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https://github.com/librempeg/librempeg
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dbb7009ea5
"Copyright (c) 2003 The ffmpeg Project" and "part of libav" is ROTFL Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
181 lines
6.0 KiB
C
181 lines
6.0 KiB
C
/*
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* Westwood Studios AUD Format Demuxer
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* Copyright (c) 2003 The ffmpeg Project
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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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* Westwood Studios AUD file demuxer
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* by Mike Melanson (melanson@pcisys.net)
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* for more information on the Westwood file formats, visit:
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* http://www.pcisys.net/~melanson/codecs/
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* http://www.geocities.com/SiliconValley/8682/aud3.txt
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*
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* Implementation note: There is no definite file signature for AUD files.
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* The demuxer uses a probabilistic strategy for content detection. This
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* entails performing sanity checks on certain header values in order to
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* qualify a file. Refer to wsaud_probe() for the precise parameters.
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*/
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "internal.h"
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#define AUD_HEADER_SIZE 12
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#define AUD_CHUNK_PREAMBLE_SIZE 8
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#define AUD_CHUNK_SIGNATURE 0x0000DEAF
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static int wsaud_probe(AVProbeData *p)
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{
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int field;
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/* Probabilistic content detection strategy: There is no file signature
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* so perform sanity checks on various header parameters:
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* 8000 <= sample rate (16 bits) <= 48000 ==> 40001 acceptable numbers
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* flags <= 0x03 (2 LSBs are used) ==> 4 acceptable numbers
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* compression type (8 bits) = 1 or 99 ==> 2 acceptable numbers
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* first audio chunk signature (32 bits) ==> 1 acceptable number
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* The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
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* 320008 acceptable number combinations.
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*/
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if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
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return 0;
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/* check sample rate */
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field = AV_RL16(&p->buf[0]);
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if ((field < 8000) || (field > 48000))
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return 0;
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/* enforce the rule that the top 6 bits of this flags field are reserved (0);
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* this might not be true, but enforce it until deemed unnecessary */
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if (p->buf[10] & 0xFC)
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return 0;
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/* note: only check for WS IMA (type 99) right now since there is no
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* support for type 1 */
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if (p->buf[11] != 99 && p->buf[11] != 1)
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return 0;
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/* read ahead to the first audio chunk and validate the first header signature */
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if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
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return 0;
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/* return 1/2 certainty since this file check is a little sketchy */
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return AVPROBE_SCORE_MAX / 2;
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}
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static int wsaud_read_header(AVFormatContext *s)
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{
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AVIOContext *pb = s->pb;
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AVStream *st;
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unsigned char header[AUD_HEADER_SIZE];
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int sample_rate, channels, codec;
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if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
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return AVERROR(EIO);
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sample_rate = AV_RL16(&header[0]);
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channels = (header[10] & 0x1) + 1;
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codec = header[11];
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/* initialize the audio decoder stream */
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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switch (codec) {
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case 1:
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if (channels != 1) {
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av_log_ask_for_sample(s, "Stereo WS-SND1 is not supported.\n");
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return AVERROR_PATCHWELCOME;
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}
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st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
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break;
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case 99:
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st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
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st->codec->bits_per_coded_sample = 4;
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st->codec->bit_rate = channels * sample_rate * 4;
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break;
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default:
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av_log_ask_for_sample(s, "Unknown codec: %d\n", codec);
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return AVERROR_PATCHWELCOME;
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}
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avpriv_set_pts_info(st, 64, 1, sample_rate);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->channels = channels;
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st->codec->sample_rate = sample_rate;
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return 0;
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}
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static int wsaud_read_packet(AVFormatContext *s,
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AVPacket *pkt)
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{
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AVIOContext *pb = s->pb;
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unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
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unsigned int chunk_size;
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int ret = 0;
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AVStream *st = s->streams[0];
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if (avio_read(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
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AUD_CHUNK_PREAMBLE_SIZE)
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return AVERROR(EIO);
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/* validate the chunk */
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if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
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return AVERROR_INVALIDDATA;
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chunk_size = AV_RL16(&preamble[0]);
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if (st->codec->codec_id == CODEC_ID_WESTWOOD_SND1) {
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/* For Westwood SND1 audio we need to add the output size and input
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size to the start of the packet to match what is in VQA.
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Specifically, this is needed to signal when a packet should be
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decoding as raw 8-bit pcm or variable-size ADPCM. */
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int out_size = AV_RL16(&preamble[2]);
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if ((ret = av_new_packet(pkt, chunk_size + 4)))
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return ret;
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if ((ret = avio_read(pb, &pkt->data[4], chunk_size)) != chunk_size)
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return ret < 0 ? ret : AVERROR(EIO);
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AV_WL16(&pkt->data[0], out_size);
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AV_WL16(&pkt->data[2], chunk_size);
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pkt->duration = out_size;
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} else {
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ret = av_get_packet(pb, pkt, chunk_size);
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if (ret != chunk_size)
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return AVERROR(EIO);
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/* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
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pkt->duration = (chunk_size * 2) / st->codec->channels;
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}
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pkt->stream_index = st->index;
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return ret;
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}
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AVInputFormat ff_wsaud_demuxer = {
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.name = "wsaud",
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.long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
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.read_probe = wsaud_probe,
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.read_header = wsaud_read_header,
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.read_packet = wsaud_read_packet,
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};
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