From 22e28216336da876e1fd17f380ce42eaf1446769 Mon Sep 17 00:00:00 2001 From: Tim Redfern Date: Mon, 17 Feb 2014 13:36:38 +0000 Subject: chasing indexing error --- ffmpeg/libavdevice/oss_audio.c | 328 ----------------------------------------- 1 file changed, 328 deletions(-) delete mode 100644 ffmpeg/libavdevice/oss_audio.c (limited to 'ffmpeg/libavdevice/oss_audio.c') diff --git a/ffmpeg/libavdevice/oss_audio.c b/ffmpeg/libavdevice/oss_audio.c deleted file mode 100644 index 916908c..0000000 --- a/ffmpeg/libavdevice/oss_audio.c +++ /dev/null @@ -1,328 +0,0 @@ -/* - * Linux audio play and grab interface - * Copyright (c) 2000, 2001 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 "config.h" -#include -#include -#include -#include -#include -#if HAVE_SOUNDCARD_H -#include -#else -#include -#endif -#include -#include -#include - -#include "libavutil/internal.h" -#include "libavutil/log.h" -#include "libavutil/opt.h" -#include "libavutil/time.h" -#include "libavcodec/avcodec.h" -#include "avdevice.h" -#include "libavformat/internal.h" - -#define AUDIO_BLOCK_SIZE 4096 - -typedef struct { - AVClass *class; - int fd; - int sample_rate; - int channels; - int frame_size; /* in bytes ! */ - enum AVCodecID codec_id; - unsigned int flip_left : 1; - uint8_t buffer[AUDIO_BLOCK_SIZE]; - int buffer_ptr; -} AudioData; - -static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device) -{ - AudioData *s = s1->priv_data; - int audio_fd; - int tmp, err; - char *flip = getenv("AUDIO_FLIP_LEFT"); - - if (is_output) - audio_fd = avpriv_open(audio_device, O_WRONLY); - else - audio_fd = avpriv_open(audio_device, O_RDONLY); - if (audio_fd < 0) { - av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno)); - return AVERROR(EIO); - } - - if (flip && *flip == '1') { - s->flip_left = 1; - } - - /* non blocking mode */ - if (!is_output) { - if (fcntl(audio_fd, F_SETFL, O_NONBLOCK) < 0) { - av_log(s1, AV_LOG_WARNING, "%s: Could not enable non block mode (%s)\n", audio_device, strerror(errno)); - } - } - - s->frame_size = AUDIO_BLOCK_SIZE; - - /* select format : favour native format */ - err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp); - -#if HAVE_BIGENDIAN - if (tmp & AFMT_S16_BE) { - tmp = AFMT_S16_BE; - } else if (tmp & AFMT_S16_LE) { - tmp = AFMT_S16_LE; - } else { - tmp = 0; - } -#else - if (tmp & AFMT_S16_LE) { - tmp = AFMT_S16_LE; - } else if (tmp & AFMT_S16_BE) { - tmp = AFMT_S16_BE; - } else { - tmp = 0; - } -#endif - - switch(tmp) { - case AFMT_S16_LE: - s->codec_id = AV_CODEC_ID_PCM_S16LE; - break; - case AFMT_S16_BE: - s->codec_id = AV_CODEC_ID_PCM_S16BE; - break; - default: - av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n"); - close(audio_fd); - return AVERROR(EIO); - } - err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp); - if (err < 0) { - av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno)); - goto fail; - } - - tmp = (s->channels == 2); - err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp); - if (err < 0) { - av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno)); - goto fail; - } - - tmp = s->sample_rate; - err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp); - if (err < 0) { - av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno)); - goto fail; - } - s->sample_rate = tmp; /* store real sample rate */ - s->fd = audio_fd; - - return 0; - fail: - close(audio_fd); - return AVERROR(EIO); -} - -static int audio_close(AudioData *s) -{ - close(s->fd); - return 0; -} - -/* sound output support */ -static int audio_write_header(AVFormatContext *s1) -{ - AudioData *s = s1->priv_data; - AVStream *st; - int ret; - - st = s1->streams[0]; - s->sample_rate = st->codec->sample_rate; - s->channels = st->codec->channels; - ret = audio_open(s1, 1, s1->filename); - if (ret < 0) { - return AVERROR(EIO); - } else { - return 0; - } -} - -static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) -{ - AudioData *s = s1->priv_data; - int len, ret; - int size= pkt->size; - uint8_t *buf= pkt->data; - - while (size > 0) { - len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size); - memcpy(s->buffer + s->buffer_ptr, buf, len); - s->buffer_ptr += len; - if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) { - for(;;) { - ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE); - if (ret > 0) - break; - if (ret < 0 && (errno != EAGAIN && errno != EINTR)) - return AVERROR(EIO); - } - s->buffer_ptr = 0; - } - buf += len; - size -= len; - } - return 0; -} - -static int audio_write_trailer(AVFormatContext *s1) -{ - AudioData *s = s1->priv_data; - - audio_close(s); - return 0; -} - -/* grab support */ - -static int audio_read_header(AVFormatContext *s1) -{ - AudioData *s = s1->priv_data; - AVStream *st; - int ret; - - st = avformat_new_stream(s1, NULL); - if (!st) { - return AVERROR(ENOMEM); - } - - ret = audio_open(s1, 0, s1->filename); - if (ret < 0) { - return AVERROR(EIO); - } - - /* take real parameters */ - st->codec->codec_type = AVMEDIA_TYPE_AUDIO; - st->codec->codec_id = s->codec_id; - st->codec->sample_rate = s->sample_rate; - st->codec->channels = s->channels; - - avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ - return 0; -} - -static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) -{ - AudioData *s = s1->priv_data; - int ret, bdelay; - int64_t cur_time; - struct audio_buf_info abufi; - - if ((ret=av_new_packet(pkt, s->frame_size)) < 0) - return ret; - - ret = read(s->fd, pkt->data, pkt->size); - if (ret <= 0){ - av_free_packet(pkt); - pkt->size = 0; - if (ret<0) return AVERROR(errno); - else return AVERROR_EOF; - } - pkt->size = ret; - - /* compute pts of the start of the packet */ - cur_time = av_gettime(); - bdelay = ret; - if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) { - bdelay += abufi.bytes; - } - /* subtract time represented by the number of bytes in the audio fifo */ - cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels); - - /* convert to wanted units */ - pkt->pts = cur_time; - - if (s->flip_left && s->channels == 2) { - int i; - short *p = (short *) pkt->data; - - for (i = 0; i < ret; i += 4) { - *p = ~*p; - p += 2; - } - } - return 0; -} - -static int audio_read_close(AVFormatContext *s1) -{ - AudioData *s = s1->priv_data; - - audio_close(s); - return 0; -} - -#if CONFIG_OSS_INDEV -static const AVOption options[] = { - { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, - { "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, - { NULL }, -}; - -static const AVClass oss_demuxer_class = { - .class_name = "OSS demuxer", - .item_name = av_default_item_name, - .option = options, - .version = LIBAVUTIL_VERSION_INT, -}; - -AVInputFormat ff_oss_demuxer = { - .name = "oss", - .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) capture"), - .priv_data_size = sizeof(AudioData), - .read_header = audio_read_header, - .read_packet = audio_read_packet, - .read_close = audio_read_close, - .flags = AVFMT_NOFILE, - .priv_class = &oss_demuxer_class, -}; -#endif - -#if CONFIG_OSS_OUTDEV -AVOutputFormat ff_oss_muxer = { - .name = "oss", - .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"), - .priv_data_size = sizeof(AudioData), - /* XXX: we make the assumption that the soundcard accepts this format */ - /* XXX: find better solution with "preinit" method, needed also in - other formats */ - .audio_codec = AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE), - .video_codec = AV_CODEC_ID_NONE, - .write_header = audio_write_header, - .write_packet = audio_write_packet, - .write_trailer = audio_write_trailer, - .flags = AVFMT_NOFILE, -}; -#endif -- cgit v1.2.3