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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 "modules/audio_coding/neteq/tools/packet.h"
#include "api/array_view.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
namespace test {
Packet::Packet(rtc::CopyOnWriteBuffer packet,
size_t virtual_packet_length_bytes,
double time_ms,
const RtpHeaderExtensionMap* extension_map)
: packet_(std::move(packet)),
virtual_packet_length_bytes_(virtual_packet_length_bytes),
time_ms_(time_ms),
valid_header_(ParseHeader(extension_map)) {}
Packet::Packet(const RTPHeader& header,
size_t virtual_packet_length_bytes,
size_t virtual_payload_length_bytes,
double time_ms)
: header_(header),
virtual_packet_length_bytes_(virtual_packet_length_bytes),
virtual_payload_length_bytes_(virtual_payload_length_bytes),
time_ms_(time_ms),
valid_header_(true) {}
Packet::~Packet() = default;
bool Packet::ExtractRedHeaders(std::list<RTPHeader*>* headers) const {
//
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |1| block PT | timestamp offset | block length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |1| ... |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |0| block PT |
// +-+-+-+-+-+-+-+-+
//
const uint8_t* payload_ptr = payload();
const uint8_t* payload_end_ptr = payload_ptr + payload_length_bytes();
// Find all RED headers with the extension bit set to 1. That is, all headers
// but the last one.
while ((payload_ptr < payload_end_ptr) && (*payload_ptr & 0x80)) {
RTPHeader* header = new RTPHeader;
CopyToHeader(header);
header->payloadType = payload_ptr[0] & 0x7F;
uint32_t offset = (payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2);
header->timestamp -= offset;
headers->push_front(header);
payload_ptr += 4;
}
// Last header.
RTC_DCHECK_LT(payload_ptr, payload_end_ptr);
if (payload_ptr >= payload_end_ptr) {
return false; // Payload too short.
}
RTPHeader* header = new RTPHeader;
CopyToHeader(header);
header->payloadType = payload_ptr[0] & 0x7F;
headers->push_front(header);
return true;
}
void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) {
while (!headers->empty()) {
delete headers->front();
headers->pop_front();
}
}
bool Packet::ParseHeader(const RtpHeaderExtensionMap* extension_map) {
// Use RtpPacketReceived instead of RtpPacket because former already has a
// converter into legacy RTPHeader.
webrtc::RtpPacketReceived rtp_packet(extension_map);
// Because of the special case of dummy packets that have padding marked in
// the RTP header, but do not have rtp payload with the padding size, handle
// padding manually. Regular RTP packet parser reports failure, but it is fine
// in this context.
bool padding = (packet_[0] & 0b0010'0000);
size_t padding_size = 0;
if (padding) {
// Clear the padding bit to prevent failure when rtp payload is omited.
rtc::CopyOnWriteBuffer packet(packet_);
packet.MutableData()[0] &= ~0b0010'0000;
if (!rtp_packet.Parse(std::move(packet))) {
return false;
}
if (rtp_packet.payload_size() > 0) {
padding_size = rtp_packet.data()[rtp_packet.size() - 1];
}
if (padding_size > rtp_packet.payload_size()) {
return false;
}
} else {
if (!rtp_packet.Parse(packet_)) {
return false;
}
}
rtp_payload_ = rtc::MakeArrayView(packet_.data() + rtp_packet.headers_size(),
rtp_packet.payload_size() - padding_size);
rtp_packet.GetHeader(&header_);
RTC_CHECK_GE(virtual_packet_length_bytes_, rtp_packet.size());
RTC_DCHECK_GE(virtual_packet_length_bytes_, rtp_packet.headers_size());
virtual_payload_length_bytes_ =
virtual_packet_length_bytes_ - rtp_packet.headers_size();
return true;
}
void Packet::CopyToHeader(RTPHeader* destination) const {
*destination = header_;
}
} // namespace test
} // namespace webrtc
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