| /* |
| * 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 <stdlib.h> |
| #include <stdio.h> |
| #include <stdint.h> |
| #include <string.h> |
| #include <errno.h> |
| #if HAVE_SOUNDCARD_H |
| #include <soundcard.h> |
| #else |
| #include <sys/soundcard.h> |
| #endif |
| #include <unistd.h> |
| #include <fcntl.h> |
| #include <sys/ioctl.h> |
| |
| #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 |