[FFmpeg-cvslog] r16707 - in trunk/libavdevice: Makefile audio.c oss_audio.c
benoit
subversion
Wed Jan 21 09:43:38 CET 2009
Author: benoit
Date: Wed Jan 21 09:43:38 2009
New Revision: 16707
Log:
Rename audio.c to oss_audio.c in libavdevice.
Added:
trunk/libavdevice/oss_audio.c (props changed)
- copied unchanged from r16706, trunk/libavdevice/audio.c
Deleted:
trunk/libavdevice/audio.c
Modified:
trunk/libavdevice/Makefile
Modified: trunk/libavdevice/Makefile
==============================================================================
--- trunk/libavdevice/Makefile Wed Jan 21 01:36:34 2009 (r16706)
+++ trunk/libavdevice/Makefile Wed Jan 21 09:43:38 2009 (r16707)
@@ -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
Copied: trunk/libavdevice/oss_audio.c (from r16706, trunk/libavdevice/audio.c)
==============================================================================
--- /dev/null 00:00:00 1970 (empty, because file is newly added)
+++ trunk/libavdevice/oss_audio.c Wed Jan 21 09:43:38 2009 (r16707, copy of r16706, trunk/libavdevice/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 <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 "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
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