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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 MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
#include <stdint.h>
#include <cstddef>
#include <memory>
#include <optional>
#include <vector>
#include "api/array_view.h"
#include "api/video/video_codec_type.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
namespace webrtc {
class RtpPacketToSend;
class RtpPacketizer {
public:
struct PayloadSizeLimits {
int max_payload_len = 1200;
int first_packet_reduction_len = 0;
int last_packet_reduction_len = 0;
// Reduction len for packet that is first & last at the same time.
int single_packet_reduction_len = 0;
};
// If type is not set, returns a raw packetizer.
static std::unique_ptr<RtpPacketizer> Create(
std::optional<VideoCodecType> type,
ArrayView<const uint8_t> payload,
PayloadSizeLimits limits,
// Codec-specific details.
const RTPVideoHeader& rtp_video_header);
virtual ~RtpPacketizer() = default;
// Returns number of remaining packets to produce by the packetizer.
virtual size_t NumPackets() const = 0;
// Get the next payload with payload header.
// Write payload and set marker bit of the `packet`.
// Returns true on success, false otherwise.
virtual bool NextPacket(RtpPacketToSend* packet) = 0;
// Split payload_len into sum of integers with respect to `limits`.
// Returns empty vector on failure.
static std::vector<int> SplitAboutEqually(int payload_len,
const PayloadSizeLimits& limits);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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