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426a80ce08
Also changes default value of frame ID in RTPVideoHeader to kNoPictureId. Special care should be take so that picture ID will not be set in RTPVideoHeader unless the client on the end supports deserializing extended generic header. Bug: webrtc:9582 Change-Id: Ib096373ed187f31e51d481193a2bda56de68f167 Reviewed-on: https://webrtc-review.googlesource.com/92084 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24250}
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) 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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std::string ToString() 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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