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/*
* Copyright (c) 2015 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.
*/
//
// This file contains the declaration of the VP9 packetizer class.
// A packetizer object is created for each encoded video frame. The
// constructor is called with the payload data and size.
//
// After creating the packetizer, the method NextPacket is called
// repeatedly to get all packets for the frame. The method returns
// false as long as there are more packets left to fetch.
//
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
#include <queue>
#include <string>
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RtpPacketizerVp9 : public RtpPacketizer {
public:
RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr, size_t max_payload_length);
virtual ~RtpPacketizerVp9();
ProtectionType GetProtectionType() override;
StorageType GetStorageType(uint32_t retransmission_settings) override;
std::string ToString() override;
// The payload data must be one encoded VP9 frame.
void SetPayloadData(const uint8_t* payload,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) override;
// Gets the next payload with VP9 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.
// |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, false otherwise.
bool NextPacket(uint8_t* buffer,
size_t* bytes_to_send,
bool* last_packet) override;
typedef struct {
size_t payload_start_pos;
size_t size;
bool layer_begin;
bool layer_end;
} PacketInfo;
typedef std::queue<PacketInfo> PacketInfoQueue;
private:
// Calculates all packet sizes and loads info to packet queue.
void GeneratePackets();
// Writes the payload descriptor header and copies payload to the |buffer|.
// |packet_info| determines which part of the payload to write.
// |bytes_to_send| contains the number of written bytes to the buffer.
// Returns true on success, false otherwise.
bool WriteHeaderAndPayload(const PacketInfo& packet_info,
uint8_t* buffer,
size_t* bytes_to_send) const;
// Writes payload descriptor header to |buffer|.
// Returns true on success, false otherwise.
bool WriteHeader(const PacketInfo& packet_info,
uint8_t* buffer,
size_t* header_length) const;
const RTPVideoHeaderVP9 hdr_;
const size_t max_payload_length_; // The max length in bytes of one packet.
const uint8_t* payload_; // The payload data to be packetized.
size_t payload_size_; // The size in bytes of the payload data.
PacketInfoQueue packets_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9);
};
class RtpDepacketizerVp9 : public RtpDepacketizer {
public:
virtual ~RtpDepacketizerVp9() {}
bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload,
size_t payload_length) override;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
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