b7686af27c
Make member variable payload_size_ int instead of unsigned to avoid warnings when comparing (> and >=). Review URL: http://webrtc-codereview.appspot.com/40001 git-svn-id: http://webrtc.googlecode.com/svn/trunk@86 4adac7df-926f-26a2-2b94-8c16560cd09d
107 lines
4.0 KiB
C++
107 lines
4.0 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 kSloppy 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 WEBRTC_MODULES_RTP_RTCP_RTP_FORMAT_VP8_H_
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#define WEBRTC_MODULES_RTP_RTCP_RTP_FORMAT_VP8_H_
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#include "module_common_types.h"
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#include "typedefs.h"
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namespace webrtc
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{
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enum VP8PacketizerMode
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{
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kStrict = 0, // split partitions if too large; never aggregate, balance size
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kAggregate, // split partitions if too large; aggregate whole partitions
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kSloppy, // split entire payload without considering partition limits
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kNumModes,
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};
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// Packetizer for VP8.
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class RtpFormatVp8
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{
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public:
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// Initialize with payload from encoder and fragmentation info.
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// The payload_data must be exactly one encoded VP8 frame.
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RtpFormatVp8(const WebRtc_UWord8* payload_data,
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WebRtc_UWord32 payload_size,
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const RTPFragmentationHeader& fragmentation,
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VP8PacketizerMode mode);
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// Initialize without fragmentation info. Mode kSloppy will be used.
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// The payload_data must be exactly one encoded VP8 frame.
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RtpFormatVp8(const WebRtc_UWord8* payload_data,
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WebRtc_UWord32 payload_size);
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// Get the next payload with VP8 payload header.
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// max_payload_len limits the sum length of payload and VP8 payload header.
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// buffer is a pointer to where the output will be written.
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// bytes_to_send is an output variable that will contain number of bytes
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// written to buffer.
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// Returns true for the last packet of the frame, false otherwise (i.e.,
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// call the function again to get the next packet).
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int NextPacket(int max_payload_len, WebRtc_UWord8* buffer,
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int* bytes_to_send, bool* last_packet);
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private:
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enum AggregationMode
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{
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kAggrNone = 0, // no aggregation
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kAggrPartitions, // aggregate intact partitions
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kAggrFragments // aggregate intact and fragmented partitions
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};
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static const AggregationMode aggr_modes_[kNumModes];
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static const bool bal_modes_[kNumModes];
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static const bool sep_first_modes_[kNumModes];
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// Calculate size of next chunk to send. Returns 0 if none can be sent.
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int CalcNextSize(int max_payload_len, int remaining_bytes,
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bool split_payload) const;
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// Write the payload header and copy the payload to the buffer.
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// Will copy send_bytes bytes from the current position on the payload data.
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// last_fragment indicates that this packet ends with the last byte of a
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// partition.
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int WriteHeaderAndPayload(int send_bytes, bool end_of_fragment,
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WebRtc_UWord8* buffer);
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const WebRtc_UWord8* payload_data_;
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const int payload_size_;
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RTPFragmentationHeader part_info_;
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int payload_bytes_sent_;
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int part_ix_;
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bool beginning_; // first partition in this frame
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bool first_fragment_; // first fragment of a partition
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const int vp8_header_bytes_; // length of VP8 payload header
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AggregationMode aggr_mode_;
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bool balance_;
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bool separate_first_;
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};
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}
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#endif /* WEBRTC_MODULES_RTP_RTCP_RTP_FORMAT_VP8_H_ */
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