blob: 845e7ac496ccb13e124c71eabba7a8844a2f3914 [file] [log] [blame]
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h"
#include <assert.h>
#include <string.h>
#include <memory>
#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/trace_event.h"
namespace webrtc {
RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy(
RtpData* data_callback) {
return new RTPReceiverVideo(data_callback);
}
RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
: RTPReceiverStrategy(data_callback) {
}
RTPReceiverVideo::~RTPReceiverVideo() {
}
bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const {
// Always do this for video packets.
return true;
}
int32_t RTPReceiverVideo::OnNewPayloadTypeCreated(
const CodecInst& audio_codec) {
RTC_NOTREACHED();
return 0;
}
int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
bool is_red,
const uint8_t* payload,
size_t payload_length,
int64_t timestamp_ms,
bool is_first_packet) {
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
"seqnum", rtp_header->header.sequenceNumber, "timestamp",
rtp_header->header.timestamp);
rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
const size_t payload_data_length =
payload_length - rtp_header->header.paddingLength;
if (payload == NULL || payload_data_length == 0) {
return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0
: -1;
}
if (first_packet_received_()) {
LOG(LS_INFO) << "Received first video RTP packet";
}
// We are not allowed to hold a critical section when calling below functions.
std::unique_ptr<RtpDepacketizer> depacketizer(
RtpDepacketizer::Create(rtp_header->type.Video.codec));
if (depacketizer.get() == NULL) {
LOG(LS_ERROR) << "Failed to create depacketizer.";
return -1;
}
rtp_header->type.Video.is_first_packet_in_frame = is_first_packet;
RtpDepacketizer::ParsedPayload parsed_payload;
if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
return -1;
rtp_header->frameType = parsed_payload.frame_type;
rtp_header->type = parsed_payload.type;
rtp_header->type.Video.rotation = kVideoRotation_0;
rtp_header->type.Video.content_type = VideoContentType::UNSPECIFIED;
rtp_header->type.Video.video_timing.is_timing_frame = false;
// Retrieve the video rotation information.
if (rtp_header->header.extension.hasVideoRotation) {
rtp_header->type.Video.rotation =
rtp_header->header.extension.videoRotation;
}
if (rtp_header->header.extension.hasVideoContentType) {
rtp_header->type.Video.content_type =
rtp_header->header.extension.videoContentType;
}
if (rtp_header->header.extension.has_video_timing) {
rtp_header->type.Video.video_timing =
rtp_header->header.extension.video_timing;
rtp_header->type.Video.video_timing.is_timing_frame = true;
}
rtp_header->type.Video.playout_delay =
rtp_header->header.extension.playout_delay;
return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
parsed_payload.payload_length,
rtp_header) == 0
? 0
: -1;
}
RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
uint16_t last_payload_length) const {
return kRtpDead;
}
int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
RtpFeedback* callback,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const {
// TODO(pbos): Remove as soon as audio can handle a changing payload type
// without this callback.
return 0;
}
} // namespace webrtc