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/*
* Copyright (c) 2014 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 <string>
#include "webrtc/base/logging.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
namespace webrtc {
static const size_t kGenericHeaderLength = 1;
RtpPacketizerGeneric::RtpPacketizerGeneric(FrameType frame_type,
size_t max_payload_len)
: payload_data_(NULL),
payload_size_(0),
max_payload_len_(max_payload_len - kGenericHeaderLength),
frame_type_(frame_type) {
}
RtpPacketizerGeneric::~RtpPacketizerGeneric() {
}
void RtpPacketizerGeneric::SetPayloadData(
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
payload_data_ = payload_data;
payload_size_ = payload_size;
// Fragment packets more evenly by splitting the payload up evenly.
size_t num_packets =
(payload_size_ + max_payload_len_ - 1) / max_payload_len_;
payload_length_ = (payload_size_ + num_packets - 1) / num_packets;
assert(payload_length_ <= max_payload_len_);
generic_header_ = RtpFormatVideoGeneric::kFirstPacketBit;
}
bool RtpPacketizerGeneric::NextPacket(uint8_t* buffer,
size_t* bytes_to_send,
bool* last_packet) {
if (payload_size_ < payload_length_) {
payload_length_ = payload_size_;
}
payload_size_ -= payload_length_;
*bytes_to_send = payload_length_ + kGenericHeaderLength;
assert(payload_length_ <= max_payload_len_);
uint8_t* out_ptr = buffer;
// Put generic header in packet
if (frame_type_ == kVideoFrameKey) {
generic_header_ |= RtpFormatVideoGeneric::kKeyFrameBit;
}
*out_ptr++ = generic_header_;
// Remove first-packet bit, following packets are intermediate
generic_header_ &= ~RtpFormatVideoGeneric::kFirstPacketBit;
// Put payload in packet
memcpy(out_ptr, payload_data_, payload_length_);
payload_data_ += payload_length_;
*last_packet = payload_size_ <= 0;
return true;
}
ProtectionType RtpPacketizerGeneric::GetProtectionType() {
return kProtectedPacket;
}
StorageType RtpPacketizerGeneric::GetStorageType(
uint32_t retransmission_settings) {
return kAllowRetransmission;
}
std::string RtpPacketizerGeneric::ToString() {
return "RtpPacketizerGeneric";
}
bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) {
assert(parsed_payload != NULL);
if (payload_data_length == 0) {
LOG(LS_ERROR) << "Empty payload.";
return false;
}
uint8_t generic_header = *payload_data++;
--payload_data_length;
parsed_payload->frame_type =
((generic_header & RtpFormatVideoGeneric::kKeyFrameBit) != 0)
? kVideoFrameKey
: kVideoFrameDelta;
parsed_payload->type.Video.isFirstPacket =
(generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
parsed_payload->type.Video.codec = kRtpVideoGeneric;
parsed_payload->type.Video.width = 0;
parsed_payload->type.Video.height = 0;
parsed_payload->payload = payload_data;
parsed_payload->payload_length = payload_data_length;
return true;
}
} // namespace webrtc
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