westwood: split the AUD and VQA demuxers into separate files.
They have no code in common.
This commit is contained in:
parent
15cea3695d
commit
e924a4ba71
@ -308,8 +308,8 @@ OBJS-$(CONFIG_WC3_DEMUXER) += wc3movie.o
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OBJS-$(CONFIG_WEBM_MUXER) += matroskaenc.o matroska.o \
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riff.o isom.o avc.o \
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flacenc_header.o avlanguage.o
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OBJS-$(CONFIG_WSAUD_DEMUXER) += westwood.o
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OBJS-$(CONFIG_WSVQA_DEMUXER) += westwood.o
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OBJS-$(CONFIG_WSAUD_DEMUXER) += westwood_aud.o
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OBJS-$(CONFIG_WSVQA_DEMUXER) += westwood_vqa.o
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OBJS-$(CONFIG_WTV_DEMUXER) += wtv.o asfdec.o asf.o asfcrypt.o \
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avlanguage.o mpegts.o isom.o riff.o
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OBJS-$(CONFIG_WV_DEMUXER) += wv.o apetag.o
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173
libavformat/westwood_aud.c
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173
libavformat/westwood_aud.c
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@ -0,0 +1,173 @@
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/*
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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 Libav.
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*
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* Libav 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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* Libav 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 Libav; 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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typedef struct WsAudDemuxContext {
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int audio_samplerate;
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int audio_channels;
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int audio_bits;
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enum CodecID audio_type;
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int audio_stream_index;
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int64_t audio_frame_counter;
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} WsAudDemuxContext;
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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)
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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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AVFormatParameters *ap)
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{
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WsAudDemuxContext *wsaud = s->priv_data;
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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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if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
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return AVERROR(EIO);
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wsaud->audio_samplerate = AV_RL16(&header[0]);
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if (header[11] == 99)
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wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
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else
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return AVERROR_INVALIDDATA;
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/* flag 0 indicates stereo */
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wsaud->audio_channels = (header[10] & 0x1) + 1;
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/* flag 1 indicates 16 bit audio */
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wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
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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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avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = wsaud->audio_type;
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st->codec->codec_tag = 0; /* no tag */
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st->codec->channels = wsaud->audio_channels;
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st->codec->sample_rate = wsaud->audio_samplerate;
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st->codec->bits_per_coded_sample = wsaud->audio_bits;
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st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
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st->codec->bits_per_coded_sample / 4;
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st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
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wsaud->audio_stream_index = st->index;
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wsaud->audio_frame_counter = 0;
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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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WsAudDemuxContext *wsaud = s->priv_data;
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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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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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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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pkt->stream_index = wsaud->audio_stream_index;
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pkt->pts = wsaud->audio_frame_counter;
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pkt->pts /= wsaud->audio_samplerate;
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/* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
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wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
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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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.priv_data_size = sizeof(WsAudDemuxContext),
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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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@ -1,5 +1,5 @@
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/*
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* Westwood Studios Multimedia Formats Demuxer (VQA, AUD)
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* Westwood Studios VQA 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 Libav.
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@ -21,26 +21,17 @@
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/**
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* @file
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* Westwood Studios VQA & AUD file demuxers
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* Westwood Studios VQA 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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#define FORM_TAG MKBETAG('F', 'O', 'R', 'M')
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#define WVQA_TAG MKBETAG('W', 'V', 'Q', 'A')
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#define VQHD_TAG MKBETAG('V', 'Q', 'H', 'D')
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@ -63,15 +54,6 @@
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#define VQA_FRAMERATE 15
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#define VQA_PREAMBLE_SIZE 8
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typedef struct WsAudDemuxContext {
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int audio_samplerate;
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int audio_channels;
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int audio_bits;
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enum CodecID audio_type;
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int audio_stream_index;
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int64_t audio_frame_counter;
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} WsAudDemuxContext;
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typedef struct WsVqaDemuxContext {
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int audio_samplerate;
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int audio_channels;
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@ -83,119 +65,6 @@ typedef struct WsVqaDemuxContext {
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int64_t audio_frame_counter;
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} WsVqaDemuxContext;
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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)
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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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AVFormatParameters *ap)
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{
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WsAudDemuxContext *wsaud = s->priv_data;
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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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if (avio_read(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
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return AVERROR(EIO);
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wsaud->audio_samplerate = AV_RL16(&header[0]);
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if (header[11] == 99)
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wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
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else
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return AVERROR_INVALIDDATA;
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/* flag 0 indicates stereo */
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wsaud->audio_channels = (header[10] & 0x1) + 1;
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/* flag 1 indicates 16 bit audio */
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wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;
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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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avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = wsaud->audio_type;
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st->codec->codec_tag = 0; /* no tag */
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st->codec->channels = wsaud->audio_channels;
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st->codec->sample_rate = wsaud->audio_samplerate;
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st->codec->bits_per_coded_sample = wsaud->audio_bits;
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st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
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st->codec->bits_per_coded_sample / 4;
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st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;
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wsaud->audio_stream_index = st->index;
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wsaud->audio_frame_counter = 0;
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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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WsAudDemuxContext *wsaud = s->priv_data;
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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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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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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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pkt->stream_index = wsaud->audio_stream_index;
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pkt->pts = wsaud->audio_frame_counter;
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pkt->pts /= wsaud->audio_samplerate;
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/* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
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wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;
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return ret;
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}
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static int wsvqa_probe(AVProbeData *p)
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{
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/* need 12 bytes to qualify */
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@ -372,17 +241,6 @@ static int wsvqa_read_packet(AVFormatContext *s,
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return ret;
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}
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#if CONFIG_WSAUD_DEMUXER
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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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.priv_data_size = sizeof(WsAudDemuxContext),
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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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#endif
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#if CONFIG_WSVQA_DEMUXER
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AVInputFormat ff_wsvqa_demuxer = {
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.name = "wsvqa",
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.long_name = NULL_IF_CONFIG_SMALL("Westwood Studios VQA format"),
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@ -391,4 +249,3 @@ AVInputFormat ff_wsvqa_demuxer = {
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.read_header = wsvqa_read_header,
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.read_packet = wsvqa_read_packet,
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};
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#endif
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