Merge commit '8d26c193fb42d08602ac93ece039d4718d029adc'

* commit '8d26c193fb42d08602ac93ece039d4718d029adc':
  avdevice: Apply a more consistent file naming scheme

Conflicts:
	libavdevice/Makefile
	libavdevice/alsa.h
	libavdevice/alsa_dec.c
	libavdevice/alsa_enc.c
	libavdevice/sndio_enc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 144 insertions, 0 deletions

diff --git a/libavdevice/oss.c b/libavdevice/oss.c
new file mode 100644
index 0000000000..d74112825b
--- /dev/null
+++ b/libavdevice/oss.c
@@ -0,0 +1,144 @@
+/*
+ * 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 <string.h>
+
+#if HAVE_SOUNDCARD_H
+#include <soundcard.h>
+#else
+#include <sys/soundcard.h>
+#endif
+
+#if HAVE_UNISTD_H
+#include <unistd.h>
+#endif
+#include <fcntl.h>
+#include <sys/ioctl.h>
+
+#include "libavutil/log.h"
+
+#include "libavcodec/avcodec.h"
+#include "avdevice.h"
+
+#include "oss.h"
+
+int ff_oss_audio_open(AVFormatContext *s1, int is_output,
+                      const char *audio_device)
+{
+    OSSAudioData *s = s1->priv_data;
+    int audio_fd;
+    int tmp, err;
+    char *flip = getenv("AUDIO_FLIP_LEFT");
+
+    if (is_output)
+        audio_fd = avpriv_open(audio_device, O_WRONLY);
+    else
+        audio_fd = avpriv_open(audio_device, O_RDONLY);
+    if (audio_fd < 0) {
+        av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, av_err2str(AVERROR(errno)));
+        return AVERROR(EIO);
+    }
+
+    if (flip && *flip == '1') {
+        s->flip_left = 1;
+    }
+
+    /* non blocking mode */
+    if (!is_output) {
+        if (fcntl(audio_fd, F_SETFL, O_NONBLOCK) < 0) {
+            av_log(s1, AV_LOG_WARNING, "%s: Could not enable non block mode (%s)\n", audio_device, av_err2str(AVERROR(errno)));
+        }
+    }
+
+    s->frame_size = OSS_AUDIO_BLOCK_SIZE;
+
+#define CHECK_IOCTL_ERROR(event)                                              \
+    if (err < 0) {                                                            \
+        av_log(s1, AV_LOG_ERROR, #event ": %s\n", av_err2str(AVERROR(errno)));\
+        goto fail;                                                            \
+    }
+
+    /* select format : favour native format
+     * We don't CHECK_IOCTL_ERROR here because even if failed OSS still may be
+     * usable. If OSS is not usable the SNDCTL_DSP_SETFMTS later is going to
+     * fail anyway. */
+    err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
+    if (err < 0) {
+        av_log(s1, AV_LOG_WARNING, "SNDCTL_DSP_GETFMTS: %s\n", av_err2str(AVERROR(errno)));
+    }
+
+#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 = AV_CODEC_ID_PCM_S16LE;
+        break;
+    case AFMT_S16_BE:
+        s->codec_id = AV_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);
+    CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMTS)
+
+    tmp = (s->channels == 2);
+    err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
+    CHECK_IOCTL_ERROR(SNDCTL_DSP_STEREO)
+
+    tmp = s->sample_rate;
+    err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
+    CHECK_IOCTL_ERROR(SNDCTL_DSP_SPEED)
+    s->sample_rate = tmp; /* store real sample rate */
+    s->fd = audio_fd;
+
+    return 0;
+ fail:
+    close(audio_fd);
+    return AVERROR(EIO);
+#undef CHECK_IOCTL_ERROR
+}
+
+int ff_oss_audio_close(OSSAudioData *s)
+{
+    close(s->fd);
+    return 0;
+}