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f7f8a1f176
merge construction and call to SetPayloadData Add NumPackets instead of SetPayloadData Remove virtual ToString() as unused move CHECK(rtp_video_header) from RtpPacketizer::Create to RtpSenderVideo::SendVideo Bug: webrtc:9680 Change-Id: I074644e048c797eb836f79979df363fe1ea0075e Reviewed-on: https://webrtc-review.googlesource.com/96543 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24474}
99 lines
3.4 KiB
C++
99 lines
3.4 KiB
C++
/*
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* Copyright (c) 2015 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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//
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// This file contains the declaration of the VP9 packetizer class.
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// A packetizer object is created for each encoded video frame. The
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// constructor is called with the payload data and size.
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//
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// After creating the packetizer, the method NextPacket is called
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// repeatedly to get all packets for the frame. The method returns
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// false as long as there are more packets left to fetch.
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//
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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
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#include <queue>
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#include <string>
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/source/rtp_format.h"
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class RtpPacketizerVp9 : public RtpPacketizer {
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public:
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RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr,
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size_t max_payload_length,
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size_t last_packet_reduction_len);
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~RtpPacketizerVp9() override;
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// The payload data must be one encoded VP9 layer frame.
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size_t SetPayloadData(const uint8_t* payload,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation);
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size_t NumPackets() const override;
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// Gets the next payload with VP9 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* packet) override;
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typedef struct {
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size_t payload_start_pos;
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size_t size;
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bool layer_begin;
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bool layer_end;
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} PacketInfo;
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typedef std::queue<PacketInfo> PacketInfoQueue;
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private:
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// Calculates all packet sizes and loads info to packet queue.
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void GeneratePackets();
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// Writes the payload descriptor header and copies payload to the |buffer|.
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// |packet_info| determines which part of the payload to write.
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// |last| indicates if the packet is the last packet in the frame.
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// Returns true on success, false otherwise.
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bool WriteHeaderAndPayload(const PacketInfo& packet_info,
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RtpPacketToSend* packet,
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bool last) const;
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// Writes payload descriptor header to |buffer|.
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// Returns true on success, false otherwise.
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bool WriteHeader(const PacketInfo& packet_info,
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uint8_t* buffer,
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size_t* header_length) const;
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const RTPVideoHeaderVP9 hdr_;
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const size_t max_payload_length_; // The max length in bytes of one packet.
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const uint8_t* payload_; // The payload data to be packetized.
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size_t payload_size_; // The size in bytes of the payload data.
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const size_t last_packet_reduction_len_;
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PacketInfoQueue packets_;
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9);
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};
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class RtpDepacketizerVp9 : public RtpDepacketizer {
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public:
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~RtpDepacketizerVp9() override = default;
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bool Parse(ParsedPayload* parsed_payload,
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const uint8_t* payload,
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size_t payload_length) override;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP9_H_
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