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6148 serge 1
/*
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 * PMP demuxer.
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 * Copyright (c) 2011 Reimar Döffinger
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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 "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "internal.h"
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typedef struct {
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    int cur_stream;
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    int num_streams;
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    int audio_packets;
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    int current_packet;
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    uint32_t *packet_sizes;
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    int packet_sizes_alloc;
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} PMPContext;
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static int pmp_probe(AVProbeData *p) {
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    if (AV_RN32(p->buf) == AV_RN32("pmpm") &&
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        AV_RL32(p->buf + 4) == 1)
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        return AVPROBE_SCORE_MAX;
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    return 0;
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}
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static int pmp_header(AVFormatContext *s)
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{
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    PMPContext *pmp = s->priv_data;
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    AVIOContext *pb = s->pb;
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    int tb_num, tb_den;
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    uint32_t index_cnt;
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    int audio_codec_id = AV_CODEC_ID_NONE;
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    int srate, channels;
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    unsigned i;
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    uint64_t pos;
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    int64_t fsize = avio_size(pb);
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    AVStream *vst = avformat_new_stream(s, NULL);
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    if (!vst)
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        return AVERROR(ENOMEM);
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    vst->codec->codec_type = AVMEDIA_TYPE_VIDEO;
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    avio_skip(pb, 8);
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    switch (avio_rl32(pb)) {
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    case 0:
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        vst->codec->codec_id = AV_CODEC_ID_MPEG4;
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        break;
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    case 1:
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        vst->codec->codec_id = AV_CODEC_ID_H264;
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        break;
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    default:
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        av_log(s, AV_LOG_ERROR, "Unsupported video format\n");
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        break;
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    }
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    index_cnt = avio_rl32(pb);
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    vst->codec->width  = avio_rl32(pb);
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    vst->codec->height = avio_rl32(pb);
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    tb_num = avio_rl32(pb);
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    tb_den = avio_rl32(pb);
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    avpriv_set_pts_info(vst, 32, tb_num, tb_den);
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    vst->nb_frames = index_cnt;
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    vst->duration = index_cnt;
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    switch (avio_rl32(pb)) {
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    case 0:
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        audio_codec_id = AV_CODEC_ID_MP3;
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        break;
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    case 1:
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        av_log(s, AV_LOG_ERROR, "AAC not yet correctly supported\n");
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        audio_codec_id = AV_CODEC_ID_AAC;
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        break;
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    default:
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        av_log(s, AV_LOG_ERROR, "Unsupported audio format\n");
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        break;
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    }
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    pmp->num_streams = avio_rl16(pb) + 1;
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    avio_skip(pb, 10);
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    srate = avio_rl32(pb);
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    channels = avio_rl32(pb) + 1;
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    pos = avio_tell(pb) + 4LL*index_cnt;
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    for (i = 0; i < index_cnt; i++) {
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        uint32_t size = avio_rl32(pb);
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        int flags = size & 1 ? AVINDEX_KEYFRAME : 0;
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        if (url_feof(pb)) {
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            av_log(s, AV_LOG_FATAL, "Encountered EOF while reading index.\n");
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            return AVERROR_INVALIDDATA;
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        }
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        size >>= 1;
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        if (size < 9 + 4*pmp->num_streams) {
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            av_log(s, AV_LOG_ERROR, "Packet too small\n");
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            return AVERROR_INVALIDDATA;
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        }
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        av_add_index_entry(vst, pos, i, size, 0, flags);
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        pos += size;
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        if (fsize > 0 && i == 0 && pos > fsize) {
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            av_log(s, AV_LOG_ERROR, "File ends before first packet\n");
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            return AVERROR_INVALIDDATA;
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        }
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    }
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    for (i = 1; i < pmp->num_streams; i++) {
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        AVStream *ast = avformat_new_stream(s, NULL);
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        if (!ast)
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            return AVERROR(ENOMEM);
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        ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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        ast->codec->codec_id = audio_codec_id;
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        ast->codec->channels = channels;
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        ast->codec->sample_rate = srate;
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        avpriv_set_pts_info(ast, 32, 1, srate);
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    }
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    return 0;
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}
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static int pmp_packet(AVFormatContext *s, AVPacket *pkt)
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{
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    PMPContext *pmp = s->priv_data;
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    AVIOContext *pb = s->pb;
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    int ret = 0;
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    int i;
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    if (url_feof(pb))
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        return AVERROR_EOF;
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    if (pmp->cur_stream == 0) {
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        int num_packets;
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        pmp->audio_packets = avio_r8(pb);
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        if (!pmp->audio_packets) {
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            avpriv_request_sample(s, "0 audio packets");
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            return AVERROR_PATCHWELCOME;
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        }
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        num_packets = (pmp->num_streams - 1) * pmp->audio_packets + 1;
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        avio_skip(pb, 8);
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        pmp->current_packet = 0;
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        av_fast_malloc(&pmp->packet_sizes,
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                       &pmp->packet_sizes_alloc,
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                       num_packets * sizeof(*pmp->packet_sizes));
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        if (!pmp->packet_sizes_alloc) {
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            av_log(s, AV_LOG_ERROR, "Cannot (re)allocate packet buffer\n");
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            return AVERROR(ENOMEM);
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        }
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        for (i = 0; i < num_packets; i++)
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            pmp->packet_sizes[i] = avio_rl32(pb);
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    }
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    ret = av_get_packet(pb, pkt, pmp->packet_sizes[pmp->current_packet]);
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    if (ret >= 0) {
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        ret = 0;
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        // FIXME: this is a hack that should be removed once
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        // compute_pkt_fields() can handle timestamps properly
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        if (pmp->cur_stream == 0)
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            pkt->dts = s->streams[0]->cur_dts++;
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        pkt->stream_index = pmp->cur_stream;
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    }
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    if (pmp->current_packet % pmp->audio_packets == 0)
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        pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams;
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    pmp->current_packet++;
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    return ret;
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}
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static int pmp_seek(AVFormatContext *s, int stream_index, int64_t ts, int flags)
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{
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    PMPContext *pmp = s->priv_data;
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    pmp->cur_stream = 0;
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    // fall back on default seek now
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    return -1;
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}
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static int pmp_close(AVFormatContext *s)
182
{
183
    PMPContext *pmp = s->priv_data;
184
    av_freep(&pmp->packet_sizes);
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    return 0;
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}
187
 
188
AVInputFormat ff_pmp_demuxer = {
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    .name           = "pmp",
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    .long_name      = NULL_IF_CONFIG_SMALL("Playstation Portable PMP"),
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    .priv_data_size = sizeof(PMPContext),
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    .read_probe     = pmp_probe,
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    .read_header    = pmp_header,
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    .read_packet    = pmp_packet,
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    .read_seek      = pmp_seek,
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    .read_close     = pmp_close,
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};