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8037fc6ffa
Bug: webrtc:342905193 No-Try: True Change-Id: Icc968be43b8830038ea9a1f5f604307220457807 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021 Auto-Submit: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42911}
63 lines
2 KiB
C++
63 lines
2 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 <stdint.h>
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#include <memory>
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#include <optional>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/rtp_rtcp/source/rtp_video_header.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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// If type is not set, returns a raw packetizer.
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static std::unique_ptr<RtpPacketizer> Create(
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std::optional<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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// TODO(bugs.webrtc.org/15927): remove after rollout.
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bool enable_av1_even_split = false);
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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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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
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