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Currently, AVStream contains an embedded AVCodecContext instance, which is used by demuxers to export stream parameters to the caller and by muxers to receive stream parameters from the caller. It is also used internally as the codec context that is passed to parsers. In addition, it is also widely used by the callers as the decoding (when demuxer) or encoding (when muxing) context, though this has been officially discouraged since Libav 11. There are multiple important problems with this approach: - the fields in AVCodecContext are in general one of * stream parameters * codec options * codec state However, it's not clear which ones are which. It is consequently unclear which fields are a demuxer allowed to set or a muxer allowed to read. This leads to erratic behaviour depending on whether decoding or encoding is being performed or not (and whether it uses the AVStream embedded codec context). - various synchronization issues arising from the fact that the same context is used by several different APIs (muxers/demuxers, parsers, bitstream filters and encoders/decoders) simultaneously, with there being no clear rules for who can modify what and the different processes being typically delayed with respect to each other. - avformat_find_stream_info() making it necessary to support opening and closing a single codec context multiple times, thus complicating the semantics of freeing various allocated objects in the codec context. Those problems are resolved by replacing the AVStream embedded codec context with a newly added AVCodecParameters instance, which stores only the stream parameters exported by the demuxers or read by the muxers.
184 lines
6.2 KiB
C
184 lines
6.2 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 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/channel_layout.h"
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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_EXTENSION;
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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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avpriv_request_sample(s, "Stereo WS-SND1");
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return AVERROR_PATCHWELCOME;
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}
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st->codecpar->codec_id = AV_CODEC_ID_WESTWOOD_SND1;
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break;
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case 99:
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st->codecpar->codec_id = AV_CODEC_ID_ADPCM_IMA_WS;
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st->codecpar->bits_per_coded_sample = 4;
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st->codecpar->bit_rate = channels * sample_rate * 4;
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break;
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default:
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avpriv_request_sample(s, "Unknown codec: %d", 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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codecpar->channels = channels;
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st->codecpar->channel_layout = channels == 1 ? AV_CH_LAYOUT_MONO :
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AV_CH_LAYOUT_STEREO;
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st->codecpar->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->codecpar->codec_id == AV_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->codecpar->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"),
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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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