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bd396fdffa
Bug: webrtc:13485 Change-Id: I4e7e29a7661d51e12bb2ee12e319f6cef49482d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318005 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41107}
66 lines
2.1 KiB
C++
66 lines
2.1 KiB
C++
/*
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* Copyright (c) 2023 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_PACKETIZER_H265_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_
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#include <deque>
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#include <queue>
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#include <string>
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#include "api/array_view.h"
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#include "modules/rtp_rtcp/source/rtp_format.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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namespace webrtc {
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class RtpPacketizerH265 : public RtpPacketizer {
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public:
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// Initialize with payload from encoder.
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// The payload_data must be exactly one encoded H.265 frame.
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// For H265 we only support tx-mode SRST.
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RtpPacketizerH265(rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits);
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RtpPacketizerH265(const RtpPacketizerH265&) = delete;
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RtpPacketizerH265& operator=(const RtpPacketizerH265&) = delete;
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~RtpPacketizerH265() override;
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size_t NumPackets() const override;
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// Get the next payload with H.265 payload header.
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// Write payload and set marker bit of the `packet`.
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// Returns true on success or false if there was no payload to packetize.
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bool NextPacket(RtpPacketToSend* rtp_packet) override;
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private:
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struct PacketUnit {
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rtc::ArrayView<const uint8_t> source_fragment;
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bool first_fragment = false;
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bool last_fragment = false;
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bool aggregated = false;
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uint16_t header = 0;
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};
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std::deque<rtc::ArrayView<const uint8_t>> input_fragments_;
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std::queue<PacketUnit> packets_;
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bool GeneratePackets();
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bool PacketizeFu(size_t fragment_index);
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int PacketizeAp(size_t fragment_index);
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void NextAggregatePacket(RtpPacketToSend* rtp_packet);
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void NextFragmentPacket(RtpPacketToSend* rtp_packet);
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const PayloadSizeLimits limits_;
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size_t num_packets_left_ = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_
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