blob: 8cad6a916d0531115049067dbb1d20eb2c57bd04 [file] [log] [blame]
* 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/constructormagic.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class RtpPacketizer {
static RtpPacketizer* Create(RtpVideoCodecTypes type,
size_t max_payload_len,
const RTPVideoTypeHeader* rtp_type_header,
FrameType frame_type);
virtual ~RtpPacketizer() {}
virtual void SetPayloadData(const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) = 0;
// Get the next payload with payload header.
// buffer is a pointer to where the output will be written.
// bytes_to_send is an output variable that will contain number of bytes
// written to buffer. The parameter last_packet is true for the last packet of
// the frame, false otherwise (i.e., call the function again to get the
// next packet).
// Returns true on success or false if there was no payload to packetize.
virtual bool NextPacket(uint8_t* buffer,
size_t* bytes_to_send,
bool* last_packet) = 0;
virtual ProtectionType GetProtectionType() = 0;
virtual StorageType GetStorageType(uint32_t retransmission_settings) = 0;
virtual std::string ToString() = 0;
// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
// of the parsed payload, rather than just a pointer into the incoming buffer.
// This way we can move some parsing out from the jitter buffer into here, and
// the jitter buffer can just store that pointer rather than doing a copy there.
class RtpDepacketizer {
struct ParsedPayload {
const uint8_t* payload;
size_t payload_length;
FrameType frame_type;
RTPTypeHeader type;
static RtpDepacketizer* Create(RtpVideoCodecTypes type);
virtual ~RtpDepacketizer() {}
// Parses the RTP payload, parsed result will be saved in |parsed_payload|.
virtual bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) = 0;
} // namespace webrtc