fcebe0e1ca
Assumption extra needed bytes for single packet needs is sum of extra bytes for first and last packet moved up to RTPSenderVideo from individual packetizers. There it can be fixed. Bug: webrtc:9868 Change-Id: I24c80ffa5c174afd3fe3e92fa86ef75560bb961e Reviewed-on: https://webrtc-review.googlesource.com/c/105662 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25160}
88 lines
3.0 KiB
C++
88 lines
3.0 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/array_view.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class RtpPacketToSend;
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class RtpPacketizer {
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public:
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struct PayloadSizeLimits {
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int max_payload_len = 1200;
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int first_packet_reduction_len = 0;
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int last_packet_reduction_len = 0;
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// Reduction len for packet that is first & last at the same time.
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int single_packet_reduction_len = 0;
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};
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static std::unique_ptr<RtpPacketizer> Create(
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VideoCodecType type,
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rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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// Codec-specific details.
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const RTPVideoHeader& rtp_video_header,
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FrameType frame_type,
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const RTPFragmentationHeader* fragmentation);
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virtual ~RtpPacketizer() = default;
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// Returns number of remaining packets to produce by the packetizer.
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virtual size_t NumPackets() const = 0;
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// Get the next payload with payload header.
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// Write payload and set marker bit of the |packet|.
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// Returns true on success, false otherwise.
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virtual bool NextPacket(RtpPacketToSend* packet) = 0;
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// Split payload_len into sum of integers with respect to |limits|.
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// Returns empty vector on failure.
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static std::vector<int> SplitAboutEqually(int payload_len,
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const PayloadSizeLimits& limits);
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};
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// TODO(sprang): Update the depacketizer to return a std::unqie_ptr with a copy
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// of the parsed payload, rather than just a pointer into the incoming buffer.
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// This way we can move some parsing out from the jitter buffer into here, and
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// the jitter buffer can just store that pointer rather than doing a copy there.
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class RtpDepacketizer {
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public:
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struct ParsedPayload {
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RTPVideoHeader& video_header() { return video; }
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const RTPVideoHeader& video_header() const { return video; }
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RTPVideoHeader video;
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const uint8_t* payload;
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size_t payload_length;
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FrameType frame_type;
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};
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static RtpDepacketizer* Create(VideoCodecType type);
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virtual ~RtpDepacketizer() {}
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// Parses the RTP payload, parsed result will be saved in |parsed_payload|.
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virtual bool Parse(ParsedPayload* parsed_payload,
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const uint8_t* payload_data,
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size_t payload_data_length) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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