Mercurial > libavformat.hg
view audio.c @ 1755:41160bae24b5 libavformat
Fix audio corruption when cutting.
Patch by Wolfram Gloger, wmglo dent dot med dot uni-muenchen dot de.
author | takis |
---|---|
date | Mon, 05 Feb 2007 02:08:57 +0000 |
parents | 0899bfe4105c |
children | 2f59a73884af |
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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 __OpenBSD__ #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"); /* open linux audio device */ if (!audio_device) #ifdef __OpenBSD__ audio_device = "/dev/sound"; #else audio_device = "/dev/dsp"; #endif 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, NULL); 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 -ENOMEM; } s->sample_rate = ap->sample_rate; s->channels = ap->channels; ret = audio_open(s, 0, ap->device); 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