* Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
* Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * 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 FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <alsa/asoundlib.h>
#include "config.h"
#include "libavformat/avformat.h"
+#include "libavutil/log.h"
/* XXX: we make the assumption that the soundcard accepts this format */
/* XXX: find better solution with "preinit" method, needed also in
other formats */
-#if HAVE_BIGENDIAN
-#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16BE
-#else
-#define DEFAULT_CODEC_ID CODEC_ID_PCM_S16LE
-#endif
+#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
-typedef struct {
+#define ALSA_BUFFER_SIZE_MAX 32768
+
+typedef struct AlsaData {
+ AVClass *class;
snd_pcm_t *h;
int frame_size; ///< preferred size for reads and writes
int period_size; ///< bytes per sample * channels
+ int sample_rate; ///< sample rate set by user
+ int channels; ///< number of channels set by user
+ void (*reorder_func)(const void *, void *, int);
+ void *reorder_buf;
+ int reorder_buf_size; ///< in frames
} AlsaData;
/**
* @param sample_rate in: requested sample rate;
* out: actually selected sample rate
* @param channels number of channels
- * @param codec_id in: requested CodecID or CODEC_ID_NONE;
- * out: actually selected CodecID, changed only if
- * CODEC_ID_NONE was requested
+ * @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 CodecID *codec_id);
+ int channels, enum AVCodecID *codec_id);
/**
* Close the ALSA PCM.
*/
int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
+int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
+
#endif /* AVDEVICE_ALSA_AUDIO_H */