librempeg/libavformat/westwood.c
Michael Niedermayer 9d76cf0b18 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtpdec: Templatize the code for different g726 bitrate variants
  rv40: move loop filter to rv34dsp context
  lavf: make av_set_pts_info private.
  rtpdec: Add support for G726 audio
  rtpdec: Add an init function that can do custom codec context initialization
  avconv: make copy_tb on by default.
  matroskadec: don't set codec timebase.
  rmdec: don't set codec timebase.
  avconv: compute next_pts from input packet duration when possible.
  lavf: estimate frame duration from r_frame_rate.
  avconv: update InputStream.pts in the streamcopy case.

Conflicts:
	avconv.c
	libavdevice/alsa-audio-dec.c
	libavdevice/bktr.c
	libavdevice/fbdev.c
	libavdevice/libdc1394.c
	libavdevice/oss_audio.c
	libavdevice/v4l.c
	libavdevice/v4l2.c
	libavdevice/vfwcap.c
	libavdevice/x11grab.c
	libavformat/au.c
	libavformat/eacdata.c
	libavformat/flvdec.c
	libavformat/mpegts.c
	libavformat/mxfenc.c
	libavformat/rtpdec_g726.c
	libavformat/wtv.c
	libavformat/xmv.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-01 02:54:24 +01:00

388 lines
13 KiB
C

/*
* 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
* 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"
#include "internal.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;
AVIOContext *pb = s->pb;
AVStream *st;
unsigned char header[AUD_HEADER_SIZE];
if (avio_read(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 = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avpriv_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
st->codec->codec_type = AVMEDIA_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;
AVIOContext *pb = s->pb;
unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
unsigned int chunk_size;
int ret = 0;
if (avio_read(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;
AVIOContext *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 = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
wsvqa->video_stream_index = st->index;
st->codec->codec_type = AVMEDIA_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 */
avio_seek(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 (avio_read(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 = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avpriv_set_pts_info(st, 33, 1, VQA_FRAMERATE);
st->codec->codec_type = AVMEDIA_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 (avio_read(pb, scratch, VQA_PREAMBLE_SIZE) != VQA_PREAMBLE_SIZE)
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;
}
avio_skip(pb, chunk_size);
} while (chunk_tag != FINF_TAG);
return 0;
}
static int wsvqa_read_packet(AVFormatContext *s,
AVPacket *pkt)
{
WsVqaDemuxContext *wsvqa = s->priv_data;
AVIOContext *pb = s->pb;
int ret = -1;
unsigned char preamble[VQA_PREAMBLE_SIZE];
unsigned int chunk_type;
unsigned int chunk_size;
int skip_byte;
while (avio_read(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 = avio_read(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)
avio_skip(pb, 1);
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);
}
avio_skip(pb, chunk_size + skip_byte);
}
}
return ret;
}
#if CONFIG_WSAUD_DEMUXER
AVInputFormat ff_wsaud_demuxer = {
.name = "wsaud",
.long_name = NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
.priv_data_size = sizeof(WsAudDemuxContext),
.read_probe = wsaud_probe,
.read_header = wsaud_read_header,
.read_packet = wsaud_read_packet,
};
#endif
#if CONFIG_WSVQA_DEMUXER
AVInputFormat ff_wsvqa_demuxer = {
.name = "wsvqa",
.long_name = NULL_IF_CONFIG_SMALL("Westwood Studios VQA format"),
.priv_data_size = sizeof(WsVqaDemuxContext),
.read_probe = wsvqa_probe,
.read_header = wsvqa_read_header,
.read_packet = wsvqa_read_packet,
};
#endif