| /* |
| * Shorten decoder |
| * Copyright (c) 2005 Jeff Muizelaar |
| * |
| * 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 |
| */ |
| |
| /** |
| * @file |
| * Shorten decoder |
| * @author Jeff Muizelaar |
| * |
| */ |
| |
| #include <limits.h> |
| #include "avcodec.h" |
| #include "bytestream.h" |
| #include "get_bits.h" |
| #include "golomb.h" |
| #include "internal.h" |
| |
| #define MAX_CHANNELS 8 |
| #define MAX_BLOCKSIZE 65535 |
| |
| #define OUT_BUFFER_SIZE 16384 |
| |
| #define ULONGSIZE 2 |
| |
| #define WAVE_FORMAT_PCM 0x0001 |
| |
| #define DEFAULT_BLOCK_SIZE 256 |
| |
| #define TYPESIZE 4 |
| #define CHANSIZE 0 |
| #define LPCQSIZE 2 |
| #define ENERGYSIZE 3 |
| #define BITSHIFTSIZE 2 |
| |
| #define TYPE_S8 1 |
| #define TYPE_U8 2 |
| #define TYPE_S16HL 3 |
| #define TYPE_U16HL 4 |
| #define TYPE_S16LH 5 |
| #define TYPE_U16LH 6 |
| |
| #define NWRAP 3 |
| #define NSKIPSIZE 1 |
| |
| #define LPCQUANT 5 |
| #define V2LPCQOFFSET (1 << LPCQUANT) |
| |
| #define FNSIZE 2 |
| #define FN_DIFF0 0 |
| #define FN_DIFF1 1 |
| #define FN_DIFF2 2 |
| #define FN_DIFF3 3 |
| #define FN_QUIT 4 |
| #define FN_BLOCKSIZE 5 |
| #define FN_BITSHIFT 6 |
| #define FN_QLPC 7 |
| #define FN_ZERO 8 |
| #define FN_VERBATIM 9 |
| |
| /** indicates if the FN_* command is audio or non-audio */ |
| static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 }; |
| |
| #define VERBATIM_CKSIZE_SIZE 5 |
| #define VERBATIM_BYTE_SIZE 8 |
| #define CANONICAL_HEADER_SIZE 44 |
| |
| typedef struct ShortenContext { |
| AVCodecContext *avctx; |
| GetBitContext gb; |
| |
| int min_framesize, max_framesize; |
| unsigned channels; |
| |
| int32_t *decoded[MAX_CHANNELS]; |
| int32_t *decoded_base[MAX_CHANNELS]; |
| int32_t *offset[MAX_CHANNELS]; |
| int *coeffs; |
| uint8_t *bitstream; |
| int bitstream_size; |
| int bitstream_index; |
| unsigned int allocated_bitstream_size; |
| int header_size; |
| uint8_t header[OUT_BUFFER_SIZE]; |
| int version; |
| int cur_chan; |
| int bitshift; |
| int nmean; |
| int internal_ftype; |
| int nwrap; |
| int blocksize; |
| int bitindex; |
| int32_t lpcqoffset; |
| int got_header; |
| int got_quit_command; |
| } ShortenContext; |
| |
| static av_cold int shorten_decode_init(AVCodecContext *avctx) |
| { |
| ShortenContext *s = avctx->priv_data; |
| s->avctx = avctx; |
| |
| return 0; |
| } |
| |
| static int allocate_buffers(ShortenContext *s) |
| { |
| int i, chan, err; |
| |
| for (chan = 0; chan < s->channels; chan++) { |
| if (FFMAX(1, s->nmean) >= UINT_MAX / sizeof(int32_t)) { |
| av_log(s->avctx, AV_LOG_ERROR, "nmean too large\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| if (s->blocksize + (uint64_t)s->nwrap >= UINT_MAX / sizeof(int32_t)) { |
| av_log(s->avctx, AV_LOG_ERROR, |
| "s->blocksize + s->nwrap too large\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if ((err = av_reallocp_array(&s->offset[chan], |
| sizeof(int32_t), |
| FFMAX(1, s->nmean))) < 0) |
| return err; |
| |
| if ((err = av_reallocp_array(&s->decoded_base[chan], (s->blocksize + s->nwrap), |
| sizeof(s->decoded_base[0][0]))) < 0) |
| return err; |
| for (i = 0; i < s->nwrap; i++) |
| s->decoded_base[chan][i] = 0; |
| s->decoded[chan] = s->decoded_base[chan] + s->nwrap; |
| } |
| |
| if ((err = av_reallocp_array(&s->coeffs, s->nwrap, sizeof(*s->coeffs))) < 0) |
| return err; |
| |
| return 0; |
| } |
| |
| static inline unsigned int get_uint(ShortenContext *s, int k) |
| { |
| if (s->version != 0) |
| k = get_ur_golomb_shorten(&s->gb, ULONGSIZE); |
| return get_ur_golomb_shorten(&s->gb, k); |
| } |
| |
| static void fix_bitshift(ShortenContext *s, int32_t *buffer) |
| { |
| int i; |
| |
| if (s->bitshift != 0) |
| for (i = 0; i < s->blocksize; i++) |
| buffer[i] <<= s->bitshift; |
| } |
| |
| static int init_offset(ShortenContext *s) |
| { |
| int32_t mean = 0; |
| int chan, i; |
| int nblock = FFMAX(1, s->nmean); |
| /* initialise offset */ |
| switch (s->internal_ftype) { |
| case TYPE_U8: |
| s->avctx->sample_fmt = AV_SAMPLE_FMT_U8P; |
| mean = 0x80; |
| break; |
| case TYPE_S16HL: |
| case TYPE_S16LH: |
| s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P; |
| break; |
| default: |
| av_log(s->avctx, AV_LOG_ERROR, "unknown audio type\n"); |
| return AVERROR_PATCHWELCOME; |
| } |
| |
| for (chan = 0; chan < s->channels; chan++) |
| for (i = 0; i < nblock; i++) |
| s->offset[chan][i] = mean; |
| return 0; |
| } |
| |
| static int decode_wave_header(AVCodecContext *avctx, const uint8_t *header, |
| int header_size) |
| { |
| int len, bps; |
| short wave_format; |
| GetByteContext gb; |
| |
| bytestream2_init(&gb, header, header_size); |
| |
| if (bytestream2_get_le32(&gb) != MKTAG('R', 'I', 'F', 'F')) { |
| av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| bytestream2_skip(&gb, 4); /* chunk size */ |
| |
| if (bytestream2_get_le32(&gb) != MKTAG('W', 'A', 'V', 'E')) { |
| av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| while (bytestream2_get_le32(&gb) != MKTAG('f', 'm', 't', ' ')) { |
| len = bytestream2_get_le32(&gb); |
| bytestream2_skip(&gb, len); |
| if (len < 0 || bytestream2_get_bytes_left(&gb) < 16) { |
| av_log(avctx, AV_LOG_ERROR, "no fmt chunk found\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| } |
| len = bytestream2_get_le32(&gb); |
| |
| if (len < 16) { |
| av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| wave_format = bytestream2_get_le16(&gb); |
| |
| switch (wave_format) { |
| case WAVE_FORMAT_PCM: |
| break; |
| default: |
| av_log(avctx, AV_LOG_ERROR, "unsupported wave format\n"); |
| return AVERROR(ENOSYS); |
| } |
| |
| bytestream2_skip(&gb, 2); // skip channels (already got from shorten header) |
| avctx->sample_rate = bytestream2_get_le32(&gb); |
| bytestream2_skip(&gb, 4); // skip bit rate (represents original uncompressed bit rate) |
| bytestream2_skip(&gb, 2); // skip block align (not needed) |
| bps = bytestream2_get_le16(&gb); |
| avctx->bits_per_coded_sample = bps; |
| |
| if (bps != 16 && bps != 8) { |
| av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample: %d\n", bps); |
| return AVERROR(ENOSYS); |
| } |
| |
| len -= 16; |
| if (len > 0) |
| av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len); |
| |
| return 0; |
| } |
| |
| static const int fixed_coeffs[][3] = { |
| { 0, 0, 0 }, |
| { 1, 0, 0 }, |
| { 2, -1, 0 }, |
| { 3, -3, 1 } |
| }; |
| |
| static int decode_subframe_lpc(ShortenContext *s, int command, int channel, |
| int residual_size, int32_t coffset) |
| { |
| int pred_order, sum, qshift, init_sum, i, j; |
| const int *coeffs; |
| |
| if (command == FN_QLPC) { |
| /* read/validate prediction order */ |
| pred_order = get_ur_golomb_shorten(&s->gb, LPCQSIZE); |
| if ((unsigned)pred_order > s->nwrap) { |
| av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n", |
| pred_order); |
| return AVERROR(EINVAL); |
| } |
| /* read LPC coefficients */ |
| for (i = 0; i < pred_order; i++) |
| s->coeffs[i] = get_sr_golomb_shorten(&s->gb, LPCQUANT); |
| coeffs = s->coeffs; |
| |
| qshift = LPCQUANT; |
| } else { |
| /* fixed LPC coeffs */ |
| pred_order = command; |
| if (pred_order >= FF_ARRAY_ELEMS(fixed_coeffs)) { |
| av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n", |
| pred_order); |
| return AVERROR_INVALIDDATA; |
| } |
| coeffs = fixed_coeffs[pred_order]; |
| qshift = 0; |
| } |
| |
| /* subtract offset from previous samples to use in prediction */ |
| if (command == FN_QLPC && coffset) |
| for (i = -pred_order; i < 0; i++) |
| s->decoded[channel][i] -= coffset; |
| |
| /* decode residual and do LPC prediction */ |
| init_sum = pred_order ? (command == FN_QLPC ? s->lpcqoffset : 0) : coffset; |
| for (i = 0; i < s->blocksize; i++) { |
| sum = init_sum; |
| for (j = 0; j < pred_order; j++) |
| sum += coeffs[j] * s->decoded[channel][i - j - 1]; |
| s->decoded[channel][i] = get_sr_golomb_shorten(&s->gb, residual_size) + |
| (sum >> qshift); |
| } |
| |
| /* add offset to current samples */ |
| if (command == FN_QLPC && coffset) |
| for (i = 0; i < s->blocksize; i++) |
| s->decoded[channel][i] += coffset; |
| |
| return 0; |
| } |
| |
| static int read_header(ShortenContext *s) |
| { |
| int i, ret; |
| int maxnlpc = 0; |
| /* shorten signature */ |
| if (get_bits_long(&s->gb, 32) != AV_RB32("ajkg")) { |
| av_log(s->avctx, AV_LOG_ERROR, "missing shorten magic 'ajkg'\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| s->lpcqoffset = 0; |
| s->blocksize = DEFAULT_BLOCK_SIZE; |
| s->nmean = -1; |
| s->version = get_bits(&s->gb, 8); |
| s->internal_ftype = get_uint(s, TYPESIZE); |
| |
| s->channels = get_uint(s, CHANSIZE); |
| if (!s->channels) { |
| av_log(s->avctx, AV_LOG_ERROR, "No channels reported\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| if (s->channels > MAX_CHANNELS) { |
| av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels); |
| s->channels = 0; |
| return AVERROR_INVALIDDATA; |
| } |
| s->avctx->channels = s->channels; |
| |
| /* get blocksize if version > 0 */ |
| if (s->version > 0) { |
| int skip_bytes; |
| unsigned blocksize; |
| |
| blocksize = get_uint(s, av_log2(DEFAULT_BLOCK_SIZE)); |
| if (!blocksize || blocksize > MAX_BLOCKSIZE) { |
| av_log(s->avctx, AV_LOG_ERROR, |
| "invalid or unsupported block size: %d\n", |
| blocksize); |
| return AVERROR(EINVAL); |
| } |
| s->blocksize = blocksize; |
| |
| maxnlpc = get_uint(s, LPCQSIZE); |
| s->nmean = get_uint(s, 0); |
| |
| skip_bytes = get_uint(s, NSKIPSIZE); |
| if ((unsigned)skip_bytes > get_bits_left(&s->gb)/8) { |
| av_log(s->avctx, AV_LOG_ERROR, "invalid skip_bytes: %d\n", skip_bytes); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| for (i = 0; i < skip_bytes; i++) |
| skip_bits(&s->gb, 8); |
| } |
| s->nwrap = FFMAX(NWRAP, maxnlpc); |
| |
| if ((ret = allocate_buffers(s)) < 0) |
| return ret; |
| |
| if ((ret = init_offset(s)) < 0) |
| return ret; |
| |
| if (s->version > 1) |
| s->lpcqoffset = V2LPCQOFFSET; |
| |
| if (get_ur_golomb_shorten(&s->gb, FNSIZE) != FN_VERBATIM) { |
| av_log(s->avctx, AV_LOG_ERROR, |
| "missing verbatim section at beginning of stream\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| s->header_size = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE); |
| if (s->header_size >= OUT_BUFFER_SIZE || |
| s->header_size < CANONICAL_HEADER_SIZE) { |
| av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n", |
| s->header_size); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| for (i = 0; i < s->header_size; i++) |
| s->header[i] = (char)get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE); |
| |
| if ((ret = decode_wave_header(s->avctx, s->header, s->header_size)) < 0) |
| return ret; |
| |
| s->cur_chan = 0; |
| s->bitshift = 0; |
| |
| s->got_header = 1; |
| |
| return 0; |
| } |
| |
| static int shorten_decode_frame(AVCodecContext *avctx, void *data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| AVFrame *frame = data; |
| const uint8_t *buf = avpkt->data; |
| int buf_size = avpkt->size; |
| ShortenContext *s = avctx->priv_data; |
| int i, input_buf_size = 0; |
| int ret; |
| |
| /* allocate internal bitstream buffer */ |
| if (s->max_framesize == 0) { |
| void *tmp_ptr; |
| s->max_framesize = 8192; // should hopefully be enough for the first header |
| tmp_ptr = av_fast_realloc(s->bitstream, &s->allocated_bitstream_size, |
| s->max_framesize + AV_INPUT_BUFFER_PADDING_SIZE); |
| if (!tmp_ptr) { |
| av_log(avctx, AV_LOG_ERROR, "error allocating bitstream buffer\n"); |
| return AVERROR(ENOMEM); |
| } |
| memset(tmp_ptr, 0, s->allocated_bitstream_size); |
| s->bitstream = tmp_ptr; |
| } |
| |
| /* append current packet data to bitstream buffer */ |
| if (1 && s->max_framesize) { //FIXME truncated |
| buf_size = FFMIN(buf_size, s->max_framesize - s->bitstream_size); |
| input_buf_size = buf_size; |
| |
| if (s->bitstream_index + s->bitstream_size + buf_size + AV_INPUT_BUFFER_PADDING_SIZE > |
| s->allocated_bitstream_size) { |
| memmove(s->bitstream, &s->bitstream[s->bitstream_index], |
| s->bitstream_size); |
| s->bitstream_index = 0; |
| } |
| if (buf) |
| memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf, |
| buf_size); |
| buf = &s->bitstream[s->bitstream_index]; |
| buf_size += s->bitstream_size; |
| s->bitstream_size = buf_size; |
| |
| /* do not decode until buffer has at least max_framesize bytes or |
| * the end of the file has been reached */ |
| if (buf_size < s->max_framesize && avpkt->data) { |
| *got_frame_ptr = 0; |
| return input_buf_size; |
| } |
| } |
| /* init and position bitstream reader */ |
| if ((ret = init_get_bits8(&s->gb, buf, buf_size)) < 0) |
| return ret; |
| skip_bits(&s->gb, s->bitindex); |
| |
| /* process header or next subblock */ |
| if (!s->got_header) { |
| if ((ret = read_header(s)) < 0) |
| return ret; |
| *got_frame_ptr = 0; |
| goto finish_frame; |
| } |
| |
| /* if quit command was read previously, don't decode anything */ |
| if (s->got_quit_command) { |
| *got_frame_ptr = 0; |
| return avpkt->size; |
| } |
| |
| s->cur_chan = 0; |
| while (s->cur_chan < s->channels) { |
| unsigned cmd; |
| int len; |
| |
| if (get_bits_left(&s->gb) < 3 + FNSIZE) { |
| *got_frame_ptr = 0; |
| break; |
| } |
| |
| cmd = get_ur_golomb_shorten(&s->gb, FNSIZE); |
| |
| if (cmd > FN_VERBATIM) { |
| av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd); |
| *got_frame_ptr = 0; |
| break; |
| } |
| |
| if (!is_audio_command[cmd]) { |
| /* process non-audio command */ |
| switch (cmd) { |
| case FN_VERBATIM: |
| len = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE); |
| while (len--) |
| get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE); |
| break; |
| case FN_BITSHIFT: { |
| unsigned bitshift = get_ur_golomb_shorten(&s->gb, BITSHIFTSIZE); |
| if (bitshift > 31) { |
| av_log(avctx, AV_LOG_ERROR, "bitshift %d is invalid\n", |
| bitshift); |
| return AVERROR_INVALIDDATA; |
| } |
| s->bitshift = bitshift; |
| break; |
| } |
| case FN_BLOCKSIZE: { |
| unsigned blocksize = get_uint(s, av_log2(s->blocksize)); |
| if (blocksize > s->blocksize) { |
| av_log(avctx, AV_LOG_ERROR, |
| "Increasing block size is not supported\n"); |
| return AVERROR_PATCHWELCOME; |
| } |
| if (!blocksize || blocksize > MAX_BLOCKSIZE) { |
| av_log(avctx, AV_LOG_ERROR, "invalid or unsupported " |
| "block size: %d\n", blocksize); |
| return AVERROR(EINVAL); |
| } |
| s->blocksize = blocksize; |
| break; |
| } |
| case FN_QUIT: |
| s->got_quit_command = 1; |
| break; |
| } |
| if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) { |
| *got_frame_ptr = 0; |
| break; |
| } |
| } else { |
| /* process audio command */ |
| int residual_size = 0; |
| int channel = s->cur_chan; |
| int32_t coffset; |
| |
| /* get Rice code for residual decoding */ |
| if (cmd != FN_ZERO) { |
| residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE); |
| /* This is a hack as version 0 differed in the definition |
| * of get_sr_golomb_shorten(). */ |
| if (s->version == 0) |
| residual_size--; |
| } |
| |
| /* calculate sample offset using means from previous blocks */ |
| if (s->nmean == 0) |
| coffset = s->offset[channel][0]; |
| else { |
| int32_t sum = (s->version < 2) ? 0 : s->nmean / 2; |
| for (i = 0; i < s->nmean; i++) |
| sum += s->offset[channel][i]; |
| coffset = sum / s->nmean; |
| if (s->version >= 2) |
| coffset = s->bitshift == 0 ? coffset : coffset >> s->bitshift - 1 >> 1; |
| } |
| |
| /* decode samples for this channel */ |
| if (cmd == FN_ZERO) { |
| for (i = 0; i < s->blocksize; i++) |
| s->decoded[channel][i] = 0; |
| } else { |
| if ((ret = decode_subframe_lpc(s, cmd, channel, |
| residual_size, coffset)) < 0) |
| return ret; |
| } |
| |
| /* update means with info from the current block */ |
| if (s->nmean > 0) { |
| int32_t sum = (s->version < 2) ? 0 : s->blocksize / 2; |
| for (i = 0; i < s->blocksize; i++) |
| sum += s->decoded[channel][i]; |
| |
| for (i = 1; i < s->nmean; i++) |
| s->offset[channel][i - 1] = s->offset[channel][i]; |
| |
| if (s->version < 2) |
| s->offset[channel][s->nmean - 1] = sum / s->blocksize; |
| else |
| s->offset[channel][s->nmean - 1] = (sum / s->blocksize) << s->bitshift; |
| } |
| |
| /* copy wrap samples for use with next block */ |
| for (i = -s->nwrap; i < 0; i++) |
| s->decoded[channel][i] = s->decoded[channel][i + s->blocksize]; |
| |
| /* shift samples to add in unused zero bits which were removed |
| * during encoding */ |
| fix_bitshift(s, s->decoded[channel]); |
| |
| /* if this is the last channel in the block, output the samples */ |
| s->cur_chan++; |
| if (s->cur_chan == s->channels) { |
| uint8_t *samples_u8; |
| int16_t *samples_s16; |
| int chan; |
| |
| /* get output buffer */ |
| frame->nb_samples = s->blocksize; |
| if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| return ret; |
| |
| for (chan = 0; chan < s->channels; chan++) { |
| samples_u8 = ((uint8_t **)frame->extended_data)[chan]; |
| samples_s16 = ((int16_t **)frame->extended_data)[chan]; |
| for (i = 0; i < s->blocksize; i++) { |
| switch (s->internal_ftype) { |
| case TYPE_U8: |
| *samples_u8++ = av_clip_uint8(s->decoded[chan][i]); |
| break; |
| case TYPE_S16HL: |
| case TYPE_S16LH: |
| *samples_s16++ = av_clip_int16(s->decoded[chan][i]); |
| break; |
| } |
| } |
| } |
| |
| *got_frame_ptr = 1; |
| } |
| } |
| } |
| if (s->cur_chan < s->channels) |
| *got_frame_ptr = 0; |
| |
| finish_frame: |
| s->bitindex = get_bits_count(&s->gb) - 8 * (get_bits_count(&s->gb) / 8); |
| i = get_bits_count(&s->gb) / 8; |
| if (i > buf_size) { |
| av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size); |
| s->bitstream_size = 0; |
| s->bitstream_index = 0; |
| return AVERROR_INVALIDDATA; |
| } |
| if (s->bitstream_size) { |
| s->bitstream_index += i; |
| s->bitstream_size -= i; |
| return input_buf_size; |
| } else |
| return i; |
| } |
| |
| static av_cold int shorten_decode_close(AVCodecContext *avctx) |
| { |
| ShortenContext *s = avctx->priv_data; |
| int i; |
| |
| for (i = 0; i < s->channels; i++) { |
| s->decoded[i] = NULL; |
| av_freep(&s->decoded_base[i]); |
| av_freep(&s->offset[i]); |
| } |
| av_freep(&s->bitstream); |
| av_freep(&s->coeffs); |
| |
| return 0; |
| } |
| |
| AVCodec ff_shorten_decoder = { |
| .name = "shorten", |
| .long_name = NULL_IF_CONFIG_SMALL("Shorten"), |
| .type = AVMEDIA_TYPE_AUDIO, |
| .id = AV_CODEC_ID_SHORTEN, |
| .priv_data_size = sizeof(ShortenContext), |
| .init = shorten_decode_init, |
| .close = shorten_decode_close, |
| .decode = shorten_decode_frame, |
| .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_DR1, |
| .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, |
| AV_SAMPLE_FMT_U8P, |
| AV_SAMPLE_FMT_NONE }, |
| }; |