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820021d246
instead reparse nalu boundaries from the bitstream. H264 is the last use of the RTPFragmentationHeader and this would allow to avoid passing and precalculating this legacy structure. Bug: webrtc:6471 Change-Id: Ia6e8bf0836fd5c022423d836894cde81f136d1f1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178911 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31746}
89 lines
3.1 KiB
C++
89 lines
3.1 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_H264_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <deque>
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#include <memory>
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#include <queue>
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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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#include "modules/video_coding/codecs/h264/include/h264_globals.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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class RtpPacketizerH264 : 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 H264 frame.
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RtpPacketizerH264(rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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H264PacketizationMode packetization_mode);
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~RtpPacketizerH264() override;
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size_t NumPackets() const override;
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// Get the next payload with H264 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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bool NextPacket(RtpPacketToSend* rtp_packet) override;
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private:
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// A packet unit (H264 packet), to be put into an RTP packet:
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// If a NAL unit is too large for an RTP packet, this packet unit will
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// represent a FU-A packet of a single fragment of the NAL unit.
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// If a NAL unit is small enough to fit within a single RTP packet, this
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// packet unit may represent a single NAL unit or a STAP-A packet, of which
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// there may be multiple in a single RTP packet (if so, aggregated = true).
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struct PacketUnit {
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PacketUnit(rtc::ArrayView<const uint8_t> source_fragment,
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bool first_fragment,
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bool last_fragment,
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bool aggregated,
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uint8_t header)
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: source_fragment(source_fragment),
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first_fragment(first_fragment),
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last_fragment(last_fragment),
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aggregated(aggregated),
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header(header) {}
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rtc::ArrayView<const uint8_t> source_fragment;
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bool first_fragment;
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bool last_fragment;
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bool aggregated;
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uint8_t header;
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};
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bool GeneratePackets(H264PacketizationMode packetization_mode);
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bool PacketizeFuA(size_t fragment_index);
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size_t PacketizeStapA(size_t fragment_index);
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bool PacketizeSingleNalu(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_;
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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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RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerH264);
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H264_H_
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