mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 06:40:43 +01:00
5ab67a5d71
This CL is in preparation to change the RTPVideoTypeHeader into an absl::variant. Bug: none Change-Id: I1672d866df0395f3417d8e278cc67f017ab0ff98 Reviewed-on: https://webrtc-review.googlesource.com/87261 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23856}
74 lines
2.6 KiB
C++
74 lines
2.6 KiB
C++
/*
|
|
* 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 <string>
|
|
|
|
#include "common_types.h" // NOLINT(build/include)
|
|
#include "modules/include/module_common_types.h"
|
|
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
#include "rtc_base/constructormagic.h"
|
|
|
|
namespace webrtc {
|
|
class RtpPacketToSend;
|
|
|
|
class RtpPacketizer {
|
|
public:
|
|
static RtpPacketizer* Create(VideoCodecType type,
|
|
size_t max_payload_len,
|
|
size_t last_packet_reduction_len,
|
|
const RTPVideoHeader* rtp_video_header,
|
|
FrameType frame_type);
|
|
|
|
virtual ~RtpPacketizer() {}
|
|
|
|
// Returns total number of packets which would be produced by the packetizer.
|
|
virtual size_t SetPayloadData(
|
|
const uint8_t* payload_data,
|
|
size_t payload_size,
|
|
const RTPFragmentationHeader* fragmentation) = 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;
|
|
|
|
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 {
|
|
public:
|
|
struct ParsedPayload {
|
|
RTPVideoHeader& video_header() { return video; }
|
|
const RTPVideoHeader& video_header() const { return video; }
|
|
RTPVideoHeader video;
|
|
|
|
const uint8_t* payload;
|
|
size_t payload_length;
|
|
FrameType frame_type;
|
|
};
|
|
|
|
static RtpDepacketizer* Create(VideoCodecType 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
|
|
#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
|