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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:13:27 +0000 |
commit | 40a355a42d4a9444dc753c04c6608dade2f06a23 (patch) | |
tree | 871fc667d2de662f171103ce5ec067014ef85e61 /third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h | |
parent | Adding upstream version 124.0.1. (diff) | |
download | firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.tar.xz firefox-40a355a42d4a9444dc753c04c6608dade2f06a23.zip |
Adding upstream version 125.0.1.upstream/125.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h')
-rw-r--r-- | third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h | 66 |
1 files changed, 66 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h new file mode 100644 index 0000000000..95442f795c --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_h265.h @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2023 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_PACKETIZER_H265_H_ +#define MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_ + +#include <deque> +#include <queue> +#include <string> + +#include "api/array_view.h" +#include "modules/rtp_rtcp/source/rtp_format.h" +#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" + +namespace webrtc { + +class RtpPacketizerH265 : public RtpPacketizer { + public: + // Initialize with payload from encoder. + // The payload_data must be exactly one encoded H.265 frame. + // For H265 we only support tx-mode SRST. + RtpPacketizerH265(rtc::ArrayView<const uint8_t> payload, + PayloadSizeLimits limits); + + RtpPacketizerH265(const RtpPacketizerH265&) = delete; + RtpPacketizerH265& operator=(const RtpPacketizerH265&) = delete; + + ~RtpPacketizerH265() override; + + size_t NumPackets() const override; + + // Get the next payload with H.265 payload header. + // Write payload and set marker bit of the `packet`. + // Returns true on success or false if there was no payload to packetize. + bool NextPacket(RtpPacketToSend* rtp_packet) override; + + private: + struct PacketUnit { + rtc::ArrayView<const uint8_t> source_fragment; + bool first_fragment = false; + bool last_fragment = false; + bool aggregated = false; + uint16_t header = 0; + }; + std::deque<rtc::ArrayView<const uint8_t>> input_fragments_; + std::queue<PacketUnit> packets_; + + bool GeneratePackets(); + bool PacketizeFu(size_t fragment_index); + int PacketizeAp(size_t fragment_index); + + void NextAggregatePacket(RtpPacketToSend* rtp_packet); + void NextFragmentPacket(RtpPacketToSend* rtp_packet); + + const PayloadSizeLimits limits_; + size_t num_packets_left_ = 0; +}; +} // namespace webrtc +#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKETIZER_H265_H_ |