Mercurial > libavformat.hg
view audio.c @ 2136:b53a19eed95d libavformat
remove duplicate PAT scaning code and actual PAT scan
also disabling the nonsense 3yr old rawts change (it would after the
above chage totally break the demuxer if its left enabled)
author | michael |
---|---|
date | Mon, 04 Jun 2007 14:44:52 +0000 |
parents | 69ff78484350 |
children | b21c2af60bc9 |
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/* * 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 "avformat.h" #include <stdlib.h> #include <stdio.h> #include <string.h> #ifdef HAVE_SOUNDCARD_H #include <soundcard.h> #else #include <sys/soundcard.h> #endif #include <unistd.h> #include <fcntl.h> #include <sys/ioctl.h> #include <sys/mman.h> #include <sys/time.h> #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_t buffer[AUDIO_BLOCK_SIZE]; int buffer_ptr; } AudioData; static int audio_open(AudioData *s, int is_output, const char *audio_device) { int audio_fd; int tmp, err; char *flip = getenv("AUDIO_FLIP_LEFT"); 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 AVERROR_IO; } 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: av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n"); close(audio_fd); return AVERROR_IO; } 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 AVERROR_IO; } 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, s1->filename); if (ret < 0) { return AVERROR_IO; } 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 = 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 AVERROR_IO; } 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->sample_rate <= 0 || ap->channels <= 0) return -1; st = av_new_stream(s1, 0); if (!st) { return AVERROR(ENOMEM); } s->sample_rate = ap->sample_rate; s->channels = ap->channels; ret = audio_open(s, 0, s1->filename); if (ret < 0) { av_free(st); return AVERROR_IO; } /* 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(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 (av_new_packet(pkt, s->frame_size) < 0) return AVERROR_IO; for(;;) { struct timeval tv; fd_set fds; tv.tv_sec = 0; tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */ FD_ZERO(&fds); FD_SET(s->fd, &fds); /* This will block until data is available or we get a timeout */ (void) select(s->fd + 1, &fds, 0, 0, &tv); 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; pkt->pts = av_gettime(); return 0; } if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) { av_free_packet(pkt); return AVERROR_IO; } } 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; 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; } #ifdef CONFIG_AUDIO_DEMUXER AVInputFormat audio_demuxer = { "audio_device", "audio grab and output", sizeof(AudioData), NULL, audio_read_header, audio_read_packet, audio_read_close, .flags = AVFMT_NOFILE, }; #endif #ifdef CONFIG_AUDIO_MUXER AVOutputFormat audio_muxer = { "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, }; #endif