Home | History | Annotate | Download | only in source 
      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 
     28 RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy(
     29     RtpData* data_callback) {
     30   return new RTPReceiverVideo(data_callback);
     31 }
     32 
     33 RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
     34     : RTPReceiverStrategy(data_callback) {
     35 }
     36 
     37 RTPReceiverVideo::~RTPReceiverVideo() {
     38 }
     39 
     40 bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const {
     41   // Always do this for video packets.
     42   return true;
     43 }
     44 
     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 
     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 
    103 int RTPReceiverVideo::GetPayloadTypeFrequency() const {
    104   return kVideoPayloadTypeFrequency;
    105 }
    106 
    107 RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
    108     uint16_t last_payload_length) const {
    109   return kRtpDead;
    110 }
    111 
    112 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