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a4d873786f
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}
83 lines
2.8 KiB
C++
83 lines
2.8 KiB
C++
/*
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* Copyright (c) 2011 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 VP8 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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* together with the fragmentation information and a packetizer mode
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* of choice. Alternatively, if no fragmentation info is available, the
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* second constructor can be used with only payload data and size; in that
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* case the mode kEqualSize is used.
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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_VP8_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
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#include <stddef.h>
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#include <cstdint>
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#include <vector>
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#include "absl/container/inlined_vector.h"
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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 "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
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#include "rtc_base/constructor_magic.h"
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namespace webrtc {
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// Packetizer for VP8.
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class RtpPacketizerVp8 : 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 VP8 frame.
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RtpPacketizerVp8(rtc::ArrayView<const uint8_t> payload,
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PayloadSizeLimits limits,
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const RTPVideoHeaderVP8& hdr_info);
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~RtpPacketizerVp8() override;
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size_t NumPackets() const override;
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// Get the next payload with VP8 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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// VP8 header can use up to 6 bytes.
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using RawHeader = absl::InlinedVector<uint8_t, 6>;
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static RawHeader BuildHeader(const RTPVideoHeaderVP8& header);
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RawHeader hdr_;
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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(RtpPacketizerVp8);
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};
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// Depacketizer for VP8.
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class RtpDepacketizerVp8 : public RtpDepacketizer {
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public:
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~RtpDepacketizerVp8() override = default;
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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_VP8_H_
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