libavdevice/oss_audio.c - third_party/ffmpeg - Git at Google

 /*
  * 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 <stdlib.h>
 #include <stdio.h>
 #include <stdint.h>
 #include <string.h>
 #include <errno.h>
 #if HAVE_SOUNDCARD_H
 #include <soundcard.h>
 #else
 #include <sys/soundcard.h>
 #endif
 #include <unistd.h>
 #include <fcntl.h>
 #include <sys/ioctl.h>
 #include <sys/time.h>
 #include <sys/select.h>

 #include "libavutil/log.h"
 #include "libavutil/opt.h"
 #include "libavcodec/avcodec.h"
 #include "avdevice.h"
 #include "libavformat/internal.h"

 #define AUDIO_BLOCK_SIZE 4096

 typedef struct {
     AVClass *class;
     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;

     /* select format : favour native format */
     err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);

 #if HAVE_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 = 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, AVFormatParameters *ap)
 {
     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, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
     { "channels",    "", offsetof(AudioData, channels),    AV_OPT_TYPE_INT, {.dbl = 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("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("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(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
     .video_codec    = CODEC_ID_NONE,
     .write_header   = audio_write_header,
     .write_packet   = audio_write_packet,
     .write_trailer  = audio_write_trailer,
     .flags          = AVFMT_NOFILE,
 };
 #endif
	/*
	* 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 <stdlib.h>
	#include <stdio.h>
	#include <stdint.h>
	#include <string.h>
	#include <errno.h>
	#if HAVE_SOUNDCARD_H
	#include <soundcard.h>
	#else
	#include <sys/soundcard.h>
	#endif
	#include <unistd.h>
	#include <fcntl.h>
	#include <sys/ioctl.h>
	#include <sys/time.h>
	#include <sys/select.h>

	#include "libavutil/log.h"
	#include "libavutil/opt.h"
	#include "libavcodec/avcodec.h"
	#include "avdevice.h"
	#include "libavformat/internal.h"

	#define AUDIO_BLOCK_SIZE 4096

	typedef struct {
	AVClass *class;
	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;

	/* select format : favour native format */
	err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);

	#if HAVE_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 = 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, AVFormatParameters ap)
	{
	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, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
	{ "channels", "", offsetof(AudioData, channels), AV_OPT_TYPE_INT, {.dbl = 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("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("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(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
	.video_codec = CODEC_ID_NONE,
	.write_header = audio_write_header,
	.write_packet = audio_write_packet,
	.write_trailer = audio_write_trailer,
	.flags = AVFMT_NOFILE,
	};
	#endif