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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavformat/spdifdec.c
Michael Niedermayer e37f161e66 Merge remote-tracking branch 'qatar/master'
* qatar/master: (71 commits)
  movenc: Allow writing to a non-seekable output if using empty moov
  movenc: Support adding isml (smooth streaming live) metadata
  libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set
  sunrast: Document the different Sun Raster file format types.
  sunrast: Add a check for experimental type.
  libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat
  lavf: remove disabled FF_API_SET_PTS_INFO cruft
  lavf: remove disabled FF_API_OLD_INTERRUPT_CB cruft
  lavf: remove disabled FF_API_REORDER_PRIVATE cruft
  lavf: remove disabled FF_API_SEEK_PUBLIC cruft
  lavf: remove disabled FF_API_STREAM_COPY cruft
  lavf: remove disabled FF_API_PRELOAD cruft
  lavf: remove disabled FF_API_NEW_STREAM cruft
  lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft
  lavf: remove disabled FF_API_MUXRATE cruft
  lavf: remove disabled FF_API_FILESIZE cruft
  lavf: remove disabled FF_API_TIMESTAMP cruft
  lavf: remove disabled FF_API_LOOP_OUTPUT cruft
  lavf: remove disabled FF_API_LOOP_INPUT cruft
  lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft
  ...

Conflicts:
	doc/APIchanges
	libavcodec/8bps.c
	libavcodec/avcodec.h
	libavcodec/libx264.c
	libavcodec/mjpegbdec.c
	libavcodec/options.c
	libavcodec/sunrast.c
	libavcodec/utils.c
	libavcodec/version.h
	libavcodec/x86/h264_deblock.asm
	libavdevice/libdc1394.c
	libavdevice/v4l2.c
	libavformat/avformat.h
	libavformat/avio.c
	libavformat/avio.h
	libavformat/aviobuf.c
	libavformat/dv.c
	libavformat/mov.c
	libavformat/utils.c
	libavformat/version.h
	libavformat/wtv.c
	libavutil/Makefile
	libavutil/file.c
	libswscale/x86/input.asm
	libswscale/x86/swscale_mmx.c
	libswscale/x86/swscale_template.c
	tests/ref/lavf/ffm

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-01-28 07:53:34 +01:00

236 lines
7.2 KiB
C

/*
* IEC 61937 demuxer
* Copyright (c) 2010 Anssi Hannula <anssi.hannula at iki.fi>
*
* 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
* IEC 61937 demuxer, used for compressed data in S/PDIF
* @author Anssi Hannula
*/
#include "avformat.h"
#include "spdif.h"
#include "libavcodec/ac3.h"
#include "libavcodec/aacadtsdec.h"
static int spdif_get_offset_and_codec(AVFormatContext *s,
enum IEC61937DataType data_type,
const char *buf, int *offset,
enum CodecID *codec)
{
AACADTSHeaderInfo aac_hdr;
GetBitContext gbc;
switch (data_type & 0xff) {
case IEC61937_AC3:
*offset = AC3_FRAME_SIZE << 2;
*codec = CODEC_ID_AC3;
break;
case IEC61937_MPEG1_LAYER1:
*offset = spdif_mpeg_pkt_offset[1][0];
*codec = CODEC_ID_MP1;
break;
case IEC61937_MPEG1_LAYER23:
*offset = spdif_mpeg_pkt_offset[1][0];
*codec = CODEC_ID_MP3;
break;
case IEC61937_MPEG2_EXT:
*offset = 4608;
*codec = CODEC_ID_MP3;
break;
case IEC61937_MPEG2_AAC:
init_get_bits(&gbc, buf, AAC_ADTS_HEADER_SIZE * 8);
if (avpriv_aac_parse_header(&gbc, &aac_hdr)) {
if (s) /* be silent during a probe */
av_log(s, AV_LOG_ERROR, "Invalid AAC packet in IEC 61937\n");
return AVERROR_INVALIDDATA;
}
*offset = aac_hdr.samples << 2;
*codec = CODEC_ID_AAC;
break;
case IEC61937_MPEG2_LAYER1_LSF:
*offset = spdif_mpeg_pkt_offset[0][0];
*codec = CODEC_ID_MP1;
break;
case IEC61937_MPEG2_LAYER2_LSF:
*offset = spdif_mpeg_pkt_offset[0][1];
*codec = CODEC_ID_MP2;
break;
case IEC61937_MPEG2_LAYER3_LSF:
*offset = spdif_mpeg_pkt_offset[0][2];
*codec = CODEC_ID_MP3;
break;
case IEC61937_DTS1:
*offset = 2048;
*codec = CODEC_ID_DTS;
break;
case IEC61937_DTS2:
*offset = 4096;
*codec = CODEC_ID_DTS;
break;
case IEC61937_DTS3:
*offset = 8192;
*codec = CODEC_ID_DTS;
break;
default:
if (s) { /* be silent during a probe */
av_log(s, AV_LOG_WARNING, "Data type 0x%04x", data_type);
av_log_missing_feature(s, " in IEC 61937 is", 1);
}
return AVERROR_PATCHWELCOME;
}
return 0;
}
/* Largest offset between bursts we currently handle, i.e. AAC with
aac_hdr.samples = 4096 */
#define SPDIF_MAX_OFFSET 16384
static int spdif_probe(AVProbeData *p)
{
const uint8_t *buf = p->buf;
const uint8_t *probe_end = p->buf + FFMIN(2 * SPDIF_MAX_OFFSET, p->buf_size - 1);
const uint8_t *expected_code = buf + 7;
uint32_t state = 0;
int sync_codes = 0;
int consecutive_codes = 0;
int offset;
enum CodecID codec;
for (; buf < probe_end; buf++) {
state = (state << 8) | *buf;
if (state == (AV_BSWAP16C(SYNCWORD1) << 16 | AV_BSWAP16C(SYNCWORD2))
&& buf[1] < 0x37) {
sync_codes++;
if (buf == expected_code) {
if (++consecutive_codes >= 2)
return AVPROBE_SCORE_MAX;
} else
consecutive_codes = 0;
if (buf + 4 + AAC_ADTS_HEADER_SIZE > p->buf + p->buf_size)
break;
/* continue probing to find more sync codes */
probe_end = FFMIN(buf + SPDIF_MAX_OFFSET, p->buf + p->buf_size - 1);
/* skip directly to the next sync code */
if (!spdif_get_offset_and_codec(NULL, (buf[2] << 8) | buf[1],
&buf[5], &offset, &codec)) {
if (buf + offset >= p->buf + p->buf_size)
break;
expected_code = buf + offset;
buf = expected_code - 7;
}
}
}
if (!sync_codes)
return 0;
if (sync_codes >= 6)
/* good amount of sync codes but with unexpected offsets */
return AVPROBE_SCORE_MAX / 2;
/* some sync codes were found */
return AVPROBE_SCORE_MAX / 8;
}
static int spdif_read_header(AVFormatContext *s)
{
s->ctx_flags |= AVFMTCTX_NOHEADER;
return 0;
}
static int spdif_read_packet(AVFormatContext *s, AVPacket *pkt)
{
AVIOContext *pb = s->pb;
enum IEC61937DataType data_type;
enum CodecID codec_id;
uint32_t state = 0;
int pkt_size_bits, offset, ret;
while (state != (AV_BSWAP16C(SYNCWORD1) << 16 | AV_BSWAP16C(SYNCWORD2))) {
state = (state << 8) | avio_r8(pb);
if (url_feof(pb))
return AVERROR_EOF;
}
data_type = avio_rl16(pb);
pkt_size_bits = avio_rl16(pb);
if (pkt_size_bits % 16)
av_log_ask_for_sample(s, "Packet does not end to a 16-bit boundary.");
ret = av_new_packet(pkt, FFALIGN(pkt_size_bits, 16) >> 3);
if (ret)
return ret;
pkt->pos = avio_tell(pb) - BURST_HEADER_SIZE;
if (avio_read(pb, pkt->data, pkt->size) < pkt->size) {
av_free_packet(pkt);
return AVERROR_EOF;
}
ff_spdif_bswap_buf16((uint16_t *)pkt->data, (uint16_t *)pkt->data, pkt->size >> 1);
ret = spdif_get_offset_and_codec(s, data_type, pkt->data,
&offset, &codec_id);
if (ret) {
av_free_packet(pkt);
return ret;
}
/* skip over the padding to the beginning of the next frame */
avio_skip(pb, offset - pkt->size - BURST_HEADER_SIZE);
if (!s->nb_streams) {
/* first packet, create a stream */
AVStream *st = avformat_new_stream(s, NULL);
if (!st) {
av_free_packet(pkt);
return AVERROR(ENOMEM);
}
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = codec_id;
} else if (codec_id != s->streams[0]->codec->codec_id) {
av_log_missing_feature(s, "codec change in IEC 61937", 0);
return AVERROR_PATCHWELCOME;
}
if (!s->bit_rate && s->streams[0]->codec->sample_rate)
/* stream bitrate matches 16-bit stereo PCM bitrate for currently
supported codecs */
s->bit_rate = 2 * 16 * s->streams[0]->codec->sample_rate;
return 0;
}
AVInputFormat ff_spdif_demuxer = {
.name = "spdif",
.long_name = NULL_IF_CONFIG_SMALL("IEC 61937 (compressed data in S/PDIF)"),
.read_probe = spdif_probe,
.read_header = spdif_read_header,
.read_packet = spdif_read_packet,
.flags = AVFMT_GENERIC_INDEX,
};