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180 lines
5.3 KiB
180 lines
5.3 KiB
/* |
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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 Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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 PMPContext { |
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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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{ |
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if (!memcmp(p->buf, "pmpm\1\0\0\0", 8)) |
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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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int 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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int i; |
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uint64_t pos; |
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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_WARNING, "AAC is 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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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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pos = avio_tell(pb) + 4 * index_cnt; |
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for (i = 0; i < index_cnt; i++) { |
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int size = avio_rl32(pb); |
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int flags = size & 1 ? AVINDEX_KEYFRAME : 0; |
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size >>= 1; |
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av_add_index_entry(vst, pos, i, size, 0, flags); |
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pos += size; |
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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 (pb->eof_reached) |
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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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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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pmp->current_packet++; |
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if (pmp->current_packet == 1 || pmp->current_packet > pmp->audio_packets) |
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pmp->cur_stream = (pmp->cur_stream + 1) % pmp->num_streams; |
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return ret; |
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} |
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static int pmp_seek(AVFormatContext *s, int stream_idx, 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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// fallback to default seek now |
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return -1; |
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} |
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static int pmp_close(AVFormatContext *s) |
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{ |
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PMPContext *pmp = s->priv_data; |
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av_freep(&pmp->packet_sizes); |
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return 0; |
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} |
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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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};
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