Update audio_decoder

source/modules/daz-plugins/audio_decoder/ad_ffmpeg.c

@@ -23,7 +23,7 @@
 #include <unistd.h>
 #include <math.h>
 
-#include "audio_decoder/ad_plugin.h"
+#include "ad_plugin.h"
 
 #ifdef HAVE_FFMPEG
 
@@ -215,16 +215,24 @@ static ssize_t ad_read_ffmpeg(void *sf, float* d, size_t len) {
 
       /* decode all chunks in packet */
       int data_size= AVCODEC_MAX_AUDIO_FRAME_SIZE;
-#if 0 // TODO  ffcompat.h -- this works but is not optimal (channels may not be planar/interleaved)
-      AVFrame avf; // TODO statically allocate
+
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 0, 0)
+      /* This works but is not optimal (channels may not be planar/interleaved) */
+      AVFrame avf; // TODO statically allocate as poart of priv->..
       memset(&avf, 0, sizeof(AVFrame)); // not sure if that is needed
       int got_frame = 0;
-ret = avcodec_decode_audio4(priv->codecContext, &avf, &got_frame, &priv->packet);
+      ret = avcodec_decode_audio4(priv->codecContext, &avf, &got_frame, &priv->packet);
       data_size = avf.linesize[0];
       memcpy(priv->m_tmpBuffer, avf.data[0], avf.linesize[0] * sizeof(uint8_t));
-#else // this was deprecated in LIBAVCODEC_VERSION_MAJOR 53
-      ret = avcodec_decode_audio3(priv->codecContext, 
+#elif LIBAVUTIL_VERSION_INT > AV_VERSION_INT(49, 15, 0) && LIBAVCODEC_VERSION_INT > AV_VERSION_INT(52, 20, 1) // ??
+      // this was deprecated in LIBAVCODEC_VERSION_MAJOR 53
+      ret = avcodec_decode_audio3(priv->codecContext,
           priv->m_tmpBuffer, &data_size, &priv->packet);
+#else
+			int len = priv->packet.size;
+			uint8_t *ptr = priv->packet.data;
+      ret = avcodec_decode_audio2(priv->codecContext,
+          priv->m_tmpBuffer, &data_size, ptr, len);
 #endif
 
       if (ret < 0 || ret > priv->pkt_len) {

source/modules/daz-plugins/audio_decoder/ad_plugin.c

@@ -16,6 +16,7 @@
    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
 
 */
+
 #include <stdio.h>
 #include <stdlib.h>
 #include <stdarg.h>
@@ -23,7 +24,7 @@
 #include <unistd.h>
 #include <math.h>
 
-#include "audio_decoder/ad_plugin.h"
+#include "ad_plugin.h"
 
 int ad_debug_level = 0;
 
@@ -118,7 +119,7 @@ ssize_t ad_read_mono_dbl(void *sf, struct adinfo *nfo, double* d, size_t len){
 		buf = (float*) realloc((void*)buf, bufsiz * sizeof(float));
 	}
 
-	len = (size_t)ad_read(sf, buf, bufsiz);
+	len = ad_read(sf, buf, bufsiz);
 
 	for (f=0;f< (len/chn);f++) {
 		double val=0.0;
@@ -127,7 +128,7 @@ ssize_t ad_read_mono_dbl(void *sf, struct adinfo *nfo, double* d, size_t len){
 		}
 		d[f]= val/chn;
 	}
-	return (ssize_t)len/chn;
+	return len/chn;
 }
 
 

source/modules/daz-plugins/audio_decoder/ad_soundfile.c

@@ -16,6 +16,7 @@
    Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
 
 */
+
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
@@ -24,7 +25,7 @@
 #include <math.h>
 #include <sndfile.h>
 
-#include "audio_decoder/ad_plugin.h"
+#include "ad_plugin.h"
 
 /* internal abstraction */
 
@@ -52,12 +53,12 @@ static int ad_info_sndfile(void *sf, struct adinfo *nfo) {
 	sndfile_audio_decoder *priv = (sndfile_audio_decoder*) sf;
 	if (!priv) return -1;
 	if (nfo) {
-		nfo->channels    = (unsigned int)priv->sfinfo.channels;
+		nfo->channels    = priv->sfinfo.channels;
 		nfo->frames      = priv->sfinfo.frames;
-		nfo->sample_rate = (unsigned int)priv->sfinfo.samplerate;
+		nfo->sample_rate = priv->sfinfo.samplerate;
 		nfo->length      = priv->sfinfo.samplerate ? (priv->sfinfo.frames * 1000) / priv->sfinfo.samplerate : 0;
 		nfo->bit_depth   = parse_bit_depth(priv->sfinfo.format);
-		nfo->bit_rate    = nfo->bit_depth * priv->sfinfo.channels * priv->sfinfo.samplerate;
+		nfo->bit_rate    = nfo->bit_depth * nfo->channels * nfo->sample_rate;
 		nfo->meta_data   = NULL;
 	}
 	return 0;
@@ -98,7 +99,7 @@ static int64_t ad_seek_sndfile(void *sf, int64_t pos) {
 static ssize_t ad_read_sndfile(void *sf, float* d, size_t len) {
 	sndfile_audio_decoder *priv = (sndfile_audio_decoder*) sf;
 	if (!priv) return -1;
-	return sf_read_float (priv->sffile, d, (sf_count_t)len);
+	return sf_read_float (priv->sffile, d, len);
 }
 
 static int ad_eval_sndfile(const char *f) { 

source/modules/daz-plugins/audio_decoder/ffcompat.h

@@ -1,6 +1,6 @@
 /* ffmpeg compatibility wrappers
  *
- * Copyright 2012 Robin Gareus <robin@gareus.org>
+ * Copyright 2012,2013 Robin Gareus <robin@gareus.org>
  *
  * Permission is hereby granted, free of charge, to any person obtaining a copy
  * of this software and associated documentation files (the "Software"), to deal
@@ -26,15 +26,27 @@
  * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
  * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
+#ifndef FFCOMPAT_H
+#define FFCOMPAT_H
+
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
+#include <libavutil/avutil.h>
+
+#ifndef AVCODEC_MAX_AUDIO_FRAME_SIZE
+#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000
+#endif
+
+#if LIBAVUTIL_VERSION_INT < AV_VERSION_INT(50, 0, 0)
+#define AVMEDIA_TYPE_AUDIO CODEC_TYPE_AUDIO
+#endif
 
 #if LIBAVFORMAT_VERSION_INT < AV_VERSION_INT(53, 2, 0)
 static inline int avformat_open_input(AVFormatContext **ps, const char *filename, void *fmt, void **options)
 {
 	return av_open_input_file(ps, filename, NULL, 0, NULL);
 }
-#endif
+#endif /* avformat < 53.2.0 */
 
 #if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(53, 5, 0)
 static inline AVCodecContext *
@@ -79,3 +91,5 @@ avformat_close_input(AVFormatContext **s)
 }
 
 #endif /* < 53.5.0 */
+
+#endif /* FFCOMPAT_H */