mirror of https://github.com/FFmpeg/FFmpeg.git
parent
8d3db95f20
commit
6ce13070bd
4 changed files with 117 additions and 198 deletions
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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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#ifndef AVDEVICE_OSS_H |
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#define AVDEVICE_OSS_H |
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#include "libavcodec/avcodec.h" |
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#include "libavformat/avformat.h" |
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#define OSS_AUDIO_BLOCK_SIZE 4096 |
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typedef struct OSSAudioData { |
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AVClass *class; |
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int fd; |
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int sample_rate; |
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int channels; |
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int frame_size; /* in bytes ! */ |
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enum AVCodecID codec_id; |
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unsigned int flip_left : 1; |
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uint8_t buffer[OSS_AUDIO_BLOCK_SIZE]; |
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int buffer_ptr; |
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} OSSAudioData; |
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int ff_oss_audio_open(AVFormatContext *s1, int is_output, |
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const char *audio_device); |
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int ff_oss_audio_close(OSSAudioData *s); |
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#endif /* AVDEVICE_OSS_H */ |
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@ -1,146 +0,0 @@ |
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/*
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* Linux audio play interface |
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* Copyright (c) 2000, 2001 Fabrice Bellard |
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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 "config.h" |
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#include <stdint.h> |
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#if HAVE_SOUNDCARD_H |
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#include <soundcard.h> |
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#else |
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#include <sys/soundcard.h> |
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#endif |
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#include <unistd.h> |
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#include <fcntl.h> |
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#include <sys/ioctl.h> |
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#include "libavutil/internal.h" |
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#include "libavutil/opt.h" |
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#include "libavutil/time.h" |
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#include "libavcodec/avcodec.h" |
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#include "libavformat/avformat.h" |
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#include "libavformat/internal.h" |
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#include "oss.h" |
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static int audio_read_header(AVFormatContext *s1) |
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{ |
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OSSAudioData *s = s1->priv_data; |
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AVStream *st; |
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int ret; |
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st = avformat_new_stream(s1, NULL); |
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if (!st) { |
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return AVERROR(ENOMEM); |
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} |
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ret = ff_oss_audio_open(s1, 0, s1->filename); |
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if (ret < 0) { |
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return AVERROR(EIO); |
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} |
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/* take real parameters */ |
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st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; |
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st->codecpar->codec_id = s->codec_id; |
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st->codecpar->sample_rate = s->sample_rate; |
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st->codecpar->channels = s->channels; |
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avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ |
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return 0; |
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} |
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static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) |
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{ |
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OSSAudioData *s = s1->priv_data; |
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int ret, bdelay; |
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int64_t cur_time; |
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struct audio_buf_info abufi; |
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if ((ret=av_new_packet(pkt, s->frame_size)) < 0) |
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return ret; |
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ret = read(s->fd, pkt->data, pkt->size); |
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if (ret <= 0){ |
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av_packet_unref(pkt); |
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pkt->size = 0; |
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if (ret<0) return AVERROR(errno); |
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else return AVERROR_EOF; |
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} |
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pkt->size = ret; |
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/* compute pts of the start of the packet */ |
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cur_time = av_gettime(); |
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bdelay = ret; |
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if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) { |
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bdelay += abufi.bytes; |
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} |
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/* subtract time represented by the number of bytes in the audio fifo */ |
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cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels); |
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/* convert to wanted units */ |
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pkt->pts = cur_time; |
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if (s->flip_left && s->channels == 2) { |
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int i; |
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short *p = (short *) pkt->data; |
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for (i = 0; i < ret; i += 4) { |
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*p = ~*p; |
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p += 2; |
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} |
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} |
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return 0; |
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} |
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static int audio_read_close(AVFormatContext *s1) |
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{ |
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OSSAudioData *s = s1->priv_data; |
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ff_oss_audio_close(s); |
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return 0; |
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} |
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static const AVOption options[] = { |
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{ "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, |
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{ "channels", "", offsetof(OSSAudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, |
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{ NULL }, |
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}; |
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static const AVClass oss_demuxer_class = { |
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.class_name = "OSS 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_oss_demuxer = { |
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.name = "oss", |
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.long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"), |
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.priv_data_size = sizeof(OSSAudioData), |
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.read_header = audio_read_header, |
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.read_packet = audio_read_packet, |
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.read_close = audio_read_close, |
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.flags = AVFMT_NOFILE, |
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.priv_class = &oss_demuxer_class, |
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}; |
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