From d6e1d37100af568211f28ec0bcf7958a3a2a299e Mon Sep 17 00:00:00 2001 From: Nidhi Makhijani Date: Fri, 18 Jul 2014 16:13:15 +0530 Subject: oss_audio: Split muxer and demuxer Signed-off-by: Diego Biurrun --- libavdevice/Makefile | 4 +- libavdevice/oss_audio.c | 211 +++----------------------------------------- libavdevice/oss_audio.h | 45 ++++++++++ libavdevice/oss_audio_dec.c | 146 ++++++++++++++++++++++++++++++ libavdevice/oss_audio_enc.c | 108 +++++++++++++++++++++++ 5 files changed, 312 insertions(+), 202 deletions(-) create mode 100644 libavdevice/oss_audio.h create mode 100644 libavdevice/oss_audio_dec.c create mode 100644 libavdevice/oss_audio_enc.c (limited to 'libavdevice') diff --git a/libavdevice/Makefile b/libavdevice/Makefile index 2eb2f8e542..25e126c6b6 100644 --- a/libavdevice/Makefile +++ b/libavdevice/Makefile @@ -15,8 +15,8 @@ OBJS-$(CONFIG_BKTR_INDEV) += bktr.o OBJS-$(CONFIG_DV1394_INDEV) += dv1394.o OBJS-$(CONFIG_FBDEV_INDEV) += fbdev.o OBJS-$(CONFIG_JACK_INDEV) += jack_audio.o timefilter.o -OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o -OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o +OBJS-$(CONFIG_OSS_INDEV) += oss_audio.o oss_audio_dec.o +OBJS-$(CONFIG_OSS_OUTDEV) += oss_audio.o oss_audio_enc.o OBJS-$(CONFIG_PULSE_INDEV) += pulse.o OBJS-$(CONFIG_SNDIO_INDEV) += sndio_common.o sndio_dec.o OBJS-$(CONFIG_SNDIO_OUTDEV) += sndio_common.o sndio_enc.o diff --git a/libavdevice/oss_audio.c b/libavdevice/oss_audio.c index 95f73fbd8e..ad52d78188 100644 --- a/libavdevice/oss_audio.c +++ b/libavdevice/oss_audio.c @@ -20,45 +20,31 @@ */ #include "config.h" -#include -#include -#include + #include -#include + #if HAVE_SOUNDCARD_H #include #else #include #endif + #include #include #include -#include "libavutil/internal.h" #include "libavutil/log.h" -#include "libavutil/opt.h" -#include "libavutil/time.h" + #include "libavcodec/avcodec.h" -#include "libavformat/avformat.h" -#include "libavformat/internal.h" -#define AUDIO_BLOCK_SIZE 4096 +#include "libavformat/avformat.h" -typedef struct AudioData { - 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; +#include "oss_audio.h" -static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device) +int ff_oss_audio_open(AVFormatContext *s1, int is_output, + const char *audio_device) { - AudioData *s = s1->priv_data; + OSSAudioData *s = s1->priv_data; int audio_fd; int tmp, err; char *flip = getenv("AUDIO_FLIP_LEFT"); @@ -80,7 +66,7 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi if (!is_output) fcntl(audio_fd, F_SETFL, O_NONBLOCK); - s->frame_size = AUDIO_BLOCK_SIZE; + s->frame_size = OSS_AUDIO_BLOCK_SIZE; /* select format : favour native format */ err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp); @@ -143,183 +129,8 @@ static int audio_open(AVFormatContext *s1, int is_output, const char *audio_devi return AVERROR(EIO); } -static int audio_close(AudioData *s) +int ff_oss_audio_close(OSSAudioData *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 diff --git a/libavdevice/oss_audio.h b/libavdevice/oss_audio.h new file mode 100644 index 0000000000..87ac4adfd5 --- /dev/null +++ b/libavdevice/oss_audio.h @@ -0,0 +1,45 @@ +/* + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVDEVICE_OSS_AUDIO_H +#define AVDEVICE_OSS_AUDIO_H + +#include "libavcodec/avcodec.h" + +#include "libavformat/avformat.h" + +#define OSS_AUDIO_BLOCK_SIZE 4096 + +typedef struct OSSAudioData { + 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[OSS_AUDIO_BLOCK_SIZE]; + int buffer_ptr; +} OSSAudioData; + +int ff_oss_audio_open(AVFormatContext *s1, int is_output, + const char *audio_device); + +int ff_oss_audio_close(OSSAudioData *s); + +#endif /* AVDEVICE_OSS_AUDIO_H */ diff --git a/libavdevice/oss_audio_dec.c b/libavdevice/oss_audio_dec.c new file mode 100644 index 0000000000..601d91c31f --- /dev/null +++ b/libavdevice/oss_audio_dec.c @@ -0,0 +1,146 @@ +/* + * Linux audio play interface + * Copyright (c) 2000, 2001 Fabrice Bellard + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "config.h" + +#include + +#if HAVE_SOUNDCARD_H +#include +#else +#include +#endif + +#include +#include +#include + +#include "libavutil/internal.h" +#include "libavutil/opt.h" +#include "libavutil/time.h" + +#include "libavcodec/avcodec.h" + +#include "libavformat/avformat.h" +#include "libavformat/internal.h" + +#include "oss_audio.h" + +static int audio_read_header(AVFormatContext *s1) +{ + OSSAudioData *s = s1->priv_data; + AVStream *st; + int ret; + + st = avformat_new_stream(s1, NULL); + if (!st) { + return AVERROR(ENOMEM); + } + + ret = ff_oss_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) +{ + OSSAudioData *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) +{ + OSSAudioData *s = s1->priv_data; + + ff_oss_audio_close(s); + return 0; +} + +static const AVOption options[] = { + { "sample_rate", "", offsetof(OSSAudioData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM }, + { "channels", "", offsetof(OSSAudioData, 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(OSSAudioData), + .read_header = audio_read_header, + .read_packet = audio_read_packet, + .read_close = audio_read_close, + .flags = AVFMT_NOFILE, + .priv_class = &oss_demuxer_class, +}; diff --git a/libavdevice/oss_audio_enc.c b/libavdevice/oss_audio_enc.c new file mode 100644 index 0000000000..688982a00f --- /dev/null +++ b/libavdevice/oss_audio_enc.c @@ -0,0 +1,108 @@ +/* + * Linux audio grab interface + * Copyright (c) 2000, 2001 Fabrice Bellard + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "config.h" + +#if HAVE_SOUNDCARD_H +#include +#else +#include +#endif + +#include +#include +#include + +#include "libavutil/internal.h" + +#include "libavcodec/avcodec.h" + +#include "libavformat/avformat.h" +#include "libavformat/internal.h" + +#include "oss_audio.h" + +static int audio_write_header(AVFormatContext *s1) +{ + OSSAudioData *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 = ff_oss_audio_open(s1, 1, s1->filename); + if (ret < 0) { + return AVERROR(EIO); + } else { + return 0; + } +} + +static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) +{ + OSSAudioData *s = s1->priv_data; + int len, ret; + int size= pkt->size; + uint8_t *buf= pkt->data; + + while (size > 0) { + len = FFMIN(OSS_AUDIO_BLOCK_SIZE - s->buffer_ptr, size); + memcpy(s->buffer + s->buffer_ptr, buf, len); + s->buffer_ptr += len; + if (s->buffer_ptr >= OSS_AUDIO_BLOCK_SIZE) { + for(;;) { + ret = write(s->fd, s->buffer, OSS_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) +{ + OSSAudioData *s = s1->priv_data; + + ff_oss_audio_close(s); + return 0; +} + +AVOutputFormat ff_oss_muxer = { + .name = "oss", + .long_name = NULL_IF_CONFIG_SMALL("OSS (Open Sound System) playback"), + .priv_data_size = sizeof(OSSAudioData), + /* 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, +}; -- cgit v1.2.3