0d1b2b6880
Add RTC_DEPRACATed anonymous unions to not break downstream projects.
Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date: Mon Dec 19 09:36:33 2016 -0800
>
> Revert of Rename RTPVideoHeader.isFirstPacket to
> .is_first_packet_in_frame. (patchset #1 id:1 of
> https://codereview.webrtc.org/2574943003/ )
>
> Reason for revert:
> breaks downstream project.
>
> Can you make this change in a compatible way using anonymous
> union:
> union {
> bool is_first_packet_in_frame;
> RTC_DEPRECATED bool isFirstPacket;
> };
> (unfortunetly this this treak breaks braced initialization in
> rtp_rtcp_impl_unittest.cc,
> so that should be rewritting in a more classic way)
>
> Original issue's description:
> > Rename RTPVideoHeader.isFirstPacket to
> > .is_first_packet_in_frame.
> >
> > Name should represent the actual meaning.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2574943003
> > Cr-Commit-Position: refs/heads/master@{#15684}
> > Committed:
> > efde908380
>
> TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days
> ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2589783003
> Cr-Commit-Position: refs/heads/master@{#15686}
>
BUG=None
Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
126 lines
4.3 KiB
C++
126 lines
4.3 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 "webrtc/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 "webrtc/base/checks.h"
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#include "webrtc/base/logging.h"
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#include "webrtc/base/trace_event.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.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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}
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RTPReceiverVideo::~RTPReceiverVideo() {
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}
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bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const {
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// Always do this for video packets.
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return true;
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}
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int32_t RTPReceiverVideo::OnNewPayloadTypeCreated(
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const CodecInst& audio_codec) {
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RTC_NOTREACHED();
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return 0;
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}
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int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
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const PayloadUnion& specific_payload,
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bool is_red,
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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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bool is_first_packet) {
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TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
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"seqnum", rtp_header->header.sequenceNumber, "timestamp",
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rtp_header->header.timestamp);
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rtp_header->type.Video.codec = specific_payload.Video.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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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->type.Video.codec));
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if (depacketizer.get() == NULL) {
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LOG(LS_ERROR) << "Failed to create depacketizer.";
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return -1;
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}
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rtp_header->type.Video.is_first_packet_in_frame = is_first_packet;
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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->type = parsed_payload.type;
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rtp_header->type.Video.rotation = kVideoRotation_0;
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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->type.Video.rotation =
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rtp_header->header.extension.videoRotation;
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}
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rtp_header->type.Video.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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RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
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uint16_t last_payload_length) const {
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return kRtpDead;
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}
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int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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int8_t payload_type,
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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const PayloadUnion& specific_payload) const {
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// TODO(pbos): Remove as soon as audio can handle a changing payload type
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// without this callback.
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return 0;
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}
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} // namespace webrtc
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