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This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
89 lines
3.1 KiB
C++
89 lines
3.1 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 "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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// Packets returned by |NextPacket| will contain the generic payload header.
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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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VideoFrameType frametype);
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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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// Packets returned by |NextPacket| will contain raw payload without the
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// generic payload header.
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RtpPacketizerGeneric(rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits);
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~RtpPacketizerGeneric() override;
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size_t NumPackets() const override;
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// Get the next payload.
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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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VideoFrameType 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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// Parses the generic payload header if |generic_header_enabled| is true,
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// returns raw payload otherwise.
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explicit RtpDepacketizerGeneric(bool generic_header_enabled);
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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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private:
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bool generic_header_enabled_;
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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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