64c1e8cda5
All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined. Tests completed: 1. android standalone to android standalone 2. android standalone to chrome (with and without this change) 3. android on chrome BUG=4145 R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae Cr-Commit-Position: refs/heads/master@{#8905} Review URL: https://webrtc-codereview.appspot.com/47399004 Cr-Commit-Position: refs/heads/master@{#8917}
128 lines
4.5 KiB
C++
128 lines
4.5 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 "webrtc/modules/rtp_rtcp/interface/rtp_cvo.h"
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#include "webrtc/modules/rtp_rtcp/interface/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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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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#include "webrtc/system_wrappers/interface/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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}
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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 char payload_name[RTP_PAYLOAD_NAME_SIZE],
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int8_t payload_type,
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uint32_t frequency) {
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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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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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// We are not allowed to hold a critical section when calling below functions.
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rtc::scoped_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.isFirstPacket = 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 = ConvertCVOByteToVideoRotation(
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rtp_header->header.extension.videoRotation);
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}
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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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int RTPReceiverVideo::GetPayloadTypeFrequency() const {
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return kVideoPayloadTypeFrequency;
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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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int32_t id,
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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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// For video we just go with default values.
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if (-1 ==
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callback->OnInitializeDecoder(
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id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
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LOG(LS_ERROR) << "Failed to created decoder for payload type: "
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<< static_cast<int>(payload_type);
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return -1;
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}
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return 0;
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}
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} // namespace webrtc
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