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	6f69f7a8bf
	
	
	
		
			
			* commit '9200514ad8717c63f82101dc394f4378854325bf':
  lavf: replace AVStream.codec with AVStream.codecpar
This has been a HUGE effort from:
    - Derek Buitenhuis <derek.buitenhuis@gmail.com>
    - Hendrik Leppkes <h.leppkes@gmail.com>
    - wm4 <nfxjfg@googlemail.com>
    - Clément Bœsch <clement@stupeflix.com>
    - James Almer <jamrial@gmail.com>
    - Michael Niedermayer <michael@niedermayer.cc>
    - Rostislav Pehlivanov <atomnuker@gmail.com>
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
		
	
		
			
				
	
	
		
			150 lines
		
	
	
		
			4.0 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			150 lines
		
	
	
		
			4.0 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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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 FFmpeg.
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|  *
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|  * FFmpeg 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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|  * FFmpeg 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 FFmpeg; 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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| 
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| #include "config.h"
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| 
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| #include <stdint.h>
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| 
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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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| 
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| #if HAVE_UNISTD_H
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| #include <unistd.h>
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| #endif
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| #include <fcntl.h>
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| #include <sys/ioctl.h>
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| 
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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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| 
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| #include "libavcodec/avcodec.h"
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| 
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| #include "avdevice.h"
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| #include "libavformat/internal.h"
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| 
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| #include "oss.h"
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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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|     if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
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|         return ret;
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| 
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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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| 
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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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| 
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|     /* convert to wanted units */
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|     pkt->pts = cur_time;
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| 
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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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| 
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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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| 
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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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| 
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|     ff_oss_audio_close(s);
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|     return 0;
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| }
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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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| 
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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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|     .category       = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
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| };
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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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