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* qatar/master: (33 commits) rtpdec_qdm2: Don't try to parse data packet if no configuration is received ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand. ac3enc: clean up count_frame_bits() and count_frame_bits_fixed() mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame. srtdec: make sure we don't write past the end of buffer wmaenc: improve channel count and bitrate error handling in encode_init() matroskaenc: make sure we don't produce invalid file with no codec ID matroskadec: check that pointers were initialized before accessing them lavf: fix function name in compute_pkt_fields2 av_dlog message lavf: fix av_find_best_stream when providing a wanted stream. lavf: fix av_find_best_stream when decoder_ret is given and using a related stream. ffmpeg: factorize quality calculation tiff: add support for SamplesPerPixel tag in tiff_decode_tag() tiff: Prefer enum TiffCompr over int for TiffContext.compr. mov: Support edit list atom version 1. configure: Enable libpostproc automatically if GPL code is enabled. Cosmetics: fix prototypes in oggdec oggdec: fix memleak with continuous streams. matroskaenc: add missing new line in av_log() call dnxhdenc: add AVClass in private context. ... swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code. Above code is also just in case its not obvios to a large extended duplicates that where cherry picked from ffmpeg. Conflicts: configure ffmpeg.c libavformat/matroskaenc.c libavutil/pixfmt.h libswscale/ppc/swscale_template.c libswscale/swscale.c libswscale/swscale_template.c libswscale/utils.c libswscale/x86/swscale_template.c tests/fate/h264.mak tests/ref/lavfi/pixdesc_le tests/ref/lavfi/pixfmts_copy_le tests/ref/lavfi/pixfmts_null_le tests/ref/lavfi/pixfmts_scale_le tests/ref/lavfi/pixfmts_vflip_le Merged-by: Michael Niedermayer <michaelni@gmx.at>
318 lines
10 KiB
C
318 lines
10 KiB
C
/*
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* QDesign Music 2 (QDM2) payload for RTP
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* Copyright (c) 2010 Ronald S. Bultje
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* @brief RTP support for the QDM2 payload (todo: wiki)
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* @author Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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*/
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#include <string.h>
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#include "libavutil/intreadwrite.h"
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#include "libavcodec/avcodec.h"
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#include "rtp.h"
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#include "rtpdec.h"
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#include "rtpdec_formats.h"
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struct PayloadContext {
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/** values read from the config header, used as packet headers */
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//@{
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int block_type; ///< superblock type, value 2 .. 8
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int block_size; ///< from extradata, used as pkt length
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int subpkts_per_block; ///< max. nr. of subpackets to add per output buffer
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//@}
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/** Temporary storage for superblock restoring, per packet ID (0x80 total) */
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//@{
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uint16_t len[0x80]; ///< how much the temporary buffer is filled
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uint8_t buf[0x80][0x800]; ///< the temporary storage buffer
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unsigned int cache; ///< number of data packets that we have cached right now
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unsigned int n_pkts; ///< number of RTP packets received since last packet output / config
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uint32_t timestamp; ///< timestamp of next-to-be-returned packet
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//@}
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};
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/**
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* Parses configuration (basically the codec-specific extradata) from
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* a RTP config subpacket (starts with 0xff).
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*
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* Layout of the config subpacket (in bytes):
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* 1: 0xFF <- config ID
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* then an array {
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* 1: size <- of the current item
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* 1: item type <- 0 .. 4
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* size-2: data <- data depends on the item type
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* }
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*
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* Item 0 implies the end of the config subpacket, and has no data.
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* Item 1 implies a stream configuration without extradata.
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* Item 2 max. nr. of subpackets per superblock
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* Item 3 superblock type for the stream
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* Item 4 implies a stream configuration with extradata (size >= 0x1c).
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*
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* @return <0 on error, otherwise the number of bytes parsed from the
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* input buffer.
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*/
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static int qdm2_parse_config(PayloadContext *qdm, AVStream *st,
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const uint8_t *buf, const uint8_t *end)
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{
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const uint8_t *p = buf;
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while (end - p >= 2) {
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unsigned int item_len = p[0], config_item = p[1];
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if (item_len < 2 || end - p < item_len || config_item > 4)
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return AVERROR_INVALIDDATA;
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switch (config_item) {
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case 0: /* end of config block */
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return p - buf + item_len;
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case 1: /* stream without extradata */
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/* FIXME: set default qdm->block_size */
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break;
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case 2: /**< subpackets per block */
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if (item_len < 3)
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return AVERROR_INVALIDDATA;
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qdm->subpkts_per_block = p[2];
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break;
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case 3: /* superblock type */
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if (item_len < 4)
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return AVERROR_INVALIDDATA;
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qdm->block_type = AV_RB16(p + 2);
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break;
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case 4: /* stream with extradata */
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if (item_len < 30)
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return AVERROR_INVALIDDATA;
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av_freep(&st->codec->extradata);
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st->codec->extradata_size = 26 + item_len;
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if (!(st->codec->extradata = av_mallocz(st->codec->extradata_size + FF_INPUT_BUFFER_PADDING_SIZE))) {
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st->codec->extradata_size = 0;
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return AVERROR(ENOMEM);
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}
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AV_WB32(st->codec->extradata, 12);
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memcpy(st->codec->extradata + 4, "frma", 4);
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memcpy(st->codec->extradata + 8, "QDM2", 4);
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AV_WB32(st->codec->extradata + 12, 6 + item_len);
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memcpy(st->codec->extradata + 16, "QDCA", 4);
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memcpy(st->codec->extradata + 20, p + 2, item_len - 2);
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AV_WB32(st->codec->extradata + 18 + item_len, 8);
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AV_WB32(st->codec->extradata + 22 + item_len, 0);
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qdm->block_size = AV_RB32(p + 26);
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break;
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}
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p += item_len;
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}
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return AVERROR(EAGAIN); /* not enough data */
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}
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/**
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* Parses a single subpacket. We store this subpacket in an intermediate
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* buffer (position depends on the ID (byte[0]). When called, at least
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* 4 bytes are available for reading (see qdm2_parse_packet()).
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*
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* Layout of a single subpacket (RTP packets commonly contain multiple
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* such subpackets) - length in bytes:
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* 1: ordering ID <- 0 .. 0x7F
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* 1: subpacket type <- 0 .. 0x7F; value & 0x80 means subpacket length = 2 bytes, else 1 byte
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* 1/2: subpacket length <- length of the data following the flags/length fields
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* if (subpacket type & 0x7F) == 0x7F
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* 1: subpacket type, higher bits
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* size: subpacket data
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*
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* The subpackets come in randomly, and should be encapsulated into 1
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* or more superblocks (containing qdm->subpkts_per_block subpackets
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* each) per RTP packet, in order of ascending "ordering ID", see
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* qdm2_restore_block().
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*
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* @return <0 on error, otherwise the number of bytes parsed from the
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* input buffer.
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*/
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static int qdm2_parse_subpacket(PayloadContext *qdm, AVStream *st,
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const uint8_t *buf, const uint8_t *end)
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{
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const uint8_t *p = buf;
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unsigned int id, len, type, to_copy;
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/* parse header so we know the size of the header/data */
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id = *p++;
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type = *p++;
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if (type & 0x80) {
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len = AV_RB16(p);
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p += 2;
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type &= 0x7F;
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} else
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len = *p++;
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if (end - p < len + (type == 0x7F) || id >= 0x80)
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return AVERROR_INVALIDDATA;
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if (type == 0x7F)
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type |= *p++ << 8;
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/* copy data into a temporary buffer */
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to_copy = FFMIN(len + (p - &buf[1]), 0x800 - qdm->len[id]);
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memcpy(&qdm->buf[id][qdm->len[id]], buf + 1, to_copy);
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qdm->len[id] += to_copy;
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return p + len - buf;
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}
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/**
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* Adds a superblock header around a set of subpackets.
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*
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* @return <0 on error, else 0.
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*/
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static int qdm2_restore_block(PayloadContext *qdm, AVStream *st, AVPacket *pkt)
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{
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int to_copy, n, res, include_csum;
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uint8_t *p, *csum_pos = NULL;
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/* create packet to hold subpkts into a superblock */
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assert(qdm->cache > 0);
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for (n = 0; n < 0x80; n++)
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if (qdm->len[n] > 0)
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break;
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assert(n < 0x80);
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if ((res = av_new_packet(pkt, qdm->block_size)) < 0)
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return res;
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memset(pkt->data, 0, pkt->size);
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pkt->stream_index = st->index;
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p = pkt->data;
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/* superblock header */
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if (qdm->len[n] > 0xff) {
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*p++ = qdm->block_type | 0x80;
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AV_WB16(p, qdm->len[n]);
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p += 2;
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} else {
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*p++ = qdm->block_type;
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*p++ = qdm->len[n];
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}
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if ((include_csum = (qdm->block_type == 2 || qdm->block_type == 4))) {
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csum_pos = p;
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p += 2;
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}
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/* subpacket data */
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to_copy = FFMIN(qdm->len[n], pkt->size - (p - pkt->data));
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memcpy(p, qdm->buf[n], to_copy);
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qdm->len[n] = 0;
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/* checksum header */
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if (include_csum) {
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unsigned int total = 0;
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uint8_t *q;
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for (q = pkt->data; q < &pkt->data[qdm->block_size]; q++)
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total += *q;
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AV_WB16(csum_pos, (uint16_t) total);
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}
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return 0;
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}
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/** return 0 on packet, no more left, 1 on packet, -1 on partial packet... */
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static int qdm2_parse_packet(AVFormatContext *s, PayloadContext *qdm,
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AVStream *st, AVPacket *pkt,
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uint32_t *timestamp,
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const uint8_t *buf, int len, int flags)
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{
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int res = AVERROR_INVALIDDATA, n;
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const uint8_t *end = buf + len, *p = buf;
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if (len > 0) {
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if (len < 2)
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return AVERROR_INVALIDDATA;
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/* configuration block */
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if (*p == 0xff) {
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if (qdm->n_pkts > 0) {
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av_log(s, AV_LOG_WARNING,
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"Out of sequence config - dropping queue\n");
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qdm->n_pkts = 0;
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memset(qdm->len, 0, sizeof(qdm->len));
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}
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if ((res = qdm2_parse_config(qdm, st, ++p, end)) < 0)
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return res;
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p += res;
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/* We set codec_id to CODEC_ID_NONE initially to
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* delay decoder initialization since extradata is
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* carried within the RTP stream, not SDP. Here,
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* by setting codec_id to CODEC_ID_QDM2, we are signalling
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* to the decoder that it is OK to initialize. */
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st->codec->codec_id = CODEC_ID_QDM2;
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}
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if (st->codec->codec_id == CODEC_ID_NONE)
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return AVERROR(EAGAIN);
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/* subpackets */
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while (end - p >= 4) {
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if ((res = qdm2_parse_subpacket(qdm, st, p, end)) < 0)
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return res;
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p += res;
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}
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qdm->timestamp = *timestamp;
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if (++qdm->n_pkts < qdm->subpkts_per_block)
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return AVERROR(EAGAIN);
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qdm->cache = 0;
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for (n = 0; n < 0x80; n++)
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if (qdm->len[n] > 0)
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qdm->cache++;
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}
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/* output the subpackets into freshly created superblock structures */
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if (!qdm->cache || (res = qdm2_restore_block(qdm, st, pkt)) < 0)
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return res;
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if (--qdm->cache == 0)
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qdm->n_pkts = 0;
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*timestamp = qdm->timestamp;
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qdm->timestamp = RTP_NOTS_VALUE;
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return (qdm->cache > 0) ? 1 : 0;
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}
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static PayloadContext *qdm2_extradata_new(void)
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{
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return av_mallocz(sizeof(PayloadContext));
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}
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static void qdm2_extradata_free(PayloadContext *qdm)
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{
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av_free(qdm);
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}
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RTPDynamicProtocolHandler ff_qdm2_dynamic_handler = {
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.enc_name = "X-QDM",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = CODEC_ID_NONE,
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.alloc = qdm2_extradata_new,
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.free = qdm2_extradata_free,
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.parse_packet = qdm2_parse_packet,
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};
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