Mercurial > libavformat.hg
view vocdec.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 | 7d2f3f1b68d8 |
children | 33a244b7ca65 |
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/* * Creative Voice File demuxer. * Copyright (c) 2006 Aurelien Jacobs <aurel@gnuage.org> * * 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 "voc.h" static int voc_probe(AVProbeData *p) { int version, check; if (memcmp(p->buf, ff_voc_magic, sizeof(ff_voc_magic) - 1)) return 0; version = AV_RL16(p->buf + 22); check = AV_RL16(p->buf + 24); if (~version + 0x1234 != check) return 10; return AVPROBE_SCORE_MAX; } static int voc_read_header(AVFormatContext *s, AVFormatParameters *ap) { VocDecContext *voc = s->priv_data; ByteIOContext *pb = s->pb; int header_size; AVStream *st; url_fskip(pb, 20); header_size = get_le16(pb) - 22; if (header_size != 4) { av_log(s, AV_LOG_ERROR, "unknown header size: %d\n", header_size); return AVERROR(ENOSYS); } url_fskip(pb, header_size); st = av_new_stream(s, 0); if (!st) return AVERROR(ENOMEM); st->codec->codec_type = CODEC_TYPE_AUDIO; voc->remaining_size = 0; return 0; } int voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) { VocDecContext *voc = s->priv_data; AVCodecContext *dec = st->codec; ByteIOContext *pb = s->pb; VocType type; int size; int sample_rate = 0; int channels = 1; while (!voc->remaining_size) { type = get_byte(pb); if (type == VOC_TYPE_EOF) return AVERROR(EIO); voc->remaining_size = get_le24(pb); max_size -= 4; switch (type) { case VOC_TYPE_VOICE_DATA: dec->sample_rate = 1000000 / (256 - get_byte(pb)); if (sample_rate) dec->sample_rate = sample_rate; dec->channels = channels; dec->codec_id = codec_get_id(ff_voc_codec_tags, get_byte(pb)); dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id); voc->remaining_size -= 2; max_size -= 2; channels = 1; break; case VOC_TYPE_VOICE_DATA_CONT: break; case VOC_TYPE_EXTENDED: sample_rate = get_le16(pb); get_byte(pb); channels = get_byte(pb) + 1; sample_rate = 256000000 / (channels * (65536 - sample_rate)); voc->remaining_size = 0; max_size -= 4; break; case VOC_TYPE_NEW_VOICE_DATA: dec->sample_rate = get_le32(pb); dec->bits_per_coded_sample = get_byte(pb); dec->channels = get_byte(pb); dec->codec_id = codec_get_id(ff_voc_codec_tags, get_le16(pb)); url_fskip(pb, 4); voc->remaining_size -= 12; max_size -= 12; break; default: url_fskip(pb, voc->remaining_size); max_size -= voc->remaining_size; voc->remaining_size = 0; break; } } dec->bit_rate = dec->sample_rate * dec->bits_per_coded_sample; if (max_size <= 0) max_size = 2048; size = FFMIN(voc->remaining_size, max_size); voc->remaining_size -= size; return av_get_packet(pb, pkt, size); } static int voc_read_packet(AVFormatContext *s, AVPacket *pkt) { return voc_get_packet(s, pkt, s->streams[0], 0); } AVInputFormat voc_demuxer = { "voc", NULL_IF_CONFIG_SMALL("Creative Voice file format"), sizeof(VocDecContext), voc_probe, voc_read_header, voc_read_packet, .codec_tag=(const AVCodecTag* const []){ff_voc_codec_tags, 0}, };