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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
76 lines
2.5 KiB
C++
76 lines
2.5 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 <stdint.h>
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#include <vector>
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#include "api/array_view.h"
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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/constructor_magic.h"
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namespace webrtc {
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class RtpPacketToSend;
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struct RTPVideoHeader;
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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(rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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const RTPVideoHeader& rtp_video_header,
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FrameType frametype);
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~RtpPacketizerGeneric() override;
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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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// Fills header_ and header_size_ members.
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void BuildHeader(const RTPVideoHeader& rtp_video_header,
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FrameType frame_type);
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uint8_t header_[3];
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size_t header_size_;
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rtc::ArrayView<const uint8_t> remaining_payload_;
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std::vector<int> payload_sizes_;
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std::vector<int>::const_iterator current_packet_;
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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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