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![Michael Niedermayer](/assets/img/avatar_default.png)
* qatar/master: tiertexseq: set correct block_align for audio tiertexseq: set audio stream start time to 0 voc/avs: Do not change the sample rate mid-stream. segafilm: use the sample rate as the time base for audio streams ea: fix audio pts psx-str: fix audio pts vqf: set packet duration tta demuxer: set packet duration mpegaudio_parser: do not ignore information from the first parsed frame mpegaudio_parser: be less picky about the start position thp: set audio packet durations avcodec: add a Vorbis parser to get packet duration vorbisdec: read the previous window flag for long windows lavc: free the output packet when encoding failed or produced no output. lavc: preserve avpkt->destruct in ff_alloc_packet(). lavc: clarify the meaning of AVCodecContext.frame_number. mpegts: Pad the packet buffer in handle_packet(). mpegts: Do not call read_sl_header() when no bytes remain in the buffer. Conflicts: libavcodec/mpegaudio_parser.c libavcodec/version.h libavformat/mpegts.c tests/ref/fate/pva-demux Merged-by: Michael Niedermayer <michaelni@gmx.at>
171 lines
5.1 KiB
C
171 lines
5.1 KiB
C
/*
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* TTA demuxer
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* Copyright (c) 2006 Alex Beregszaszi
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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#include "libavcodec/get_bits.h"
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#include "avformat.h"
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#include "internal.h"
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#include "id3v1.h"
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#include "libavutil/dict.h"
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typedef struct {
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int totalframes, currentframe;
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int frame_size;
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int last_frame_size;
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} TTAContext;
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static int tta_probe(AVProbeData *p)
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{
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const uint8_t *d = p->buf;
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if (d[0] == 'T' && d[1] == 'T' && d[2] == 'A' && d[3] == '1')
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return 80;
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return 0;
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}
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static int tta_read_header(AVFormatContext *s)
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{
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TTAContext *c = s->priv_data;
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AVStream *st;
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int i, channels, bps, samplerate, datalen;
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uint64_t framepos, start_offset;
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if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX))
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ff_id3v1_read(s);
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start_offset = avio_tell(s->pb);
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if (avio_rl32(s->pb) != AV_RL32("TTA1"))
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return -1; // not tta file
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avio_skip(s->pb, 2); // FIXME: flags
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channels = avio_rl16(s->pb);
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bps = avio_rl16(s->pb);
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samplerate = avio_rl32(s->pb);
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if(samplerate <= 0 || samplerate > 1000000){
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av_log(s, AV_LOG_ERROR, "nonsense samplerate\n");
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return -1;
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}
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datalen = avio_rl32(s->pb);
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if(datalen < 0){
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av_log(s, AV_LOG_ERROR, "nonsense datalen\n");
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return -1;
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}
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avio_skip(s->pb, 4); // header crc
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c->frame_size = samplerate * 256 / 245;
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c->last_frame_size = datalen % c->frame_size;
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if (!c->last_frame_size)
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c->last_frame_size = c->frame_size;
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c->totalframes = datalen / c->frame_size + (c->last_frame_size < c->frame_size);
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c->currentframe = 0;
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if(c->totalframes >= UINT_MAX/sizeof(uint32_t) || c->totalframes <= 0){
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av_log(s, AV_LOG_ERROR, "totalframes %d invalid\n", c->totalframes);
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return -1;
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}
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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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avpriv_set_pts_info(st, 64, 1, samplerate);
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st->start_time = 0;
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st->duration = datalen;
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framepos = avio_tell(s->pb) + 4*c->totalframes + 4;
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for (i = 0; i < c->totalframes; i++) {
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uint32_t size = avio_rl32(s->pb);
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av_add_index_entry(st, framepos, i * c->frame_size, size, 0,
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AVINDEX_KEYFRAME);
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framepos += size;
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}
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avio_skip(s->pb, 4); // seektable crc
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = CODEC_ID_TTA;
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st->codec->channels = channels;
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st->codec->sample_rate = samplerate;
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st->codec->bits_per_coded_sample = bps;
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st->codec->extradata_size = avio_tell(s->pb) - start_offset;
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if(st->codec->extradata_size+FF_INPUT_BUFFER_PADDING_SIZE <= (unsigned)st->codec->extradata_size){
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//this check is redundant as avio_read should fail
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av_log(s, AV_LOG_ERROR, "extradata_size too large\n");
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return -1;
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}
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st->codec->extradata = av_mallocz(st->codec->extradata_size+FF_INPUT_BUFFER_PADDING_SIZE);
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if (!st->codec->extradata) {
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st->codec->extradata_size = 0;
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return AVERROR(ENOMEM);
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}
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avio_seek(s->pb, start_offset, SEEK_SET);
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avio_read(s->pb, st->codec->extradata, st->codec->extradata_size);
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return 0;
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}
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static int tta_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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TTAContext *c = s->priv_data;
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AVStream *st = s->streams[0];
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int size, ret;
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// FIXME!
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if (c->currentframe >= c->totalframes)
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return AVERROR_EOF;
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size = st->index_entries[c->currentframe].size;
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ret = av_get_packet(s->pb, pkt, size);
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pkt->dts = st->index_entries[c->currentframe++].timestamp;
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pkt->duration = c->currentframe == c->totalframes ? c->last_frame_size :
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c->frame_size;
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return ret;
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}
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static int tta_read_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags)
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{
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TTAContext *c = s->priv_data;
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AVStream *st = s->streams[stream_index];
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int index = av_index_search_timestamp(st, timestamp, flags);
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if (index < 0)
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return -1;
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if (avio_seek(s->pb, st->index_entries[index].pos, SEEK_SET) < 0)
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return -1;
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c->currentframe = index;
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return 0;
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}
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AVInputFormat ff_tta_demuxer = {
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.name = "tta",
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.long_name = NULL_IF_CONFIG_SMALL("True Audio"),
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.priv_data_size = sizeof(TTAContext),
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.read_probe = tta_probe,
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.read_header = tta_read_header,
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.read_packet = tta_read_packet,
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.read_seek = tta_read_seek,
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.extensions = "tta",
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};
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