diff options
Diffstat (limited to 'libavformat/audio.c')
| -rw-r--r-- | libavformat/audio.c | 330 |
1 files changed, 330 insertions, 0 deletions
diff --git a/libavformat/audio.c b/libavformat/audio.c new file mode 100644 index 0000000000..4fa155c85d --- /dev/null +++ b/libavformat/audio.c @@ -0,0 +1,330 @@ +/* + * Linux audio play and grab interface + * Copyright (c) 2000, 2001 Fabrice Bellard. + * + * This library 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 of the License, or (at your option) any later version. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include "avformat.h" + +#include <stdlib.h> +#include <stdio.h> +#include <string.h> +#include <sys/soundcard.h> +#include <unistd.h> +#include <fcntl.h> +#include <sys/ioctl.h> +#include <sys/mman.h> +#include <sys/time.h> + +const char *audio_device = "/dev/dsp"; + +#define AUDIO_BLOCK_SIZE 4096 + +typedef struct { + int fd; + int sample_rate; + int channels; + int frame_size; /* in bytes ! */ + int codec_id; + int flip_left : 1; + UINT8 buffer[AUDIO_BLOCK_SIZE]; + int buffer_ptr; +} AudioData; + +static int audio_open(AudioData *s, int is_output) +{ + int audio_fd; + int tmp, err; + char *flip = getenv("AUDIO_FLIP_LEFT"); + + /* open linux audio device */ + if (is_output) + audio_fd = open(audio_device, O_WRONLY); + else + audio_fd = open(audio_device, O_RDONLY); + if (audio_fd < 0) { + perror(audio_device); + return -EIO; + } + + if (flip && *flip == '1') { + s->flip_left = 1; + } + + /* non blocking mode */ + if (!is_output) + fcntl(audio_fd, F_SETFL, O_NONBLOCK); + + s->frame_size = AUDIO_BLOCK_SIZE; +#if 0 + tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS; + err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp); + if (err < 0) { + perror("SNDCTL_DSP_SETFRAGMENT"); + } +#endif + + /* select format : favour native format */ + err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp); + +#ifdef WORDS_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 = CODEC_ID_PCM_S16LE; + break; + case AFMT_S16_BE: + s->codec_id = CODEC_ID_PCM_S16BE; + break; + default: + fprintf(stderr, "Soundcard does not support 16 bit sample format\n"); + close(audio_fd); + return -EIO; + } + err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp); + if (err < 0) { + perror("SNDCTL_DSP_SETFMT"); + goto fail; + } + + tmp = (s->channels == 2); + err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp); + if (err < 0) { + perror("SNDCTL_DSP_STEREO"); + goto fail; + } + if (tmp) + s->channels = 2; + + tmp = s->sample_rate; + err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp); + if (err < 0) { + perror("SNDCTL_DSP_SPEED"); + goto fail; + } + s->sample_rate = tmp; /* store real sample rate */ + s->fd = audio_fd; + + return 0; + fail: + close(audio_fd); + return -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(s, 1); + if (ret < 0) { + return -EIO; + } else { + return 0; + } +} + +static int audio_write_packet(AVFormatContext *s1, int stream_index, + UINT8 *buf, int size, int force_pts) +{ + AudioData *s = s1->priv_data; + int len, ret; + + while (size > 0) { + len = AUDIO_BLOCK_SIZE - s->buffer_ptr; + if (len > size) + len = 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 -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, AVFormatParameters *ap) +{ + AudioData *s = s1->priv_data; + AVStream *st; + int ret; + + if (!ap || ap->sample_rate <= 0 || ap->channels <= 0) + return -1; + + st = av_new_stream(s1, 0); + if (!st) { + return -ENOMEM; + } + s->sample_rate = ap->sample_rate; + s->channels = ap->channels; + + ret = audio_open(s, 0); + if (ret < 0) { + av_free(st); + return -EIO; + } + + /* take real parameters */ + st->codec.codec_type = CODEC_TYPE_AUDIO; + st->codec.codec_id = s->codec_id; + st->codec.sample_rate = s->sample_rate; + st->codec.channels = s->channels; + + av_set_pts_info(s1, 48, 1, 1000000); /* 48 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 (av_new_packet(pkt, s->frame_size) < 0) + return -EIO; + for(;;) { + ret = read(s->fd, pkt->data, pkt->size); + if (ret > 0) + break; + if (ret == -1 && (errno == EAGAIN || errno == EINTR)) { + av_free_packet(pkt); + pkt->size = 0; + return 0; + } + if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) { + av_free_packet(pkt); + return -EIO; + } + } + 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; + } + /* substract 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 & ((1LL << 48) - 1); + + 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; +} + +static AVInputFormat audio_in_format = { + "audio_device", + "audio grab and output", + sizeof(AudioData), + NULL, + audio_read_header, + audio_read_packet, + audio_read_close, + .flags = AVFMT_NOFILE, +}; + +static AVOutputFormat audio_out_format = { + "audio_device", + "audio grab and output", + "", + "", + 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 */ +#ifdef WORDS_BIGENDIAN + CODEC_ID_PCM_S16BE, +#else + CODEC_ID_PCM_S16LE, +#endif + CODEC_ID_NONE, + audio_write_header, + audio_write_packet, + audio_write_trailer, + .flags = AVFMT_NOFILE, +}; + +int audio_init(void) +{ + av_register_input_format(&audio_in_format); + av_register_output_format(&audio_out_format); + return 0; +} |