3ed46bd83b
Bug: webrtc:7135 Change-Id: Ic20d98cbfb8154f5abbc2501cbcccb950148e732 Reviewed-on: https://webrtc-review.googlesource.com/92396 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24219}
105 lines
3.7 KiB
C++
105 lines
3.7 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/source/rtp_receiver_video.h"
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#include <assert.h>
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#include <string.h>
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#include <memory>
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#include "modules/rtp_rtcp/include/rtp_cvo.h"
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#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
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#include "modules/rtp_rtcp/source/rtp_format.h"
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#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include "modules/rtp_rtcp/source/rtp_utility.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy(
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RtpData* data_callback) {
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return new RTPReceiverVideo(data_callback);
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}
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RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
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: RTPReceiverStrategy(data_callback) {}
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RTPReceiverVideo::~RTPReceiverVideo() {}
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int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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const uint8_t* payload,
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size_t payload_length,
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int64_t timestamp_ms) {
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rtp_header->video_header().codec =
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specific_payload.video_payload().videoCodecType;
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RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
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const size_t payload_data_length =
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payload_length - rtp_header->header.paddingLength;
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if (payload == NULL || payload_data_length == 0) {
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return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
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: -1;
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}
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if (first_packet_received_()) {
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RTC_LOG(LS_INFO) << "Received first video RTP packet";
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}
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// We are not allowed to hold a critical section when calling below functions.
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std::unique_ptr<RtpDepacketizer> depacketizer(
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RtpDepacketizer::Create(rtp_header->video_header().codec));
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if (depacketizer.get() == NULL) {
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RTC_LOG(LS_ERROR) << "Failed to create depacketizer.";
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return -1;
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}
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RtpDepacketizer::ParsedPayload parsed_payload;
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if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
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return -1;
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rtp_header->frameType = parsed_payload.frame_type;
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rtp_header->video_header() = parsed_payload.video_header();
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rtp_header->video_header().rotation = kVideoRotation_0;
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rtp_header->video_header().content_type = VideoContentType::UNSPECIFIED;
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rtp_header->video_header().video_timing.flags = VideoSendTiming::kInvalid;
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// Retrieve the video rotation information.
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if (rtp_header->header.extension.hasVideoRotation) {
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rtp_header->video_header().rotation =
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rtp_header->header.extension.videoRotation;
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}
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if (rtp_header->header.extension.hasVideoContentType) {
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rtp_header->video_header().content_type =
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rtp_header->header.extension.videoContentType;
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}
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if (rtp_header->header.extension.has_video_timing) {
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rtp_header->video_header().video_timing =
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rtp_header->header.extension.video_timing;
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}
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rtp_header->video_header().playout_delay =
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rtp_header->header.extension.playout_delay;
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return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
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parsed_payload.payload_length,
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rtp_header) == 0
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? 0
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: -1;
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}
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} // namespace webrtc
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