mirror of https://github.com/FFmpeg/FFmpeg.git
* qatar/master: (44 commits) replacement Indeo 3 decoder gsm demuxer: do not allocate packet twice. flvenc: use first packet delay as global delay. ac3enc: doxygen update. imc: return error codes instead of 0 for error conditions. imc: return meaningful error codes instead of -1 imc: do not set channel layout for stereo imc: validate channel count imc: check for ff_fft_init() failure imc: check output buffer size before decoding imc: use DSPContext.bswap16_buf() to byte-swap packet data rtsp: add allowed_media_types option libgsm: add flush function to reset the decoder state when seeking libgsm: simplify decoding by using a loop gsm: log error message when packet is too small libgsmdec: do not needlessly set *data_size to 0 gsmdec: do not needlessly set *data_size to 0 gsmdec: add flush function to reset the decoder state when seeking libgsmdec: check output buffer size before decoding gsmdec: log error message when output buffer is too small. ... Conflicts: Changelog ffplay.c libavcodec/indeo3.c libavcodec/mjpeg_parser.c libavcodec/vp3.c libavformat/cutils.c libavformat/id3v2.c libavutil/parseutils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>pull/2/head
Michael Niedermayer
13 years ago
commit
988f585fcb
75 changed files with 1853 additions and 3644 deletions
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/*
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* RAW GSM demuxer |
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* Copyright (c) 2011 Justin Ruggles |
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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/mathematics.h" |
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#include "libavutil/opt.h" |
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#include "avformat.h" |
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#define GSM_BLOCK_SIZE 33 |
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#define GSM_BLOCK_SAMPLES 160 |
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#define GSM_SAMPLE_RATE 8000 |
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typedef struct { |
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AVClass *class; |
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int sample_rate; |
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} GSMDemuxerContext; |
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static int gsm_read_packet(AVFormatContext *s, AVPacket *pkt) |
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{ |
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int ret, size; |
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size = GSM_BLOCK_SIZE * 32; |
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pkt->pos = avio_tell(s->pb); |
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pkt->stream_index = 0; |
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ret = av_get_packet(s->pb, pkt, size); |
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if (ret < GSM_BLOCK_SIZE) { |
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av_free_packet(pkt); |
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return ret < 0 ? ret : AVERROR(EIO); |
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} |
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pkt->size = ret; |
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pkt->duration = ret / GSM_BLOCK_SIZE; |
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pkt->pts = pkt->pos / GSM_BLOCK_SIZE; |
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return 0; |
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} |
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static int gsm_read_header(AVFormatContext *s, AVFormatParameters *ap) |
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{ |
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GSMDemuxerContext *c = s->priv_data; |
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AVStream *st = avformat_new_stream(s, NULL); |
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if (!st) |
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return AVERROR(ENOMEM); |
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO; |
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st->codec->codec_id = s->iformat->value; |
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st->codec->channels = 1; |
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st->codec->sample_rate = c->sample_rate; |
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st->codec->block_align = GSM_BLOCK_SIZE; |
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st->codec->bit_rate = GSM_BLOCK_SIZE * 8 * c->sample_rate / GSM_BLOCK_SAMPLES; |
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av_set_pts_info(st, 64, GSM_BLOCK_SAMPLES, GSM_SAMPLE_RATE); |
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return 0; |
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} |
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static int gsm_read_seek2(AVFormatContext *s, int stream_index, int64_t min_ts, |
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int64_t ts, int64_t max_ts, int flags) |
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{ |
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GSMDemuxerContext *c = s->priv_data; |
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/* convert timestamps to file positions */ |
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if (!(flags & AVSEEK_FLAG_BYTE)) { |
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if (stream_index < 0) { |
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AVRational bitrate_q = { GSM_BLOCK_SAMPLES, c->sample_rate * GSM_BLOCK_SIZE }; |
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ts = av_rescale_q(ts, AV_TIME_BASE_Q, bitrate_q); |
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min_ts = av_rescale_q(min_ts, AV_TIME_BASE_Q, bitrate_q); |
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max_ts = av_rescale_q(max_ts, AV_TIME_BASE_Q, bitrate_q); |
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} else { |
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ts *= GSM_BLOCK_SIZE; |
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min_ts *= GSM_BLOCK_SIZE; |
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max_ts *= GSM_BLOCK_SIZE; |
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} |
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} |
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/* round to nearest block boundary */ |
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ts = (ts + GSM_BLOCK_SIZE / 2) / GSM_BLOCK_SIZE * GSM_BLOCK_SIZE; |
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ts = FFMAX(0, ts); |
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/* handle min/max */ |
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while (ts < min_ts) |
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ts += GSM_BLOCK_SIZE; |
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while (ts > max_ts) |
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ts -= GSM_BLOCK_SIZE; |
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if (ts < min_ts || ts > max_ts) |
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return -1; |
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return avio_seek(s->pb, ts, SEEK_SET); |
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} |
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static const AVOption options[] = { |
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{ "sample_rate", "", offsetof(GSMDemuxerContext, sample_rate), |
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AV_OPT_TYPE_INT, {.dbl = GSM_SAMPLE_RATE}, 1, INT_MAX / GSM_BLOCK_SIZE, |
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AV_OPT_FLAG_DECODING_PARAM }, |
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{ NULL }, |
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}; |
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static const AVClass class = { |
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.class_name = "gsm demuxer", |
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.item_name = av_default_item_name, |
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.option = options, |
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.version = LIBAVUTIL_VERSION_INT, |
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}; |
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AVInputFormat ff_gsm_demuxer = { |
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.name = "gsm", |
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.long_name = NULL_IF_CONFIG_SMALL("raw GSM"), |
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.priv_data_size = sizeof(GSMDemuxerContext), |
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.read_header = gsm_read_header, |
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.read_packet = gsm_read_packet, |
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.read_seek2 = gsm_read_seek2, |
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.extensions = "gsm", |
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.value = CODEC_ID_GSM, |
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.priv_class = &class, |
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}; |
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