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/*
* "Real" compatible muxer and demuxer.
* Copyright (c) 2000, 2001 Fabrice Bellard.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFORMAT_RM_H
#define AVFORMAT_RM_H
#include "avformat.h"
typedef struct {
int nb_packets;
int packet_total_size;
int packet_max_size;
/* codec related output */
int bit_rate;
float frame_rate;
int nb_frames; /* current frame number */
int total_frames; /* total number of frames */
int num;
AVCodecContext *enc;
} StreamInfo;
typedef struct {
StreamInfo streams[2];
StreamInfo *audio_stream, *video_stream;
int data_pos; /* position of the data after the header */
int nb_packets;
int old_format;
int current_stream;
int remaining_len;
uint8_t *videobuf; ///< place to store merged video frame
int videobufsize; ///< current assembled frame size
int videobufpos; ///< position for the next slice in the video buffer
int curpic_num; ///< picture number of current frame
int cur_slice, slices;
int64_t pktpos; ///< first slice position in file
/// Audio descrambling matrix parameters
uint8_t *audiobuf; ///< place to store reordered audio data
int64_t audiotimestamp; ///< Audio packet timestamp
int sub_packet_cnt; // Subpacket counter, used while reading
int sub_packet_size, sub_packet_h, coded_framesize; ///< Descrambling parameters from container
int audio_stream_num; ///< Stream number for audio packets
int audio_pkt_cnt; ///< Output packet counter
int audio_framesize; /// Audio frame size from container
int sub_packet_lengths[16]; /// Length of each aac subpacket
} RMContext;
/**
* Read the MDPR chunk, which contains stream-specific codec initialization
* parameters.
*
* @param s context containing RMContext and ByteIOContext for stream reading
* @param st the stream that the MDPR chunk belongs to and where to store the
* parameters read from the chunk into
* @return 0 on success, errno codes on error
*/
int ff_rm_read_mdpr_codecdata (AVFormatContext *s, AVStream *st);
/**
* Parse one rm-stream packet from the input bytestream.
*
* @param s context containing RMContext and ByteIOContext for stream reading
* @param st stream to which the packet to be read belongs
* @param len packet length to read from the input
* @param pkt packet location to store the parsed packet data
* @param seq pointer to an integer containing the sequence number, may be
* updated
* @param flags pointer to an integer containing the packet flags, may be
updated
* @param ts pointer to timestamp, may be updated
* @return 0 on success, errno codes on error
*/
int ff_rm_parse_packet (AVFormatContext *s, AVStream *st, int len,
AVPacket *pkt, int *seq, int *flags, int64_t *ts);
/**
* Retrieve one cached packet from the rm-context. The real container can
* store several packets (as interpreted by the codec) in a single container
* packet, which means the demuxer holds some back when the first container
* packet is parsed and returned. The result is that rm->audio_pkt_cnt is
* a positive number, the amount of cached packets. Using this function, each
* of those packets can be retrieved sequentially.
*
* @param s context containing RMContext and ByteIOContext for stream reading
* @param st stream that this packet belongs to
* @param pkt location to store the packet data
*/
void ff_rm_retrieve_cache (AVFormatContext *s, AVStream *st, AVPacket *pkt);
#endif /* AVFORMAT_RM_H */
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