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Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcodecmap.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegcodecmap.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcodecmap.c
@@ -1925,6 +1925,10 @@ gst_ffmpeg_smpfmt_to_caps (enum AVSample
   gboolean integer = TRUE;
   gboolean signedness = FALSE;
 
+#if LIBAVUTIL_VERSION_INT > AV_VERSION_INT(51,46,0)
+  sample_fmt = av_get_packed_sample_fmt (sample_fmt);
+#endif
+
   switch (sample_fmt) {
     case AV_SAMPLE_FMT_S16:
       signedness = TRUE;
@@ -2009,7 +2013,7 @@ gst_ffmpeg_codectype_to_audio_caps (AVCo
 
     ctx.channels = -1;
     caps = gst_caps_new_empty ();
-    for (i = 0; i <= AV_SAMPLE_FMT_DBL; i++) {
+    for (i = 0; i < AV_SAMPLE_FMT_NB; i++) {
       temp =
           gst_ffmpeg_smpfmt_to_caps (i, encode ? &ctx : NULL, codec_id, encode);
       if (temp != NULL) {
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegutils.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegutils.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegutils.c
@@ -47,6 +47,9 @@ gint
 av_smp_format_depth (enum AVSampleFormat smp_fmt)
 {
   gint depth = -1;
+#if LIBAVUTIL_VERSION_INT > AV_VERSION_INT(51,46,0)
+  smp_fmt = av_get_packed_sample_fmt (smp_fmt);
+#endif
   switch (smp_fmt) {
     case AV_SAMPLE_FMT_U8:
       depth = 1;
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegdec.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegdec.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegdec.c
@@ -2044,16 +2044,49 @@ out_of_segment:
   }
 }
 
+static void copy_samples_planar(unsigned bps, 
+    unsigned nb_samples,
+    unsigned nb_channels,
+    unsigned char *dst,
+    unsigned char **src)
+{
+  unsigned s, c, o = 0;
+
+  for (s = 0; s < nb_samples; s++) {
+      for (c = 0; c < nb_channels; c++) {
+          memcpy(dst, src[c] + o, bps);
+	   dst += bps;
+      }
+      o += bps;
+  }
+}
+
+static int copy_samples(AVCodecContext *avc, AVFrame *frame,
+	unsigned char *buf, int max_size)
+{
+	int channels = avc->channels;
+	int sample_size = av_get_bytes_per_sample(avc->sample_fmt);
+	int size = channels * sample_size * frame->nb_samples;
+	if (size > max_size) {
+		return -1;
+	}
+	if (av_sample_fmt_is_planar(avc->sample_fmt))
+		copy_samples_planar(sample_size, frame->nb_samples, channels, buf, frame->extended_data);
+	else memcpy(buf, frame->data[0], size);
+	return size;
+}
+
 static gint
 gst_ffmpegdec_audio_frame (GstFFMpegDec * ffmpegdec,
     AVCodec * in_plugin, guint8 * data, guint size,
     const GstTSInfo * dec_info, GstBuffer ** outbuf, GstFlowReturn * ret)
 {
-  gint len = -1;
+  gint len = -1, got_frame;
   gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
   GstClockTime out_timestamp, out_duration;
   gint64 out_offset;
   AVPacket packet;
+  AVFrame *frame;
 
   GST_DEBUG_OBJECT (ffmpegdec,
       "size:%d, offset:%" G_GINT64_FORMAT ", ts:%" GST_TIME_FORMAT ", dur:%"
@@ -2061,15 +2094,28 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
       dec_info->offset, GST_TIME_ARGS (dec_info->timestamp),
       GST_TIME_ARGS (dec_info->duration), GST_TIME_ARGS (ffmpegdec->next_out));
 
+  frame = avcodec_alloc_frame();
+  if (!frame) {
+      *outbuf = NULL;
+      len = -1;
+      goto beach;
+  }
+
   *outbuf =
       new_aligned_buffer (AVCODEC_MAX_AUDIO_FRAME_SIZE,
       GST_PAD_CAPS (ffmpegdec->srcpad));
 
   gst_avpacket_init (&packet, data, size);
-  len = avcodec_decode_audio3 (ffmpegdec->context,
-      (int16_t *) GST_BUFFER_DATA (*outbuf), &have_data, &packet);
+  len = avcodec_decode_audio4 (ffmpegdec->context, frame, &got_frame, &packet);
   GST_DEBUG_OBJECT (ffmpegdec,
-      "Decode audio: len=%d, have_data=%d", len, have_data);
+      "Decode audio: ret=%d, got_frame=%d", len, got_frame);
+  if (!got_frame) {
+      gst_buffer_unref (*outbuf);
+      *outbuf = NULL;
+      len = -1;
+      goto beach;
+  }
+  if (len >= 0) have_data = copy_samples(ffmpegdec->context, frame, GST_BUFFER_DATA (*outbuf), AVCODEC_MAX_AUDIO_FRAME_SIZE);
 
   if (len >= 0 && have_data > 0) {
     GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
@@ -2145,6 +2191,7 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
   }
 
 beach:
+  av_free(frame);
   GST_DEBUG_OBJECT (ffmpegdec, "return flow %d, out %p, len %d",
       *ret, *outbuf, len);
   return len;
Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegenc.c
===================================================================
--- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegenc.c
+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegenc.c
@@ -843,12 +843,30 @@ gst_ffmpegenc_chain_video (GstPad * pad,
   return gst_pad_push (ffmpegenc->srcpad, outbuf);
 }
 
+static void copy_samples_to_planar(unsigned bps, 
+    unsigned nb_samples,
+    unsigned nb_channels,
+    unsigned char *dst,
+    unsigned char *src)
+{
+  unsigned s, c, o = 0;
+
+  for (s = 0; s < nb_samples; s++) {
+      for (c = 0; c < nb_channels; c++) {
+          memcpy(dst + nb_samples * c, src + o, bps);
+      	   o += bps;
+      }
+      dst += bps;
+  }
+}
+
 static GstFlowReturn
 gst_ffmpegenc_encode_audio (GstFFMpegEnc * ffmpegenc, guint8 * audio_in,
     guint in_size, guint max_size, GstClockTime timestamp,
     GstClockTime duration, gboolean discont)
 {
   GstBuffer *outbuf;
+  GstBuffer *inbuf2;
   AVCodecContext *ctx;
   guint8 *audio_out;
   gint res;
@@ -864,7 +882,18 @@ gst_ffmpegenc_encode_audio (GstFFMpegEnc
   if (ffmpegenc->buffer_size != max_size)
     ffmpegenc->buffer_size = max_size;
 
+  if (av_sample_fmt_is_planar(ctx->sample_fmt)) {
+  	guint8 * audio_in2;
+	inbuf2 = gst_buffer_new_and_alloc (in_size + FF_MIN_BUFFER_SIZE);
+	audio_in2 = GST_BUFFER_DATA (inbuf2);
+	copy_samples_to_planar(av_get_bytes_per_sample(ctx->sample_fmt), in_size / (av_get_bytes_per_sample(ctx->sample_fmt) * ctx->channels),
+		ctx->channels, audio_in2, audio_in);
+	audio_in = audio_in2;
+  }
   res = avcodec_encode_audio (ctx, audio_out, max_size, (short *) audio_in);
+  if (av_sample_fmt_is_planar(ctx->sample_fmt)) {
+    gst_buffer_unref (inbuf2);
+  }
 
   if (res < 0) {
     GST_ERROR_OBJECT (ffmpegenc, "Failed to encode buffer: %d", res);