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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.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
#include <queue>
#include <string>
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
namespace webrtc {
class RtpPacketizerH264 : public RtpPacketizer {
public:
// Initialize with payload from encoder.
// The payload_data must be exactly one encoded H264 frame.
RtpPacketizerH264(FrameType frame_type, size_t max_payload_len);
virtual ~RtpPacketizerH264();
void SetPayloadData(const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) override;
// Get the next payload with H264 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.
bool NextPacket(uint8_t* buffer,
size_t* bytes_to_send,
bool* last_packet) override;
ProtectionType GetProtectionType() override;
StorageType GetStorageType(uint32_t retransmission_settings) override;
std::string ToString() override;
private:
struct Packet {
Packet(size_t offset,
size_t size,
bool first_fragment,
bool last_fragment,
bool aggregated,
uint8_t header)
: offset(offset),
size(size),
first_fragment(first_fragment),
last_fragment(last_fragment),
aggregated(aggregated),
header(header) {}
size_t offset;
size_t size;
bool first_fragment;
bool last_fragment;
bool aggregated;
uint8_t header;
};
typedef std::queue<Packet> PacketQueue;
void GeneratePackets();
void PacketizeFuA(size_t fragment_offset, size_t fragment_length);
int PacketizeStapA(size_t fragment_index,
size_t fragment_offset,
size_t fragment_length);
void NextAggregatePacket(uint8_t* buffer, size_t* bytes_to_send);
void NextFragmentPacket(uint8_t* buffer, size_t* bytes_to_send);
const uint8_t* payload_data_;
size_t payload_size_;
const size_t max_payload_len_;
RTPFragmentationHeader fragmentation_;
PacketQueue packets_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
};
// Depacketizer for H264.
class RtpDepacketizerH264 : public RtpDepacketizer {
public:
virtual ~RtpDepacketizerH264() {}
bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) override;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
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