webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
hlundin@google.com 7c53a0c67e Make r80 build on Windows
Re-submitting revision r80, but with bugfix to make it
build on Windows.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@85 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-15 09:38:28 +00:00

107 lines
4.0 KiB
C++

/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file contains the declaration of the VP8 packetizer class.
* A packetizer object is created for each encoded video frame. The
* constructor is called with the payload data and size,
* together with the fragmentation information and a packetizer mode
* of choice. Alternatively, if no fragmentation info is available, the
* second constructor can be used with only payload data and size; in that
* case the mode kSloppy is used.
*
* After creating the packetizer, the method NextPacket is called
* repeatedly to get all packets for the frame. The method returns
* false as long as there are more packets left to fetch.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_FORMAT_VP8_H_
#define WEBRTC_MODULES_RTP_RTCP_RTP_FORMAT_VP8_H_
#include "module_common_types.h"
#include "typedefs.h"
namespace webrtc
{
enum VP8PacketizerMode
{
kStrict = 0, // split partitions if too large; never aggregate, balance size
kAggregate, // split partitions if too large; aggregate whole partitions
kSloppy, // split entire payload without considering partition limits
kNumModes,
};
// Packetizer for VP8.
class RtpFormatVp8
{
public:
// Initialize with payload from encoder and fragmentation info.
// The payload_data must be exactly one encoded VP8 frame.
RtpFormatVp8(const WebRtc_UWord8* payload_data,
WebRtc_UWord32 payload_size,
const RTPFragmentationHeader& fragmentation,
VP8PacketizerMode mode);
// Initialize without fragmentation info. Mode kSloppy will be used.
// The payload_data must be exactly one encoded VP8 frame.
RtpFormatVp8(const WebRtc_UWord8* payload_data,
WebRtc_UWord32 payload_size);
// Get the next payload with VP8 payload header.
// max_payload_len limits the sum length of payload and VP8 payload header.
// buffer is a pointer to where the output will be written.
// bytes_to_send is an output variable that will contain number of bytes
// written to buffer.
// Returns true for the last packet of the frame, false otherwise (i.e.,
// call the function again to get the next packet).
int NextPacket(int max_payload_len, WebRtc_UWord8* buffer,
int* bytes_to_send, bool* last_packet);
private:
enum AggregationMode
{
kAggrNone = 0, // no aggregation
kAggrPartitions, // aggregate intact partitions
kAggrFragments // aggregate intact and fragmented partitions
};
static const AggregationMode aggr_modes_[kNumModes];
static const bool bal_modes_[kNumModes];
static const bool sep_first_modes_[kNumModes];
// Calculate size of next chunk to send. Returns 0 if none can be sent.
int CalcNextSize(int max_payload_len, int remaining_bytes,
bool split_payload) const;
// Write the payload header and copy the payload to the buffer.
// Will copy send_bytes bytes from the current position on the payload data.
// last_fragment indicates that this packet ends with the last byte of a
// partition.
int WriteHeaderAndPayload(int send_bytes, bool end_of_fragment,
WebRtc_UWord8* buffer);
const WebRtc_UWord8* payload_data_;
const WebRtc_UWord32 payload_size_;
RTPFragmentationHeader part_info_;
int payload_bytes_sent_;
int part_ix_;
bool beginning_; // first partition in this frame
bool first_fragment_; // first fragment of a partition
const int vp8_header_bytes_; // length of VP8 payload header
AggregationMode aggr_mode_;
bool balance_;
bool separate_first_;
};
}
#endif /* WEBRTC_MODULES_RTP_RTCP_RTP_FORMAT_VP8_H_ */