Mercurial > libavformat.hg
view iff.c @ 4874:5370b0c1653c libavformat
Increase the SDP buffer size (again!) and also increase the temporary
buffer size of the fmtp parameter buffer. For Vorbis RT(S)P, these
contain full Vorbis headers, which can be up to 12kb each, formatted
in base64, so 16kb total. Patch required for proper Vorbis/RTP playback,
submitted as GSoC qualification task in the thread "RTP/Vorbis payload
implementation (GSoC qual task)" by Colin McQuillan m.niloc googlemail
com.
author | rbultje |
---|---|
date | Tue, 14 Apr 2009 13:22:40 +0000 |
parents | 49c1d3b27727 |
children | c76c1e0c50a5 |
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/* * IFF (.iff) file demuxer * Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net> * * 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 libavformat/iff.c * IFF file demuxer * by Jaikrishnan Menon * for more information on the .iff file format, visit: * http://wiki.multimedia.cx/index.php?title=IFF */ #include "libavutil/intreadwrite.h" #include "avformat.h" #define ID_8SVX MKTAG('8','S','V','X') #define ID_VHDR MKTAG('V','H','D','R') #define ID_ATAK MKTAG('A','T','A','K') #define ID_RLSE MKTAG('R','L','S','E') #define ID_CHAN MKTAG('C','H','A','N') #define ID_FORM MKTAG('F','O','R','M') #define ID_ANNO MKTAG('A','N','N','O') #define ID_AUTH MKTAG('A','U','T','H') #define ID_CHRS MKTAG('C','H','R','S') #define ID_COPYRIGHT MKTAG('(','c',')',' ') #define ID_CSET MKTAG('C','S','E','T') #define ID_FVER MKTAG('F','V','E','R') #define ID_NAME MKTAG('N','A','M','E') #define ID_TEXT MKTAG('T','E','X','T') #define ID_BODY MKTAG('B','O','D','Y') #define LEFT 2 #define RIGHT 4 #define STEREO 6 #define PACKET_SIZE 1024 typedef enum {COMP_NONE, COMP_FIB, COMP_EXP} svx8_compression_type; typedef struct { uint32_t body_size; uint32_t sent_bytes; uint32_t audio_frame_count; } IffDemuxContext; static void interleave_stereo(const uint8_t *src, uint8_t *dest, int size) { uint8_t *end = dest + size; size = size>>1; while(dest < end) { *dest++ = *src; *dest++ = *(src+size); src++; } } static int iff_probe(AVProbeData *p) { const uint8_t *d = p->buf; if ( AV_RL32(d) == ID_FORM && AV_RL32(d+8) == ID_8SVX) return AVPROBE_SCORE_MAX; return 0; } static int iff_read_header(AVFormatContext *s, AVFormatParameters *ap) { IffDemuxContext *iff = s->priv_data; ByteIOContext *pb = s->pb; AVStream *st; uint32_t chunk_id, data_size; int padding, done = 0; st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); st->codec->channels = 1; url_fskip(pb, 12); while(!done && !url_feof(pb)) { chunk_id = get_le32(pb); data_size = get_be32(pb); padding = data_size & 1; switch(chunk_id) { case ID_VHDR: url_fskip(pb, 12); st->codec->sample_rate = get_be16(pb); url_fskip(pb, 1); st->codec->codec_tag = get_byte(pb); url_fskip(pb, 4); break; case ID_BODY: iff->body_size = data_size; done = 1; break; case ID_CHAN: st->codec->channels = (get_be32(pb) < 6) ? 1 : 2; break; default: url_fseek(pb, data_size + padding, SEEK_CUR); break; } } if(!st->codec->sample_rate) return AVERROR_INVALIDDATA; av_set_pts_info(st, 32, 1, st->codec->sample_rate); st->codec->codec_type = CODEC_TYPE_AUDIO; switch(st->codec->codec_tag) { case COMP_NONE: st->codec->codec_id = CODEC_ID_PCM_S8; break; case COMP_FIB: st->codec->codec_id = CODEC_ID_8SVX_FIB; break; case COMP_EXP: st->codec->codec_id = CODEC_ID_8SVX_EXP; break; default: av_log(s, AV_LOG_ERROR, "iff: unknown compression method\n"); return -1; } st->codec->bits_per_coded_sample = 8; st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * st->codec->bits_per_coded_sample; st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample; return 0; } static int iff_read_packet(AVFormatContext *s, AVPacket *pkt) { IffDemuxContext *iff = s->priv_data; ByteIOContext *pb = s->pb; int ret; if(iff->sent_bytes > iff->body_size) return AVERROR(EIO); if(s->streams[0]->codec->channels == 2) { uint8_t sample_buffer[PACKET_SIZE]; ret = get_buffer(pb, sample_buffer, PACKET_SIZE); if(av_new_packet(pkt, PACKET_SIZE) < 0) { av_log(s, AV_LOG_ERROR, "iff: cannot allocate packet \n"); return AVERROR(ENOMEM); } interleave_stereo(sample_buffer, pkt->data, PACKET_SIZE); } else { ret = av_get_packet(pb, pkt, PACKET_SIZE); } if(iff->sent_bytes == 0) pkt->flags |= PKT_FLAG_KEY; iff->sent_bytes += PACKET_SIZE; pkt->stream_index = 0; pkt->pts = iff->audio_frame_count; iff->audio_frame_count += ret / s->streams[0]->codec->channels; return ret; } AVInputFormat iff_demuxer = { "IFF", NULL_IF_CONFIG_SMALL("IFF format"), sizeof(IffDemuxContext), iff_probe, iff_read_header, iff_read_packet, };