Line data    Source code

       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             : #ifdef WIN32
      17             : #include <winsock2.h>
      18             : #else
      19             : #include <arpa/inet.h>
      20             : #endif
      21             : 
      22             : #include <memory>
      23             : 
      24             : #include "webrtc/base/checks.h"
      25             : #include "webrtc/base/logging.h"
      26             : #include "webrtc/base/trace_event.h"
      27             : #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
      28             : #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
      29             : #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
      30             : #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
      31             : #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
      32             : #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
      33             : 
      34             : namespace webrtc {
      35             : 
      36           0 : RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy(
      37             :     RtpData* data_callback) {
      38           0 :   return new RTPReceiverVideo(data_callback);
      39             : }
      40             : 
      41           0 : RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
      42           0 :     : RTPReceiverStrategy(data_callback) {
      43           0 : }
      44             : 
      45           0 : RTPReceiverVideo::~RTPReceiverVideo() {
      46           0 : }
      47             : 
      48           0 : bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const {
      49             :   // Always do this for video packets.
      50           0 :   return true;
      51             : }
      52             : 
      53           0 : int32_t RTPReceiverVideo::OnNewPayloadTypeCreated(
      54             :     const CodecInst& audio_codec) {
      55           0 :   RTC_NOTREACHED();
      56           0 :   return 0;
      57             : }
      58             : 
      59           0 : int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
      60             :                                          const PayloadUnion& specific_payload,
      61             :                                          bool is_red,
      62             :                                          const uint8_t* payload,
      63             :                                          size_t payload_length,
      64             :                                          int64_t timestamp_ms,
      65             :                                          bool is_first_packet) {
      66           0 :   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
      67             :                "seqnum", rtp_header->header.sequenceNumber, "timestamp",
      68             :                rtp_header->header.timestamp);
      69           0 :   rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
      70             : 
      71           0 :   RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
      72             :   const size_t payload_data_length =
      73           0 :       payload_length - rtp_header->header.paddingLength;
      74             : 
      75           0 :   if (payload == NULL || payload_data_length == 0) {
      76           0 :     return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
      77           0 :                                                                            : -1;
      78             :   }
      79             : 
      80           0 :   if (first_packet_received_()) {
      81           0 :     LOG(LS_INFO) << "Received first video RTP packet";
      82             :   }
      83             : 
      84             :   // We are not allowed to hold a critical section when calling below functions.
      85             :   std::unique_ptr<RtpDepacketizer> depacketizer(
      86           0 :       RtpDepacketizer::Create(rtp_header->type.Video.codec));
      87           0 :   if (depacketizer.get() == NULL) {
      88           0 :     LOG(LS_ERROR) << "Failed to create depacketizer.";
      89           0 :     return -1;
      90             :   }
      91             : 
      92           0 :   rtp_header->type.Video.is_first_packet_in_frame = is_first_packet;
      93             :   RtpDepacketizer::ParsedPayload parsed_payload;
      94           0 :   if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
      95           0 :     return -1;
      96             : 
      97           0 :   rtp_header->frameType = parsed_payload.frame_type;
      98           0 :   rtp_header->type = parsed_payload.type;
      99           0 :   rtp_header->type.Video.rotation = kVideoRotation_0;
     100             : 
     101             :   // Retrieve the video rotation information.
     102           0 :   if (rtp_header->header.extension.hasVideoRotation) {
     103           0 :     rtp_header->type.Video.rotation =
     104           0 :         rtp_header->header.extension.videoRotation;
     105             :   }
     106             : 
     107           0 :   rtp_header->type.Video.playout_delay =
     108             :       rtp_header->header.extension.playout_delay;
     109             : 
     110           0 :   return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
     111             :                                                parsed_payload.payload_length,
     112           0 :                                                rtp_header) == 0
     113           0 :              ? 0
     114           0 :              : -1;
     115             : }
     116             : 
     117           0 : RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
     118             :     uint16_t last_payload_length) const {
     119           0 :   return kRtpDead;
     120             : }
     121             : 
     122           0 : int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
     123             :     RtpFeedback* callback,
     124             :     int8_t payload_type,
     125             :     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
     126             :     const PayloadUnion& specific_payload) const {
     127             :   // TODO(pbos): Remove as soon as audio can handle a changing payload type
     128             :   // without this callback.
     129           0 :   return 0;
     130             : }
     131             : 
     132             : }  // namespace webrtc