/* * RAW PCM demuxers * Copyright (c) 2002 Fabrice Bellard * * 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 "avformat.h" #include "internal.h" #include "pcm.h" #include "libavutil/log.h" #include "libavutil/opt.h" #include "libavutil/avassert.h" typedef struct PCMAudioDemuxerContext { AVClass *class; int sample_rate; int channels; } PCMAudioDemuxerContext; static int pcm_read_header(AVFormatContext *s) { PCMAudioDemuxerContext *s1 = s->priv_data; AVStream *st; uint8_t *mime_type = NULL; st = avformat_new_stream(s, NULL); if (!st) return AVERROR(ENOMEM); st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->codec_id = s->iformat->raw_codec_id; st->codecpar->sample_rate = s1->sample_rate; st->codecpar->channels = s1->channels; av_opt_get(s->pb, "mime_type", AV_OPT_SEARCH_CHILDREN, &mime_type); if (mime_type && s->iformat->mime_type) { int rate = 0, channels = 0; size_t len = strlen(s->iformat->mime_type); if (!strncmp(s->iformat->mime_type, mime_type, len)) { uint8_t *options = mime_type + len; len = strlen(mime_type); while (options < mime_type + len) { options = strstr(options, ";"); if (!options++) break; if (!rate) sscanf(options, " rate=%d", &rate); if (!channels) sscanf(options, " channels=%d", &channels); } if (rate <= 0) { av_log(s, AV_LOG_ERROR, "Invalid sample_rate found in mime_type \"%s\"\n", mime_type); av_freep(&mime_type); return AVERROR_INVALIDDATA; } st->codecpar->sample_rate = rate; if (channels > 0) st->codecpar->channels = channels; } } av_freep(&mime_type); st->codecpar->bits_per_coded_sample = av_get_bits_per_sample(st->codecpar->codec_id); av_assert0(st->codecpar->bits_per_coded_sample > 0); st->codecpar->block_align = st->codecpar->bits_per_coded_sample * st->codecpar->channels / 8; avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate); return 0; } static const AVOption pcm_options[] = { { "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 44100}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, { "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, { NULL }, }; #define PCMDEF(name_, long_name_, ext, codec, ...) \ static const AVClass name_ ## _demuxer_class = { \ .class_name = #name_ " demuxer", \ .item_name = av_default_item_name, \ .option = pcm_options, \ .version = LIBAVUTIL_VERSION_INT, \ }; \ AVInputFormat ff_pcm_ ## name_ ## _demuxer = { \ .name = #name_, \ .long_name = NULL_IF_CONFIG_SMALL(long_name_), \ .priv_data_size = sizeof(PCMAudioDemuxerContext), \ .read_header = pcm_read_header, \ .read_packet = ff_pcm_read_packet, \ .read_seek = ff_pcm_read_seek, \ .flags = AVFMT_GENERIC_INDEX, \ .extensions = ext, \ .raw_codec_id = codec, \ .priv_class = &name_ ## _demuxer_class, \ __VA_ARGS__ \ }; PCMDEF(f64be, "PCM 64-bit floating-point big-endian", NULL, AV_CODEC_ID_PCM_F64BE) PCMDEF(f64le, "PCM 64-bit floating-point little-endian", NULL, AV_CODEC_ID_PCM_F64LE) PCMDEF(f32be, "PCM 32-bit floating-point big-endian", NULL, AV_CODEC_ID_PCM_F32BE) PCMDEF(f32le, "PCM 32-bit floating-point little-endian", NULL, AV_CODEC_ID_PCM_F32LE) PCMDEF(s32be, "PCM signed 32-bit big-endian", NULL, AV_CODEC_ID_PCM_S32BE) PCMDEF(s32le, "PCM signed 32-bit little-endian", NULL, AV_CODEC_ID_PCM_S32LE) PCMDEF(s24be, "PCM signed 24-bit big-endian", NULL, AV_CODEC_ID_PCM_S24BE) PCMDEF(s24le, "PCM signed 24-bit little-endian", NULL, AV_CODEC_ID_PCM_S24LE) PCMDEF(s16be, "PCM signed 16-bit big-endian", AV_NE("sw", NULL), AV_CODEC_ID_PCM_S16BE) PCMDEF(s16le, "PCM signed 16-bit little-endian", AV_NE(NULL, "sw"), AV_CODEC_ID_PCM_S16LE, .mime_type = "audio/L16",) PCMDEF(s8, "PCM signed 8-bit", "sb", AV_CODEC_ID_PCM_S8) PCMDEF(u32be, "PCM unsigned 32-bit big-endian", NULL, AV_CODEC_ID_PCM_U32BE) PCMDEF(u32le, "PCM unsigned 32-bit little-endian", NULL, AV_CODEC_ID_PCM_U32LE) PCMDEF(u24be, "PCM unsigned 24-bit big-endian", NULL, AV_CODEC_ID_PCM_U24BE) PCMDEF(u24le, "PCM unsigned 24-bit little-endian", NULL, AV_CODEC_ID_PCM_U24LE) PCMDEF(u16be, "PCM unsigned 16-bit big-endian", AV_NE("uw", NULL), AV_CODEC_ID_PCM_U16BE) PCMDEF(u16le, "PCM unsigned 16-bit little-endian", AV_NE(NULL, "uw"), AV_CODEC_ID_PCM_U16LE) PCMDEF(u8, "PCM unsigned 8-bit", "ub", AV_CODEC_ID_PCM_U8) PCMDEF(alaw, "PCM A-law", "al", AV_CODEC_ID_PCM_ALAW) PCMDEF(mulaw, "PCM mu-law", "ul", AV_CODEC_ID_PCM_MULAW) static const AVOption sln_options[] = { { "sample_rate", "", offsetof(PCMAudioDemuxerContext, sample_rate), AV_OPT_TYPE_INT, {.i64 = 8000}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, { "channels", "", offsetof(PCMAudioDemuxerContext, channels), AV_OPT_TYPE_INT, {.i64 = 1}, 0, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, { NULL }, }; static const AVClass sln_demuxer_class = { .class_name = "sln demuxer", .item_name = av_default_item_name, .option = sln_options, .version = LIBAVUTIL_VERSION_INT, }; AVInputFormat ff_sln_demuxer = { .name = "sln", .long_name = NULL_IF_CONFIG_SMALL("Asterisk raw pcm"), .priv_data_size = sizeof(PCMAudioDemuxerContext), .read_header = pcm_read_header, .read_packet = ff_pcm_read_packet, .read_seek = ff_pcm_read_seek, .flags = AVFMT_GENERIC_INDEX, .extensions = "sln", .raw_codec_id = AV_CODEC_ID_PCM_S16LE, .priv_class = &sln_demuxer_class, };