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e4de71677f
* qatar/master: aac_latm: reconfigure decoder on audio specific config changes latmdec: fix audio specific config parsing Add avcodec_decode_audio4(). avcodec: change number of plane pointers from 4 to 8 at next major bump. Update developers documentation with coding conventions. svq1dec: avoid undefined get_bits(0) call ARM: h264dsp_neon cosmetics ARM: make some NEON macros reusable Do not memcpy raw video frames when using null muxer fate: update asf seektest vp8: flush buffers on size changes. doc: improve general documentation for MacOSX asf: use packet dts as approximation of pts asf: do not call av_read_frame rtsp: Initialize the media_type_mask in the rtp guessing demuxer Cleaned up alacenc.c Conflicts: doc/APIchanges doc/developer.texi libavcodec/8svx.c libavcodec/aacdec.c libavcodec/ac3dec.c libavcodec/avcodec.h libavcodec/nellymoserdec.c libavcodec/tta.c libavcodec/utils.c libavcodec/version.h libavcodec/wmadec.c libavformat/asfdec.c tests/ref/seek/lavf_asf Merged-by: Michael Niedermayer <michaelni@gmx.at>
195 lines
5.8 KiB
C
195 lines
5.8 KiB
C
/*
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* Westwood SNDx codecs
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* Copyright (c) 2005 Konstantin Shishkov
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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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#include <stdint.h>
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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/**
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* @file
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* Westwood SNDx codecs
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*
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* Reference documents about VQA format and its audio codecs
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* can be found here:
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* http://www.multimedia.cx
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*/
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static const int8_t ws_adpcm_4bit[] = {
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-9, -8, -6, -5, -4, -3, -2, -1,
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0, 1, 2, 3, 4, 5, 6, 8
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};
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typedef struct WSSndContext {
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AVFrame frame;
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} WSSndContext;
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static av_cold int ws_snd_decode_init(AVCodecContext *avctx)
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{
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WSSndContext *s = avctx->priv_data;
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if (avctx->channels != 1) {
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av_log_ask_for_sample(avctx, "unsupported number of channels\n");
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return AVERROR(EINVAL);
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_U8;
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avcodec_get_frame_defaults(&s->frame);
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avctx->coded_frame = &s->frame;
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return 0;
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}
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static int ws_snd_decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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WSSndContext *s = avctx->priv_data;
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const uint8_t *buf = avpkt->data;
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int buf_size = avpkt->size;
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int in_size, out_size, ret;
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int sample = 128;
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uint8_t *samples;
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uint8_t *samples_end;
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if (!buf_size)
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return 0;
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if (buf_size < 4) {
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av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
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return AVERROR(EINVAL);
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}
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out_size = AV_RL16(&buf[0]);
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in_size = AV_RL16(&buf[2]);
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buf += 4;
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if (in_size > buf_size) {
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av_log(avctx, AV_LOG_ERROR, "Frame data is larger than input buffer\n");
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return -1;
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}
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/* get output buffer */
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s->frame.nb_samples = out_size;
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if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
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av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
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return ret;
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}
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samples = s->frame.data[0];
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samples_end = samples + out_size;
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if (in_size == out_size) {
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memcpy(samples, buf, out_size);
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*got_frame_ptr = 1;
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*(AVFrame *)data = s->frame;
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return buf_size;
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}
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while (samples < samples_end && buf - avpkt->data < buf_size) {
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int code, smp, size;
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uint8_t count;
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code = *buf >> 6;
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count = *buf & 0x3F;
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buf++;
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/* make sure we don't write past the output buffer */
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switch (code) {
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case 0: smp = 4; break;
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case 1: smp = 2; break;
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case 2: smp = (count & 0x20) ? 1 : count + 1; break;
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default: smp = count + 1; break;
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}
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if (samples_end - samples < smp)
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break;
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/* make sure we don't read past the input buffer */
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size = ((code == 2 && (count & 0x20)) || code == 3) ? 0 : count + 1;
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if ((buf - avpkt->data) + size > buf_size)
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break;
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switch (code) {
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case 0: /* ADPCM 2-bit */
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for (count++; count > 0; count--) {
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code = *buf++;
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sample += ( code & 0x3) - 2;
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sample = av_clip_uint8(sample);
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*samples++ = sample;
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sample += ((code >> 2) & 0x3) - 2;
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sample = av_clip_uint8(sample);
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*samples++ = sample;
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sample += ((code >> 4) & 0x3) - 2;
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sample = av_clip_uint8(sample);
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*samples++ = sample;
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sample += (code >> 6) - 2;
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sample = av_clip_uint8(sample);
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*samples++ = sample;
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}
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break;
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case 1: /* ADPCM 4-bit */
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for (count++; count > 0; count--) {
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code = *buf++;
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sample += ws_adpcm_4bit[code & 0xF];
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sample = av_clip_uint8(sample);
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*samples++ = sample;
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sample += ws_adpcm_4bit[code >> 4];
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sample = av_clip_uint8(sample);
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*samples++ = sample;
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}
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break;
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case 2: /* no compression */
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if (count & 0x20) { /* big delta */
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int8_t t;
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t = count;
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t <<= 3;
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sample += t >> 3;
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sample = av_clip_uint8(sample);
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*samples++ = sample;
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} else { /* copy */
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memcpy(samples, buf, smp);
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samples += smp;
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buf += smp;
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sample = buf[-1];
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}
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break;
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default: /* run */
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memset(samples, sample, smp);
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samples += smp;
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}
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}
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s->frame.nb_samples = samples - s->frame.data[0];
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*got_frame_ptr = 1;
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*(AVFrame *)data = s->frame;
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return buf_size;
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}
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AVCodec ff_ws_snd1_decoder = {
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.name = "ws_snd1",
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.type = AVMEDIA_TYPE_AUDIO,
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.id = CODEC_ID_WESTWOOD_SND1,
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.priv_data_size = sizeof(WSSndContext),
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.init = ws_snd_decode_init,
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.decode = ws_snd_decode_frame,
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.capabilities = CODEC_CAP_DR1,
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.long_name = NULL_IF_CONFIG_SMALL("Westwood Audio (SND1)"),
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};
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