ffmpeg-2.1.1: move directory

git-svn-id: svn://kolibrios.org@6148 a494cfbc-eb01-0410-851d-a64ba20cac60
2016-02-05 22:14:10 +00:00
parent a4b787f4b8
commit ecf3e862ea
4011 changed files with 1868 additions and 4 deletions
--- a/contrib/sdk/sources/ffmpeg/ffmpeg-2.1/libavdevice/alsa-audio.h
+++ b/contrib/sdk/sources/ffmpeg/ffmpeg-2.1/libavdevice/alsa-audio.h
@@ -0,0 +1,101 @@
+/*
+ * ALSA input and output
+ * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
+ * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * ALSA input and output: definitions and structures
+ * @author Luca Abeni ( lucabe72 email it )
+ * @author Benoit Fouet ( benoit fouet free fr )
+ */
+
+#ifndef AVDEVICE_ALSA_AUDIO_H
+#define AVDEVICE_ALSA_AUDIO_H
+
+#include <alsa/asoundlib.h>
+#include "config.h"
+#include "libavutil/log.h"
+#include "timefilter.h"
+#include "avdevice.h"
+
+/* XXX: we make the assumption that the soundcard accepts this format */
+/* XXX: find better solution with "preinit" method, needed also in
+        other formats */
+#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
+
+typedef void (*ff_reorder_func)(const void *, void *, int);
+
+#define ALSA_BUFFER_SIZE_MAX 65536
+
+typedef struct AlsaData {
+    AVClass *class;
+    snd_pcm_t *h;
+    int frame_size;  ///< bytes per sample * channels
+    int period_size; ///< preferred size for reads and writes, in frames
+    int sample_rate; ///< sample rate set by user
+    int channels;    ///< number of channels set by user
+    int last_period;
+    TimeFilter *timefilter;
+    void (*reorder_func)(const void *, void *, int);
+    void *reorder_buf;
+    int reorder_buf_size; ///< in frames
+} AlsaData;
+
+/**
+ * Open an ALSA PCM.
+ *
+ * @param s media file handle
+ * @param mode either SND_PCM_STREAM_CAPTURE or SND_PCM_STREAM_PLAYBACK
+ * @param sample_rate in: requested sample rate;
+ *                    out: actually selected sample rate
+ * @param channels number of channels
+ * @param codec_id in: requested AVCodecID or AV_CODEC_ID_NONE;
+ *                 out: actually selected AVCodecID, changed only if
+ *                 AV_CODEC_ID_NONE was requested
+ *
+ * @return 0 if OK, AVERROR_xxx on error
+ */
+int ff_alsa_open(AVFormatContext *s, snd_pcm_stream_t mode,
+                 unsigned int *sample_rate,
+                 int channels, enum AVCodecID *codec_id);
+
+/**
+ * Close the ALSA PCM.
+ *
+ * @param s1 media file handle
+ *
+ * @return 0
+ */
+int ff_alsa_close(AVFormatContext *s1);
+
+/**
+ * Try to recover from ALSA buffer underrun.
+ *
+ * @param s1 media file handle
+ * @param err error code reported by the previous ALSA call
+ *
+ * @return 0 if OK, AVERROR_xxx on error
+ */
+int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
+
+int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
+
+#endif /* AVDEVICE_ALSA_AUDIO_H */