From 8d26c193fb42d08602ac93ece039d4718d029adc Mon Sep 17 00:00:00 2001
From: Diego Biurrun <diego@biurrun.de>
Date: Fri, 27 Mar 2015 12:40:23 +0100
Subject: avdevice: Apply a more consistent file naming scheme

---
 libavdevice/oss.c | 139 ++++++++++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 139 insertions(+)
 create mode 100644 libavdevice/oss.c

(limited to 'libavdevice/oss.c')

diff --git a/libavdevice/oss.c b/libavdevice/oss.c
new file mode 100644
index 0000000000..eb8d454422
--- /dev/null
+++ b/libavdevice/oss.c
@@ -0,0 +1,139 @@
+/*
+ * Linux audio play and 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"
+
+#include <string.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 "libavutil/log.h"
+
+#include "libavcodec/avcodec.h"
+
+#include "libavformat/avformat.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");
+    char errbuff[128];
+
+    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, 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 = OSS_AUDIO_BLOCK_SIZE;
+
+#define CHECK_IOCTL_ERROR(event)                                              \
+    if (err < 0) {                                                            \
+        av_strerror(AVERROR(errno), errbuff, sizeof(errbuff));                \
+        av_log(s1, AV_LOG_ERROR, #event ": %s\n", errbuff);                   \
+        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. */
+    (void) 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 = 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;
+}
-- 
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