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FFmpeg/libavformat/genh.c
Michael Niedermayer dcb6d5b831 avformat/genh: Mark coef_splitted as av_unused
This avoid "libavformat/genh.c:43:14: warning: variable coef_splitted set but not used"
Fewer warnings makes it easier to see new and important warnings

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2016-02-14 21:05:50 +01:00

195 lines
6.7 KiB
C

/*
* GENH demuxer
* Copyright (c) 2015 Paul B Mahol
*
* 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
*/
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
typedef struct GENHDemuxContext {
unsigned dsp_int_type;
unsigned interleave_size;
} GENHDemuxContext;
static int genh_probe(AVProbeData *p)
{
if (AV_RL32(p->buf) != MKTAG('G','E','N','H'))
return 0;
if (AV_RL32(p->buf+4) <= 0 || AV_RL32(p->buf+4) > 0xFFFF) // channels
return 0;
return AVPROBE_SCORE_MAX / 3 * 2;
}
static int genh_read_header(AVFormatContext *s)
{
unsigned start_offset, header_size, codec, coef_type, coef[2];
GENHDemuxContext *c = s->priv_data;
av_unused unsigned coef_splitted[2];
int align, ch, ret;
AVStream *st;
avio_skip(s->pb, 4);
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->channels = avio_rl32(s->pb);
if (st->codec->channels <= 0)
return AVERROR_INVALIDDATA;
if (st->codec->channels == 1)
st->codec->channel_layout = AV_CH_LAYOUT_MONO;
else if (st->codec->channels == 2)
st->codec->channel_layout = AV_CH_LAYOUT_STEREO;
align =
c->interleave_size = avio_rl32(s->pb);
if (align < 0 || align > INT_MAX / st->codec->channels)
return AVERROR_INVALIDDATA;
st->codec->block_align = align * st->codec->channels;
st->codec->sample_rate = avio_rl32(s->pb);
avio_skip(s->pb, 4);
st->duration = avio_rl32(s->pb);
codec = avio_rl32(s->pb);
switch (codec) {
case 0: st->codec->codec_id = AV_CODEC_ID_ADPCM_PSX; break;
case 1:
case 11: st->codec->bits_per_coded_sample = 4;
st->codec->block_align = 36 * st->codec->channels;
st->codec->codec_id = AV_CODEC_ID_ADPCM_IMA_WAV; break;
case 2: st->codec->codec_id = AV_CODEC_ID_ADPCM_DTK; break;
case 3: st->codec->codec_id = st->codec->block_align > 0 ?
AV_CODEC_ID_PCM_S16BE_PLANAR :
AV_CODEC_ID_PCM_S16BE; break;
case 4: st->codec->codec_id = st->codec->block_align > 0 ?
AV_CODEC_ID_PCM_S16LE_PLANAR :
AV_CODEC_ID_PCM_S16LE; break;
case 5: st->codec->codec_id = st->codec->block_align > 0 ?
AV_CODEC_ID_PCM_S8_PLANAR :
AV_CODEC_ID_PCM_S8; break;
case 6: st->codec->codec_id = AV_CODEC_ID_SDX2_DPCM; break;
case 7: ret = ff_alloc_extradata(st->codec, 2);
if (ret < 0)
return ret;
AV_WL16(st->codec->extradata, 3);
st->codec->codec_id = AV_CODEC_ID_ADPCM_IMA_WS; break;
case 10: st->codec->codec_id = AV_CODEC_ID_ADPCM_AICA; break;
case 12: st->codec->codec_id = AV_CODEC_ID_ADPCM_THP; break;
case 13: st->codec->codec_id = AV_CODEC_ID_PCM_U8; break;
case 17: st->codec->codec_id = AV_CODEC_ID_ADPCM_IMA_QT; break;
default:
avpriv_request_sample(s, "codec %d", codec);
return AVERROR_PATCHWELCOME;
}
start_offset = avio_rl32(s->pb);
header_size = avio_rl32(s->pb);
if (header_size > start_offset)
return AVERROR_INVALIDDATA;
if (header_size == 0)
start_offset = 0x800;
coef[0] = avio_rl32(s->pb);
coef[1] = avio_rl32(s->pb);
c->dsp_int_type = avio_rl32(s->pb);
coef_type = avio_rl32(s->pb);
coef_splitted[0] = avio_rl32(s->pb);
coef_splitted[1] = avio_rl32(s->pb);
if (st->codec->codec_id == AV_CODEC_ID_ADPCM_THP) {
if (st->codec->channels > 2) {
avpriv_request_sample(s, "channels %d>2", st->codec->channels);
return AVERROR_PATCHWELCOME;
}
ff_alloc_extradata(st->codec, 32 * st->codec->channels);
for (ch = 0; ch < st->codec->channels; ch++) {
if (coef_type & 1) {
avpriv_request_sample(s, "coef_type & 1");
return AVERROR_PATCHWELCOME;
} else {
avio_seek(s->pb, coef[ch], SEEK_SET);
avio_read(s->pb, st->codec->extradata + 32 * ch, 32);
}
}
if (c->dsp_int_type == 1) {
st->codec->block_align = 8 * st->codec->channels;
if (c->interleave_size != 1 &&
c->interleave_size != 2 &&
c->interleave_size != 4)
return AVERROR_INVALIDDATA;
}
}
avio_skip(s->pb, start_offset - avio_tell(s->pb));
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
return 0;
}
static int genh_read_packet(AVFormatContext *s, AVPacket *pkt)
{
AVCodecContext *codec = s->streams[0]->codec;
GENHDemuxContext *c = s->priv_data;
int ret;
if (c->dsp_int_type == 1 && codec->codec_id == AV_CODEC_ID_ADPCM_THP &&
codec->channels > 1) {
int i, ch;
if (avio_feof(s->pb))
return AVERROR_EOF;
ret = av_new_packet(pkt, 8 * codec->channels);
if (ret < 0)
return ret;
for (i = 0; i < 8 / c->interleave_size; i++) {
for (ch = 0; ch < codec->channels; ch++) {
pkt->data[ch * 8 + i*c->interleave_size+0] = avio_r8(s->pb);
pkt->data[ch * 8 + i*c->interleave_size+1] = avio_r8(s->pb);
}
}
ret = 0;
} else if (codec->codec_id == AV_CODEC_ID_SDX2_DPCM) {
ret = av_get_packet(s->pb, pkt, codec->block_align * 1024);
} else {
ret = av_get_packet(s->pb, pkt, codec->block_align ? codec->block_align : 1024 * codec->channels);
}
pkt->stream_index = 0;
return ret;
}
AVInputFormat ff_genh_demuxer = {
.name = "genh",
.long_name = NULL_IF_CONFIG_SMALL("GENeric Header"),
.priv_data_size = sizeof(GENHDemuxContext),
.read_probe = genh_probe,
.read_header = genh_read_header,
.read_packet = genh_read_packet,
.extensions = "genh",
};