cavsdec: Make sure a sequence header has been decoded before decoding pictures

[ffmpeg.git] / libavdevice / alsa-audio.h

diff --git a/libavdevice/alsa-audio.h b/libavdevice/alsa-audio.h
index 9493831..26eaee6 100644 (file)
--- a/libavdevice/alsa-audio.h
+++ b/libavdevice/alsa-audio.h
@@ -3,20 +3,20 @@
  * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
  * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
  *
  * 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.
  *
  * 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
  * 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
  */
 
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 


@@ -33,16 +33,25 @@
 #include <alsa/asoundlib.h>
 #include "config.h"
 #include "libavformat/avformat.h"
 #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 */
 
 /* 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(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
+#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
     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;
 
 /**
 } AlsaData;
 
 /**


@@ -53,15 +62,15 @@ typedef struct {
  * @param sample_rate in: requested sample rate;
  *                    out: actually selected sample rate
  * @param channels number of channels
  * @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,
  *
  * @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.
 
 /**
  * Close the ALSA PCM.


@@ -82,4 +91,6 @@ int ff_alsa_close(AVFormatContext *s1);
  */
 int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
 
  */
 int ff_alsa_xrun_recover(AVFormatContext *s1, int err);
 

+int ff_alsa_extend_reorder_buf(AlsaData *s, int size);
+

 #endif /* AVDEVICE_ALSA_AUDIO_H */
 #endif /* AVDEVICE_ALSA_AUDIO_H */