media-plugins/gst-plugins-ffmpeg: Remove last-rited pkg, #594878

12 files changed, 0 insertions, 607 deletions

diff --git a/media-plugins/gst-plugins-ffmpeg/Manifest b/media-plugins/gst-plugins-ffmpeg/Manifest
deleted file mode 100644
index 62028ead5984..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/Manifest
+++ /dev/null
@@ -1,2 +0,0 @@
-DIST gst-ffmpeg-0.10.13_p201211-libav-9-patches.tar.xz 14076 SHA256 5afa627ccbbedc1c80bfd419cb497c746aea0f27e0165a49c41947eba31774ee SHA512 cd12490600c0e56a85ec7436ec6152be8962c0e949ee9925ad64c114c60d7a72958f7c5ac98a3bdc6988bce707a2c40060e3a20b899ef2a95a30b4166bfb8145 WHIRLPOOL a7f17fd32112ba9df89ef80717946a50cc81e02e608c7cc15066987cd0e3556a1971ef5360ac50089d7c883aae5eb9d36ed3d7834fb7a075a816ceb16fa2b960
-DIST gst-ffmpeg-0.10.13_p201211.tar.xz 4285300 SHA256 cc15841b9972d1b60c9aaa3ffae14f3dbf44c0b08c14bb9319fb8956c89864c6 SHA512 9067aa92ef7613b08a51957d35d1de80897c28cb959522d9b523de6f482635dac3fe421015febb88ece624b7baee0216dd4b72e770bf92c05d8994f186d2f396 WHIRLPOOL 88ba2ce647ca821ef8f7aa22d64cf39397e30d8f34eadec475e6b627472c4baf6c7de6c60d4ca86527a67c710193dc00eb6534fe8bc92f73cc26db1f8e404a49
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch
deleted file mode 100644
index 3c115d7b45c5..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch
+++ /dev/null
@@ -1,71 +0,0 @@
---- ext/ffmpeg/gstffmpegaudioresample.c~	2015-05-29 11:38:03.639001637 +0200
-+++ ext/ffmpeg/gstffmpegaudioresample.c	2015-05-29 20:08:24.744107000 +0200
-@@ -24,6 +24,7 @@
- #include "config.h"
- #endif
- 
-+#include <libavresample/avresample.h>
- #ifdef HAVE_FFMPEG_UNINSTALLED
- #include <avcodec.h>
- #else
-@@ -37,6 +38,60 @@
- #include "gstffmpeg.h"
- #include "gstffmpegcodecmap.h"
- 
-+struct AudioData {
-+    const AVClass *class;               /**< AVClass for logging            */
-+    uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers        */
-+    uint8_t *buffer;                    /**< data buffer                    */
-+    unsigned int buffer_size;           /**< allocated buffer size          */
-+    int allocated_samples;              /**< number of samples the buffer can hold */
-+    int nb_samples;                     /**< current number of samples      */
-+    enum AVSampleFormat sample_fmt;     /**< sample format                  */
-+    int channels;                       /**< channel count                  */
-+    int allocated_channels;             /**< allocated channel count        */
-+    int is_planar;                      /**< sample format is planar        */
-+    int planes;                         /**< number of data planes          */
-+    int sample_size;                    /**< bytes per sample               */
-+    int stride;                         /**< sample byte offset within a plane */
-+    int read_only;                      /**< data is read-only              */
-+    int allow_realloc;                  /**< realloc is allowed             */
-+    int ptr_align;                      /**< minimum data pointer alignment */
-+    int samples_align;                  /**< allocated samples alignment    */
-+    const char *name;                   /**< name for debug logging         */
-+};
-+
-+typedef struct AudioData AudioData;
-+
-+struct ReSampleContext {
-+    AVAudioResampleContext *avr;
-+    AudioData *buffer;
-+    uint8_t *filter_bank;
-+    int filter_length;
-+    int ideal_dst_incr;
-+    int dst_incr;
-+    unsigned int index;
-+    int frac;
-+    int src_incr;
-+    int compensation_distance;
-+    int phase_shift;
-+    int phase_mask;
-+    int linear;
-+    enum AVResampleFilterType filter_type;
-+    int kaiser_beta;
-+    void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
-+    void (*resample_one)(struct ResampleContext *c, void *dst0,
-+                         int dst_index, const void *src0,
-+                         unsigned int index, int frac);
-+    void (*resample_nearest)(void *dst0, int dst_index,
-+                             const void *src0, unsigned int index);
-+    int padding_size;
-+    int initial_padding_filled;
-+    int initial_padding_samples;
-+    int final_padding_filled;
-+    int final_padding_samples;
-+};
-+
-+typedef struct ReSampleContext ReSampleContext;
-+
- typedef struct _GstFFMpegAudioResample
- {
-   GstBaseTransform element;
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-channel_layout.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-channel_layout.patch
deleted file mode 100644
index 9358f76153ca..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-channel_layout.patch
+++ /dev/null
@@ -1,29 +0,0 @@
-Include correct header with correct version. Fixes build with ffmpeg-0.10.
-
-
-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
-@@ -25,10 +25,20 @@
- #include <gst/gst.h>
- #ifdef HAVE_FFMPEG_UNINSTALLED
- #include <avcodec.h>
--#include <channel_layout.h>>
-+#include <avutil.h>
-+#if (LIBAVUTIL_VERSION_INT > AV_VERSION_INT(52,6,0) || (LIBAVUTIL_VERSION_MICRO < 100 && LIBAVUTIL_VERSION_INT > AV_VERSION_INT(52,2,0)))
-+#include <channel_layout.h>
-+#else
-+#include <audioconvert.h>
-+#endif
- #else
- #include <libavcodec/avcodec.h>
-+#include <libavutil/avutil.h>
-+#if (LIBAVUTIL_VERSION_INT > AV_VERSION_INT(52,6,0) || (LIBAVUTIL_VERSION_MICRO < 100 && LIBAVUTIL_VERSION_INT > AV_VERSION_INT(52,2,0)))
- #include <libavutil/channel_layout.h>
-+#else
-+#include <libavutil/audioconvert.h>
-+#endif
- #endif
- #include <string.h>
- 
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-coma.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-coma.patch
deleted file mode 100644
index 0426d00301c5..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-coma.patch
+++ /dev/null
@@ -1,16 +0,0 @@
-Some muxers come with a ',' in their name, replace it by '_' as in the other
-places.
-
-Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegmux.c
-===================================================================
---- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegmux.c
-+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegmux.c
-@@ -936,7 +936,7 @@ gst_ffmpegmux_register (GstPlugin * plug
-     p = type_name;
- 
-     while (*p) {
--      if (*p == '.')
-+      if (*p == '.' || *p == ',')
-         *p = '_';
-       p++;
-     }
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-ffmpeg2.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-ffmpeg2.patch
deleted file mode 100644
index 0ee39dac7f53..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-ffmpeg2.patch
+++ /dev/null
@@ -1,139 +0,0 @@
-Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcfg.c
-===================================================================
---- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegcfg.c
-+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegcfg.c
-@@ -171,13 +171,10 @@ gst_ffmpeg_idct_algo_get_type (void)
-       {FF_IDCT_INT, "JPEG reference Integer", "int"},
-       {FF_IDCT_SIMPLE, "Simple", "simple"},
-       {FF_IDCT_SIMPLEMMX, "Simple MMX", "simplemmx"},
--      {FF_IDCT_LIBMPEG2MMX, "LIBMPEG2MMX", "libmpeg2mmx"},
-       {FF_IDCT_ARM, "ARM", "arm"},
-       {FF_IDCT_ALTIVEC, "ALTIVEC", "altivec"},
-       {FF_IDCT_SH4, "SH4", "sh4"},
-       {FF_IDCT_SIMPLEARM, "SIMPLEARM", "simplearm"},
--      {FF_IDCT_H264, "H264", "h264"},
--      {FF_IDCT_VP3, "VP3", "vp3"},
-       {FF_IDCT_IPP, "IPP", "ipp"},
-       {FF_IDCT_XVIDMMX, "XVIDMMX", "xvidmmx"},
-       {0, NULL, NULL},
-@@ -274,9 +271,6 @@ gst_ffmpeg_flags_get_type (void)
-           "global-headers"},
-       {CODEC_FLAG_AC_PRED, "H263 Advanced Intra Coding / MPEG4 AC prediction",
-           "aic"},
--      {CODEC_FLAG_CBP_RD, "Rate Distoration Optimization for CBP", "cbp-rd"},
--      {CODEC_FLAG_QP_RD, "Rate Distoration Optimization for QP selection",
--          "qp-rd"},
-       {CODEC_FLAG_CLOSED_GOP, "Closed GOP", "closedgop"},
-       {0, NULL, NULL},
-     };
-@@ -580,18 +574,6 @@ gst_ffmpeg_cfg_init (void)
-       -100, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
-   gst_ffmpeg_add_pspec (pspec, max_key_interval, FALSE, mpeg, NULL);
- 
--  pspec = g_param_spec_int ("luma-elim-threshold",
--      "Luma Elimination Threshold",
--      "Luma Single Coefficient Elimination Threshold",
--      -99, 99, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
--  gst_ffmpeg_add_pspec (pspec, config.luma_elim_threshold, FALSE, mpeg, NULL);
--
--  pspec = g_param_spec_int ("chroma-elim-threshold",
--      "Chroma Elimination Threshold",
--      "Chroma Single Coefficient Elimination Threshold",
--      -99, 99, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
--  gst_ffmpeg_add_pspec (pspec, config.chroma_elim_threshold, FALSE, mpeg, NULL);
--
-   pspec = g_param_spec_float ("lumi-masking", "Luminance Masking",
-       "Luminance Masking", -1.0f, 1.0f, 0.0f,
-       G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
-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
-@@ -572,13 +572,11 @@ gst_ffmpeg_codecid_to_caps (enum AVCodec
-           break;
-       }
- 
--      /* FIXME: context->sub_id must be filled in during decoding */
-       caps =
-           gst_ff_vid_caps_new (context, codec_id, encode,
-           "video/x-pn-realvideo", "systemstream", G_TYPE_BOOLEAN, FALSE,
-           "rmversion", G_TYPE_INT, version, NULL);
-       if (context) {
--        gst_caps_set_simple (caps, "format", G_TYPE_INT, context->sub_id, NULL);
-         if (context->extradata_size >= 8) {
-           gst_caps_set_simple (caps,
-               "subformat", G_TYPE_INT, GST_READ_UINT32_BE (context->extradata),
-@@ -2559,18 +2557,6 @@ gst_ffmpeg_caps_with_codecid (enum AVCod
-     }
-       break;
- 
--    case AV_CODEC_ID_RV10:
--    case AV_CODEC_ID_RV20:
--    case AV_CODEC_ID_RV30:
--    case AV_CODEC_ID_RV40:
--    {
--      gint format;
--
--      if (gst_structure_get_int (str, "format", &format))
--        context->sub_id = format;
--
--      break;
--    }
-     case AV_CODEC_ID_COOK:
-     case AV_CODEC_ID_RA_288:
-     case AV_CODEC_ID_RA_144:
-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
-@@ -572,7 +572,6 @@ gst_ffmpegenc_setcaps (GstPad * pad, Gst
-   ffmpegenc->context->coder_type = 0;
-   ffmpegenc->context->context_model = 0;
-   ffmpegenc->context->scenechange_threshold = 0;
--  ffmpegenc->context->inter_threshold = 0;
- 
-   /* and last but not least the pass; CBR, 2-pass, etc */
-   ffmpegenc->context->flags |= ffmpegenc->pass;
-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
-@@ -2082,7 +2082,7 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
-     const GstTSInfo * dec_info, GstBuffer ** outbuf, GstFlowReturn * ret)
- {
-   gint len = -1, got_frame;
--  gint have_data = AVCODEC_MAX_AUDIO_FRAME_SIZE;
-+  gint have_data = 0;
-   GstClockTime out_timestamp, out_duration;
-   gint64 out_offset;
-   AVPacket packet;
-@@ -2101,21 +2101,22 @@ gst_ffmpegdec_audio_frame (GstFFMpegDec
-       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_audio4 (ffmpegdec->context, frame, &got_frame, &packet);
-   GST_DEBUG_OBJECT (ffmpegdec,
-       "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);
-+
-+  int obuf_size = av_samples_get_buffer_size(NULL, ffmpegdec->context->channels, frame->nb_samples, frame->format, 0);
-+
-+  *outbuf=
-+      new_aligned_buffer (obuf_size,
-+      GST_PAD_CAPS (ffmpegdec->srcpad));
-+
-+  if (len >= 0) have_data = copy_samples(ffmpegdec->context, frame, GST_BUFFER_DATA (*outbuf), obuf_size);
- 
-   if (len >= 0 && have_data > 0) {
-     GST_DEBUG_OBJECT (ffmpegdec, "Creating output buffer");
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-gstffmpegpipe_redef.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-gstffmpegpipe_redef.patch
deleted file mode 100644
index f76d72affde8..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-gstffmpegpipe_redef.patch
+++ /dev/null
@@ -1,16 +0,0 @@
-This is already typedef'ed like that in gstffmpeg.h, included from this file.
-https://bugs.gentoo.org/show_bug.cgi?id=460208
-
-Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegpipe.h
-===================================================================
---- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegpipe.h
-+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegpipe.h
-@@ -48,8 +48,6 @@ G_BEGIN_DECLS
-   g_cond_signal (m->cond);                                              \
- } G_STMT_END
- 
--typedef struct _GstFFMpegPipe GstFFMpegPipe;
--
- struct _GstFFMpegPipe
- {
-   /* lock for syncing */
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-iscodec.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-iscodec.patch
deleted file mode 100644
index ef198cc33b55..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-iscodec.patch
+++ /dev/null
@@ -1,24 +0,0 @@
-Provide replacement for these functions when not available.
-(e.g ffmpeg-0.10)
-
-Index: gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegutils.h
-===================================================================
---- gst-ffmpeg-0.10.13_p201211.orig/ext/ffmpeg/gstffmpegutils.h
-+++ gst-ffmpeg-0.10.13_p201211/ext/ffmpeg/gstffmpegutils.h
-@@ -99,4 +99,16 @@ av_smp_format_depth(enum AVSampleFormat
- GstBuffer *
- new_aligned_buffer (gint size, GstCaps * caps);
- 
-+#if (LIBAVCODEC_VERSION_INT < AV_VERSION_INT(54,7,0) || (LIBAVCODEC_VERSION_MICRO >= 100 && LIBAVCODEC_VERSION_INT < AV_VERSION_INT(54,8,0)))
-+static inline int av_codec_is_encoder(AVCodec *codec)
-+{
-+  return codec && (codec->encode || codec->encode2);
-+}
-+
-+static inline int av_codec_is_decoder(AVCodec *codec)
-+{
-+  return codec && codec->decode;
-+}
-+#endif
-+
- #endif /* __GST_FFMPEG_UTILS_H__ */
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-planaraudio.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-planaraudio.patch
deleted file mode 100644
index 424bd0cc5c71..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-planaraudio.patch
+++ /dev/null
@@ -1,187 +0,0 @@
-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);
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch
deleted file mode 100644
index 5c34257b867f..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch
+++ /dev/null
@@ -1,25 +0,0 @@
-From: Ettore Di Giacinto <mudler@sabayonlinux.org>
---- ext/ffmpeg/gstffmpegdemux.c~	2015-05-29 00:52:07.601606544 +0200
-+++ ext/ffmpeg/gstffmpegdemux.c	2015-05-29 01:18:18.533113323 +0200
-@@ -781,8 +781,8 @@
-           break;
-         case GST_FORMAT_DEFAULT:
-           gst_query_set_position (query, GST_FORMAT_DEFAULT,
--              gst_util_uint64_scale (timeposition, avstream->r_frame_rate.num,
--                  GST_SECOND * avstream->r_frame_rate.den));
-+              gst_util_uint64_scale (timeposition, avstream->avg_frame_rate.num,
-+                  GST_SECOND * avstream->avg_frame_rate.den));
-           res = TRUE;
-           break;
-         case GST_FORMAT_BYTES:
-@@ -818,8 +818,8 @@
-           break;
-         case GST_FORMAT_DEFAULT:
-           gst_query_set_duration (query, GST_FORMAT_DEFAULT,
--              gst_util_uint64_scale (timeduration, avstream->r_frame_rate.num,
--                  GST_SECOND * avstream->r_frame_rate.den));
-+              gst_util_uint64_scale (timeduration, avstream->avg_frame_rate.num,
-+                  GST_SECOND * avstream->avg_frame_rate.den));
-           res = TRUE;
-           break;
-         case GST_FORMAT_BYTES:
diff --git a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch
deleted file mode 100644
index 7e54f929f674..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch
+++ /dev/null
@@ -1,11 +0,0 @@
-From: Ettore Di Giacinto <mudler@sabayonlinux.org>
---- ext/ffmpeg/gstffmpegcodecmap.c~	2015-05-29 10:50:06.207840323 +0200
-+++ ext/ffmpeg/gstffmpegcodecmap.c	2015-05-29 10:50:25.638042896 +0200
-@@ -1307,7 +1307,6 @@
-     case AV_CODEC_ID_FLIC:
-     case AV_CODEC_ID_VMDVIDEO:
-     case AV_CODEC_ID_VMDAUDIO:
--    case AV_CODEC_ID_SNOW:
-     case AV_CODEC_ID_VIXL:
-     case AV_CODEC_ID_QPEG:
-     case AV_CODEC_ID_PGMYUV:
diff --git a/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r5.ebuild b/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r5.ebuild
deleted file mode 100644
index c7a526c8d453..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r5.ebuild
+++ /dev/null
@@ -1,79 +0,0 @@
-# Copyright 1999-2016 Gentoo Foundation
-# Distributed under the terms of the GNU General Public License v2
-
-EAPI=5
-inherit eutils flag-o-matic multilib-minimal
-
-MY_PN="gst-ffmpeg"
-MY_P=${MY_PN}-${PV}
-
-# Create a major/minor combo for SLOT
-PVP=(${PV//[-\._]/ })
-SLOT=${PVP[0]}.${PVP[1]}
-
-DESCRIPTION="FFmpeg based gstreamer plugin"
-HOMEPAGE="https://gstreamer.freedesktop.org/modules/gst-ffmpeg.html"
-#SRC_URI="https://gstreamer.freedesktop.org/src/${MY_PN}/${MY_P}.tar.bz2"
-SRC_URI="https://dev.gentoo.org/~tetromino/distfiles/${PN}/${MY_P}.tar.xz
-	https://dev.gentoo.org/~tetromino/distfiles/${PN}/${MY_P}-libav-9-patches.tar.xz"
-
-LICENSE="GPL-2"
-KEYWORDS="alpha amd64 arm hppa ia64 ~mips ppc ppc64 sparc x86 ~amd64-fbsd ~x86-fbsd"
-IUSE="libav +orc"
-
-S=${WORKDIR}/${MY_P}
-
-RDEPEND="
-	>=media-libs/gstreamer-0.10.36-r2:0.10[${MULTILIB_USEDEP}]
-	>=media-libs/gst-plugins-base-0.10.36:0.10[${MULTILIB_USEDEP}]
-
-	!libav? ( >=media-video/ffmpeg-1.2.6-r1:0=[postproc,${MULTILIB_USEDEP}] )
-	libav? (
-		>=media-video/libav-11.3:0=[${MULTILIB_USEDEP}]
-		>=media-libs/libpostproc-10.20140517-r1:0=[${MULTILIB_USEDEP}] )
-	orc? ( >=dev-lang/orc-0.4.17[${MULTILIB_USEDEP}] )
-	abi_x86_32? ( !app-emulation/emul-linux-x86-gstplugins[-abi_x86_32(-)] )
-"
-DEPEND="${RDEPEND}
-	>=virtual/pkgconfig-0-r1[${MULTILIB_USEDEP}]
-"
-
-src_prepare() {
-	sed -e 's/sleep 15//' -i configure.ac configure || die
-
-	# libav-9 support backported from gst-plugins-libav-1.1.0
-	epatch ../${MY_P}-libav-9-patches/*.patch
-
-	# compat bits for older ffmpeg/libav releases
-	epatch "${FILESDIR}/${PV}-channel_layout.patch" \
-		"${FILESDIR}/${PV}-iscodec.patch" \
-		"${FILESDIR}/${PV}-coma.patch" \
-		"${FILESDIR}/${PV}-gstffmpegpipe_redef.patch"
-
-	# compatibility with recent releases
-	if has_version '>=media-video/ffmpeg-1.1' || has_version '>=media-video/libav-9' ; then
-		epatch "${FILESDIR}/${PV}-planaraudio.patch"
-		sed -i -e 's/ CODEC_ID/ AV_CODEC_ID/g' \
-			   -e 's/ CodecID/ AVCodecID/g' \
-			   ext/ffmpeg/*.{c,h}
-		epatch "${FILESDIR}/${PV}-ffmpeg2.patch"
-	fi
-
-	# libav 11 migration
-	epatch "${FILESDIR}/${PV}-r_frame_rate.patch"
-	epatch "${FILESDIR}/${PV}-audioresample.patch"
-	epatch "${FILESDIR}/${PV}-snow-codec.patch"
-}
-
-multilib_src_configure() {
-	# always use system ffmpeg if possible
-	ECONF_SOURCE=${S} \
-	econf \
-		--with-system-ffmpeg \
-		$(use_enable orc)
-}
-
-multilib_src_install_all() {
-	einstalldocs
-	prune_libtool_files --modules
-}
diff --git a/media-plugins/gst-plugins-ffmpeg/metadata.xml b/media-plugins/gst-plugins-ffmpeg/metadata.xml
deleted file mode 100644
index 42b76fdc5c1b..000000000000
--- a/media-plugins/gst-plugins-ffmpeg/metadata.xml
+++ /dev/null
@@ -1,8 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<!DOCTYPE pkgmetadata SYSTEM "http://www.gentoo.org/dtd/metadata.dtd">
-<pkgmetadata>
-	<maintainer type="project">
-		<email>gstreamer@gentoo.org</email>
-		<name>GStreamer package maintainers</name>
-	</maintainer>
-</pkgmetadata>