header->frameType = parsed_payload.frame_type;
87 rtp_header->type = parsed_payload.type;
88 rtp_header->type.Video.rotation = kVideoRotation_0;
89
90 // Retrieve the video rotation information.
91 if (rtp_header->header.extension.hasVideoRotation) {
92 rtp_header->type.Video.rotation = ConvertCVOByteToVideoRotation(
93 rtp_header->header.extension.videoRotation);
94 }
95
96 return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
97 parsed_payload.payload_length,
98 rtp_header) == 0
99 ? 0
100 : -1;
101 }
102
GetPayloadTypeFrequency() const103 int RTPReceiverVideo::GetPayloadTypeFrequency() const {
104 return kVideoPayloadTypeFrequency;
105 }
106
ProcessDeadOrAlive(uint16_t last_payload_length) const107 RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
108 uint16_t last_payload_length) const {
109 return kRtpDead;
110 }
111
InvokeOnInitializeDecoder(RtpFeedback * callback,int8_t payload_type,const char payload_name[RTP_PAYLOAD_NAME_SIZE],const PayloadUnion & specific_payload) const112 int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
113 RtpFeedback* callback,
114 int8_t payload_type,
115 const char payload_name[RTP_PAYLOAD_NAME_SIZE],
116 const PayloadUnion& specific_payload) const {
117 // For video we just go with default values.
118 if (-1 ==
119 callback->OnInitializeDecoder(payload_type, payload_name,
120 kVideoPayloadTypeFrequency, 1, 0)) {
121 LOG(LS_ERROR) << "Failed to created decoder for payload type: "
122 << static_cast<int>(payload_type);
123 return -1;
124 }
125 return 0;
126 }
127
128 } // namespace webrtc
129