From 33054e35e30ab69b600964b5f2ceefd7e4624033 Mon Sep 17 00:00:00 2001 From: Benoit Fouet Date: Wed, 21 Jan 2009 08:43:38 +0000 Subject: Rename audio.c to oss_audio.c in libavdevice. Originally committed as revision 16707 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavdevice/Makefile | 4 +- libavdevice/audio.c | 349 ------------------------------------------------ libavdevice/oss_audio.c | 349 ++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 351 insertions(+), 351 deletions(-) delete mode 100644 libavdevice/audio.c create mode 100644 libavdevice/oss_audio.c (limited to 'libavdevice') diff --git a/libavdevice/Makefile b/libavdevice/Makefile index 655c033b2c..31337766ea 100644 --- a/libavdevice/Makefile +++ b/libavdevice/Makefile @@ -10,8 +10,8 @@ OBJS = alldevices.o # input/output devices OBJS-$(CONFIG_BKTR_DEMUXER) += bktr.o OBJS-$(CONFIG_DV1394_DEMUXER) += dv1394.o -OBJS-$(CONFIG_OSS_DEMUXER) += audio.o -OBJS-$(CONFIG_OSS_MUXER) += audio.o +OBJS-$(CONFIG_OSS_DEMUXER) += oss_audio.o +OBJS-$(CONFIG_OSS_MUXER) += oss_audio.o OBJS-$(CONFIG_V4L2_DEMUXER) += v4l2.o OBJS-$(CONFIG_V4L_DEMUXER) += v4l.o OBJS-$(CONFIG_VFWCAP_DEMUXER) += vfwcap.o diff --git a/libavdevice/audio.c b/libavdevice/audio.c deleted file mode 100644 index 8f3e678db0..0000000000 --- a/libavdevice/audio.c +++ /dev/null @@ -1,349 +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 -#include - -#include "libavutil/log.h" -#include "libavcodec/avcodec.h" -#include "libavformat/avformat.h" - -#define AUDIO_BLOCK_SIZE 4096 - -typedef struct { - int fd; - int sample_rate; - int channels; - int frame_size; /* in bytes ! */ - enum CodecID 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 = open(audio_device, O_WRONLY); - else - audio_fd = 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) - 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(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 = 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(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->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(s1, 0, s1->filename); - if (ret < 0) { - av_free(st); - return AVERROR(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(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(EIO); - 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(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; - } - /* 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_DEMUXER -AVInputFormat oss_demuxer = { - "oss", - NULL_IF_CONFIG_SMALL("Open Sound System capture"), - sizeof(AudioData), - NULL, - audio_read_header, - audio_read_packet, - audio_read_close, - .flags = AVFMT_NOFILE, -}; -#endif - -#if CONFIG_OSS_MUXER -AVOutputFormat oss_muxer = { - "oss", - NULL_IF_CONFIG_SMALL("Open Sound System playback"), - "", - "", - 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 diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c new file mode 100644 index 0000000000..8f3e678db0 --- /dev/null +++ b/libavdevice/oss_audio.c @@ -0,0 +1,349 @@ +/* + * 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 +#include + +#include "libavutil/log.h" +#include "libavcodec/avcodec.h" +#include "libavformat/avformat.h" + +#define AUDIO_BLOCK_SIZE 4096 + +typedef struct { + int fd; + int sample_rate; + int channels; + int frame_size; /* in bytes ! */ + enum CodecID 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 = open(audio_device, O_WRONLY); + else + audio_fd = 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) + 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(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 = 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(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->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(s1, 0, s1->filename); + if (ret < 0) { + av_free(st); + return AVERROR(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(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(EIO); + 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(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; + } + /* 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_DEMUXER +AVInputFormat oss_demuxer = { + "oss", + NULL_IF_CONFIG_SMALL("Open Sound System capture"), + sizeof(AudioData), + NULL, + audio_read_header, + audio_read_packet, + audio_read_close, + .flags = AVFMT_NOFILE, +}; +#endif + +#if CONFIG_OSS_MUXER +AVOutputFormat oss_muxer = { + "oss", + NULL_IF_CONFIG_SMALL("Open Sound System playback"), + "", + "", + 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 -- cgit v1.2.3