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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}
80 lines
2.8 KiB
C++
80 lines
2.8 KiB
C++
/*
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* Copyright (c) 2013 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_VIDEO_GENERIC_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
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#include <string>
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#include "common_types.h" // NOLINT(build/include)
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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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namespace RtpFormatVideoGeneric {
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static const uint8_t kKeyFrameBit = 0x01;
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static const uint8_t kFirstPacketBit = 0x02;
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// If this bit is set, there will be an extended header contained in this
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// packet. This was added later so old clients will not send this.
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static const uint8_t kExtendedHeaderBit = 0x04;
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} // namespace RtpFormatVideoGeneric
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class RtpPacketizerGeneric : 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 generic frame.
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RtpPacketizerGeneric(const RTPVideoHeader& rtp_video_header,
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FrameType frametype,
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size_t max_payload_len,
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size_t last_packet_reduction_len);
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~RtpPacketizerGeneric() override;
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// Returns total number of packets to be generated.
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size_t SetPayloadData(const uint8_t* payload_data,
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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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// Get the next payload with generic 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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private:
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const absl::optional<uint16_t> picture_id_;
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const uint8_t* payload_data_;
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size_t payload_size_;
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const size_t max_payload_len_;
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const size_t last_packet_reduction_len_;
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FrameType frame_type_;
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size_t payload_len_per_packet_;
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uint8_t generic_header_;
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// Number of packets yet to be retrieved by NextPacket() call.
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size_t num_packets_left_;
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// Number of packets, which will be 1 byte more than the rest.
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size_t num_larger_packets_;
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void WriteExtendedHeader(uint8_t* out_ptr);
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RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
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};
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// Depacketizer for generic codec.
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class RtpDepacketizerGeneric : public RtpDepacketizer {
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public:
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~RtpDepacketizerGeneric() override;
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bool Parse(ParsedPayload* parsed_payload,
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const uint8_t* payload_data,
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size_t payload_data_length) override;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_GENERIC_H_
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