Mercurial > libavformat.hg
view westwood.c @ 5768:b9c18d4872a2 libavformat
Move the probe loop from av_open_input_file() into its own method
av_probe_input_buffer() so that it can be reused. Here are a few
differences to the original way things were probed:
- maximum probe buffer size can be specified as a parameter.
- offset within the stream to probe from can be specified as a parameter.
- instead of seeking back to the start each time a probe fails, stream
data is appended to the reallocated buffer. This lowers the amount
of data read from the stream (there is no repetition) and results in
fewer closed and reopened streams (when seeking fails).
Patch by Micah F. Galizia printf("%s%s@%s.%s", "micah", "galizia", "gmail", "com").
author | stefano |
---|---|
date | Sun, 07 Mar 2010 22:42:11 +0000 |
parents | 1c4ca5c32f0f |
children | 536e5527c1e0 |
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/* * Westwood Studios Multimedia Formats Demuxer (VQA, AUD) * Copyright (c) 2003 The ffmpeg Project * * 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 libavformat/westwood.c * Westwood Studios VQA & AUD file demuxers * by Mike Melanson (melanson@pcisys.net) * for more information on the Westwood file formats, visit: * http://www.pcisys.net/~melanson/codecs/ * http://www.geocities.com/SiliconValley/8682/aud3.txt * * Implementation note: There is no definite file signature for AUD files. * The demuxer uses a probabilistic strategy for content detection. This * entails performing sanity checks on certain header values in order to * qualify a file. Refer to wsaud_probe() for the precise parameters. */ #include "libavutil/intreadwrite.h" #include "avformat.h" #define AUD_HEADER_SIZE 12 #define AUD_CHUNK_PREAMBLE_SIZE 8 #define AUD_CHUNK_SIGNATURE 0x0000DEAF #define FORM_TAG MKBETAG('F', 'O', 'R', 'M') #define WVQA_TAG MKBETAG('W', 'V', 'Q', 'A') #define VQHD_TAG MKBETAG('V', 'Q', 'H', 'D') #define FINF_TAG MKBETAG('F', 'I', 'N', 'F') #define SND0_TAG MKBETAG('S', 'N', 'D', '0') #define SND1_TAG MKBETAG('S', 'N', 'D', '1') #define SND2_TAG MKBETAG('S', 'N', 'D', '2') #define VQFR_TAG MKBETAG('V', 'Q', 'F', 'R') /* don't know what these tags are for, but acknowledge their existence */ #define CINF_TAG MKBETAG('C', 'I', 'N', 'F') #define CINH_TAG MKBETAG('C', 'I', 'N', 'H') #define CIND_TAG MKBETAG('C', 'I', 'N', 'D') #define PINF_TAG MKBETAG('P', 'I', 'N', 'F') #define PINH_TAG MKBETAG('P', 'I', 'N', 'H') #define PIND_TAG MKBETAG('P', 'I', 'N', 'D') #define CMDS_TAG MKBETAG('C', 'M', 'D', 'S') #define VQA_HEADER_SIZE 0x2A #define VQA_FRAMERATE 15 #define VQA_PREAMBLE_SIZE 8 typedef struct WsAudDemuxContext { int audio_samplerate; int audio_channels; int audio_bits; enum CodecID audio_type; int audio_stream_index; int64_t audio_frame_counter; } WsAudDemuxContext; typedef struct WsVqaDemuxContext { int audio_samplerate; int audio_channels; int audio_bits; int audio_stream_index; int video_stream_index; int64_t audio_frame_counter; } WsVqaDemuxContext; static int wsaud_probe(AVProbeData *p) { int field; /* Probabilistic content detection strategy: There is no file signature * so perform sanity checks on various header parameters: * 8000 <= sample rate (16 bits) <= 48000 ==> 40001 acceptable numbers * flags <= 0x03 (2 LSBs are used) ==> 4 acceptable numbers * compression type (8 bits) = 1 or 99 ==> 2 acceptable numbers * first audio chunk signature (32 bits) ==> 1 acceptable number * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 = * 320008 acceptable number combinations. */ if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE) return 0; /* check sample rate */ field = AV_RL16(&p->buf[0]); if ((field < 8000) || (field > 48000)) return 0; /* enforce the rule that the top 6 bits of this flags field are reserved (0); * this might not be true, but enforce it until deemed unnecessary */ if (p->buf[10] & 0xFC) return 0; /* note: only check for WS IMA (type 99) right now since there is no * support for type 1 */ if (p->buf[11] != 99) return 0; /* read ahead to the first audio chunk and validate the first header signature */ if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE) return 0; /* return 1/2 certainty since this file check is a little sketchy */ return AVPROBE_SCORE_MAX / 2; } static int wsaud_read_header(AVFormatContext *s, AVFormatParameters *ap) { WsAudDemuxContext *wsaud = s->priv_data; ByteIOContext *pb = s->pb; AVStream *st; unsigned char header[AUD_HEADER_SIZE]; if (get_buffer(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE) return AVERROR(EIO); wsaud->audio_samplerate = AV_RL16(&header[0]); if (header[11] == 99) wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS; else return AVERROR_INVALIDDATA; /* flag 0 indicates stereo */ wsaud->audio_channels = (header[10] & 0x1) + 1; /* flag 1 indicates 16 bit audio */ wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8; /* initialize the audio decoder stream */ st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); av_set_pts_info(st, 33, 1, wsaud->audio_samplerate); st->codec->codec_type = CODEC_TYPE_AUDIO; st->codec->codec_id = wsaud->audio_type; st->codec->codec_tag = 0; /* no tag */ st->codec->channels = wsaud->audio_channels; st->codec->sample_rate = wsaud->audio_samplerate; st->codec->bits_per_coded_sample = wsaud->audio_bits; st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * st->codec->bits_per_coded_sample / 4; st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample; wsaud->audio_stream_index = st->index; wsaud->audio_frame_counter = 0; return 0; } static int wsaud_read_packet(AVFormatContext *s, AVPacket *pkt) { WsAudDemuxContext *wsaud = s->priv_data; ByteIOContext *pb = s->pb; unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE]; unsigned int chunk_size; int ret = 0; if (get_buffer(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) != AUD_CHUNK_PREAMBLE_SIZE) return AVERROR(EIO); /* validate the chunk */ if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE) return AVERROR_INVALIDDATA; chunk_size = AV_RL16(&preamble[0]); ret= av_get_packet(pb, pkt, chunk_size); if (ret != chunk_size) return AVERROR(EIO); pkt->stream_index = wsaud->audio_stream_index; pkt->pts = wsaud->audio_frame_counter; pkt->pts /= wsaud->audio_samplerate; /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */ wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels; return ret; } static int wsvqa_probe(AVProbeData *p) { /* need 12 bytes to qualify */ if (p->buf_size < 12) return 0; /* check for the VQA signatures */ if ((AV_RB32(&p->buf[0]) != FORM_TAG) || (AV_RB32(&p->buf[8]) != WVQA_TAG)) return 0; return AVPROBE_SCORE_MAX; } static int wsvqa_read_header(AVFormatContext *s, AVFormatParameters *ap) { WsVqaDemuxContext *wsvqa = s->priv_data; ByteIOContext *pb = s->pb; AVStream *st; unsigned char *header; unsigned char scratch[VQA_PREAMBLE_SIZE]; unsigned int chunk_tag; unsigned int chunk_size; /* initialize the video decoder stream */ st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); av_set_pts_info(st, 33, 1, VQA_FRAMERATE); wsvqa->video_stream_index = st->index; st->codec->codec_type = CODEC_TYPE_VIDEO; st->codec->codec_id = CODEC_ID_WS_VQA; st->codec->codec_tag = 0; /* no fourcc */ /* skip to the start of the VQA header */ url_fseek(pb, 20, SEEK_SET); /* the VQA header needs to go to the decoder */ st->codec->extradata_size = VQA_HEADER_SIZE; st->codec->extradata = av_mallocz(VQA_HEADER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE); header = (unsigned char *)st->codec->extradata; if (get_buffer(pb, st->codec->extradata, VQA_HEADER_SIZE) != VQA_HEADER_SIZE) { av_free(st->codec->extradata); return AVERROR(EIO); } st->codec->width = AV_RL16(&header[6]); st->codec->height = AV_RL16(&header[8]); /* initialize the audio decoder stream for VQA v1 or nonzero samplerate */ if (AV_RL16(&header[24]) || (AV_RL16(&header[0]) == 1 && AV_RL16(&header[2]) == 1)) { st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); av_set_pts_info(st, 33, 1, VQA_FRAMERATE); st->codec->codec_type = CODEC_TYPE_AUDIO; if (AV_RL16(&header[0]) == 1) st->codec->codec_id = CODEC_ID_WESTWOOD_SND1; else st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS; st->codec->codec_tag = 0; /* no tag */ st->codec->sample_rate = AV_RL16(&header[24]); if (!st->codec->sample_rate) st->codec->sample_rate = 22050; st->codec->channels = header[26]; if (!st->codec->channels) st->codec->channels = 1; st->codec->bits_per_coded_sample = 16; st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * st->codec->bits_per_coded_sample / 4; st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample; wsvqa->audio_stream_index = st->index; wsvqa->audio_samplerate = st->codec->sample_rate; wsvqa->audio_channels = st->codec->channels; wsvqa->audio_frame_counter = 0; } /* there are 0 or more chunks before the FINF chunk; iterate until * FINF has been skipped and the file will be ready to be demuxed */ do { if (get_buffer(pb, scratch, VQA_PREAMBLE_SIZE) != VQA_PREAMBLE_SIZE) { av_free(st->codec->extradata); return AVERROR(EIO); } chunk_tag = AV_RB32(&scratch[0]); chunk_size = AV_RB32(&scratch[4]); /* catch any unknown header tags, for curiousity */ switch (chunk_tag) { case CINF_TAG: case CINH_TAG: case CIND_TAG: case PINF_TAG: case PINH_TAG: case PIND_TAG: case FINF_TAG: case CMDS_TAG: break; default: av_log (s, AV_LOG_ERROR, " note: unknown chunk seen (%c%c%c%c)\n", scratch[0], scratch[1], scratch[2], scratch[3]); break; } url_fseek(pb, chunk_size, SEEK_CUR); } while (chunk_tag != FINF_TAG); return 0; } static int wsvqa_read_packet(AVFormatContext *s, AVPacket *pkt) { WsVqaDemuxContext *wsvqa = s->priv_data; ByteIOContext *pb = s->pb; int ret = -1; unsigned char preamble[VQA_PREAMBLE_SIZE]; unsigned int chunk_type; unsigned int chunk_size; int skip_byte; while (get_buffer(pb, preamble, VQA_PREAMBLE_SIZE) == VQA_PREAMBLE_SIZE) { chunk_type = AV_RB32(&preamble[0]); chunk_size = AV_RB32(&preamble[4]); skip_byte = chunk_size & 0x01; if ((chunk_type == SND1_TAG) || (chunk_type == SND2_TAG) || (chunk_type == VQFR_TAG)) { if (av_new_packet(pkt, chunk_size)) return AVERROR(EIO); ret = get_buffer(pb, pkt->data, chunk_size); if (ret != chunk_size) { av_free_packet(pkt); return AVERROR(EIO); } if (chunk_type == SND2_TAG) { pkt->stream_index = wsvqa->audio_stream_index; /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */ wsvqa->audio_frame_counter += (chunk_size * 2) / wsvqa->audio_channels; } else if(chunk_type == SND1_TAG) { pkt->stream_index = wsvqa->audio_stream_index; /* unpacked size is stored in header */ wsvqa->audio_frame_counter += AV_RL16(pkt->data) / wsvqa->audio_channels; } else { pkt->stream_index = wsvqa->video_stream_index; } /* stay on 16-bit alignment */ if (skip_byte) url_fseek(pb, 1, SEEK_CUR); return ret; } else { switch(chunk_type){ case CMDS_TAG: case SND0_TAG: break; default: av_log(s, AV_LOG_INFO, "Skipping unknown chunk 0x%08X\n", chunk_type); } url_fseek(pb, chunk_size + skip_byte, SEEK_CUR); } } return ret; } #if CONFIG_WSAUD_DEMUXER AVInputFormat wsaud_demuxer = { "wsaud", NULL_IF_CONFIG_SMALL("Westwood Studios audio format"), sizeof(WsAudDemuxContext), wsaud_probe, wsaud_read_header, wsaud_read_packet, }; #endif #if CONFIG_WSVQA_DEMUXER AVInputFormat wsvqa_demuxer = { "wsvqa", NULL_IF_CONFIG_SMALL("Westwood Studios VQA format"), sizeof(WsVqaDemuxContext), wsvqa_probe, wsvqa_read_header, wsvqa_read_packet, }; #endif