blob: e608db59211dee8853411330a48b0c5b04ddcac2 [file] [log] [blame]
/*
* Copyright (c) 2013 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 MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
#include <string>
#include "common_types.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/source/rtp_format.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace RtpFormatVideoGeneric {
static const uint8_t kKeyFrameBit = 0x01;
static const uint8_t kFirstPacketBit = 0x02;
// If this bit is set, there will be an extended header contained in this
// packet. This was added later so old clients will not send this.
static const uint8_t kExtendedHeaderBit = 0x04;
} // namespace RtpFormatVideoGeneric
class RtpPacketizerGeneric : public RtpPacketizer {
public:
// Initialize with payload from encoder.
// The payload_data must be exactly one encoded generic frame.
RtpPacketizerGeneric(const RTPVideoHeader& rtp_video_header,
FrameType frametype,
size_t max_payload_len,
size_t last_packet_reduction_len);
~RtpPacketizerGeneric() override;
// Returns total number of packets to be generated.
size_t SetPayloadData(const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) override;
// Get the next payload with generic payload header.
// Write payload and set marker bit of the |packet|.
// Returns true on success, false otherwise.
bool NextPacket(RtpPacketToSend* packet) override;
std::string ToString() override;
private:
const absl::optional<uint16_t> picture_id_;
const uint8_t* payload_data_;
size_t payload_size_;
const size_t max_payload_len_;
const size_t last_packet_reduction_len_;
FrameType frame_type_;
size_t payload_len_per_packet_;
uint8_t generic_header_;
// Number of packets yet to be retrieved by NextPacket() call.
size_t num_packets_left_;
// Number of packets, which will be 1 byte more than the rest.
size_t num_larger_packets_;
void WriteExtendedHeader(uint8_t* out_ptr);
RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
};
// Depacketizer for generic codec.
class RtpDepacketizerGeneric : public RtpDepacketizer {
public:
~RtpDepacketizerGeneric() override;
bool Parse(ParsedPayload* parsed_payload,
const uint8_t* payload_data,
size_t payload_data_length) override;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_