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/*
* 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 "webrtc/modules/rtp_rtcp/interface/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/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/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 char payload_name[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_type,
uint32_t frequency) {
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("webrtc_rtp",
"Video::ParseRtp",
"seqnum",
rtp_header->header.sequenceNumber,
"timestamp",
rtp_header->header.timestamp);
rtp_header->type.Video.codec = specific_payload.Video.videoCodecType;
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;
}
// We are not allowed to hold a critical section when calling below functions.
scoped_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.isFirstPacket = 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;
return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
parsed_payload.payload_length,
rtp_header) == 0
? 0
: -1;
}
int RTPReceiverVideo::GetPayloadTypeFrequency() const {
return kVideoPayloadTypeFrequency;
}
RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
uint16_t last_payload_length) const {
return kRtpDead;
}
int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
RtpFeedback* callback,
int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const {
// For video we just go with default values.
if (-1 ==
callback->OnInitializeDecoder(
id, payload_type, payload_name, kVideoPayloadTypeFrequency, 1, 0)) {
LOG(LS_ERROR) << "Failed to created decoder for payload type: "
<< payload_type;
return -1;
}
return 0;
}
int32_t RTPReceiverVideo::BuildRTPheader(const WebRtcRTPHeader* rtp_header,
uint8_t* data_buffer) const {
data_buffer[0] = static_cast<uint8_t>(0x80); // version 2
data_buffer[1] = static_cast<uint8_t>(rtp_header->header.payloadType);
if (rtp_header->header.markerBit) {
data_buffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1
}
RtpUtility::AssignUWord16ToBuffer(data_buffer + 2,
rtp_header->header.sequenceNumber);
RtpUtility::AssignUWord32ToBuffer(data_buffer + 4,
rtp_header->header.timestamp);
RtpUtility::AssignUWord32ToBuffer(data_buffer + 8, rtp_header->header.ssrc);
int32_t rtp_header_length = 12;
// Add the CSRCs if any
if (rtp_header->header.numCSRCs > 0) {
if (rtp_header->header.numCSRCs > 16) {
// error
assert(false);
}
uint8_t* ptr = &data_buffer[rtp_header_length];
for (uint32_t i = 0; i < rtp_header->header.numCSRCs; ++i) {
RtpUtility::AssignUWord32ToBuffer(ptr, rtp_header->header.arrOfCSRCs[i]);
ptr += 4;
}
data_buffer[0] = (data_buffer[0] & 0xf0) | rtp_header->header.numCSRCs;
// Update length of header
rtp_header_length += sizeof(uint32_t) * rtp_header->header.numCSRCs;
}
return rtp_header_length;
}
} // namespace webrtc
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