proj/gentoo: Initial commit

This commit represents a new era for Gentoo:
Storing the gentoo-x86 tree in Git, as converted from CVS.

This commit is the start of the NEW history.
Any historical data is intended to be grafted onto this point.

Creation process:
1. Take final CVS checkout snapshot
2. Remove ALL ChangeLog* files
3. Transform all Manifests to thin
4. Remove empty Manifests
5. Convert all stale $Header$/$Id$ CVS keywords to non-expanded Git $Id$
5.1. Do not touch files with -kb/-ko keyword flags.

Signed-off-by: Robin H. Johnson <robbat2@gentoo.org>
X-Thanks: Alec Warner <antarus@gentoo.org> - did the GSoC 2006 migration tests
X-Thanks: Robin H. Johnson <robbat2@gentoo.org> - infra guy, herding this project
X-Thanks: Nguyen Thai Ngoc Duy <pclouds@gentoo.org> - Former Gentoo developer, wrote Git features for the migration
X-Thanks: Brian Harring <ferringb@gentoo.org> - wrote much python to improve cvs2svn
X-Thanks: Rich Freeman <rich0@gentoo.org> - validation scripts
X-Thanks: Patrick Lauer <patrick@gentoo.org> - Gentoo dev, running new 2014 work in migration
X-Thanks: Michał Górny <mgorny@gentoo.org> - scripts, QA, nagging
X-Thanks: All of other Gentoo developers - many ideas and lots of paint on the bikeshed

14 files changed, 766 insertions, 0 deletions

diff --git a/media-plugins/gst-plugins-ffmpeg/Manifest b/media-plugins/gst-plugins-ffmpeg/Manifest
new file mode 100644
index 000000000000..62028ead5984
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/Manifest
@@ -0,0 +1,2 @@
+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
new file mode 100644
index 000000000000..3c115d7b45c5
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-audioresample.patch
@@ -0,0 +1,71 @@
+--- 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
new file mode 100644
index 000000000000..9358f76153ca
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-channel_layout.patch
@@ -0,0 +1,29 @@
+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
new file mode 100644
index 000000000000..0426d00301c5
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-coma.patch
@@ -0,0 +1,16 @@
+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
new file mode 100644
index 000000000000..0ee39dac7f53
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-ffmpeg2.patch
@@ -0,0 +1,139 @@
+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
new file mode 100644
index 000000000000..f76d72affde8
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-gstffmpegpipe_redef.patch
@@ -0,0 +1,16 @@
+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
new file mode 100644
index 000000000000..ef198cc33b55
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-iscodec.patch
@@ -0,0 +1,24 @@
+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
new file mode 100644
index 000000000000..424bd0cc5c71
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-planaraudio.patch
@@ -0,0 +1,187 @@
+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
new file mode 100644
index 000000000000..5c34257b867f
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-r_frame_rate.patch
@@ -0,0 +1,25 @@
+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
new file mode 100644
index 000000000000..7e54f929f674
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/files/0.10.13_p201211-snow-codec.patch
@@ -0,0 +1,11 @@
+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-r2.ebuild b/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r2.ebuild
new file mode 100644
index 000000000000..24ddc643cc4d
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r2.ebuild
@@ -0,0 +1,84 @@
+# Copyright 1999-2014 Gentoo Foundation
+# Distributed under the terms of the GNU General Public License v2
+# $Id$
+
+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]}.${PVP[1]}
+
+DESCRIPTION="FFmpeg based gstreamer plugin"
+HOMEPAGE="http://gstreamer.freedesktop.org/modules/gst-ffmpeg.html"
+#SRC_URI="http://gstreamer.freedesktop.org/src/${MY_PN}/${MY_P}.tar.bz2"
+SRC_URI="http://dev.gentoo.org/~tetromino/distfiles/${PN}/${MY_P}.tar.xz
+	http://dev.gentoo.org/~tetromino/distfiles/${PN}/${MY_P}-libav-9-patches.tar.xz"
+
+LICENSE="GPL-2"
+KEYWORDS="alpha amd64 arm hppa ia64 ~mips ppc ppc64 sparc x86 ~amd64-fbsd ~x86-fbsd"
+IUSE="+orc"
+
+S=${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[${MULTILIB_USEDEP}]
+	>=virtual/ffmpeg-9-r1[${MULTILIB_USEDEP}]
+	|| (
+		>=media-video/ffmpeg-1.2.6-r1:0[${MULTILIB_USEDEP}]
+		>=media-libs/libpostproc-10.20140517-r1[${MULTILIB_USEDEP}]
+	)
+	orc? ( >=dev-lang/orc-0.4.17[${MULTILIB_USEDEP}] )"
+DEPEND="${RDEPEND}
+	>=virtual/pkgconfig-0-r1[${MULTILIB_USEDEP}]"
+RDEPEND="${RDEPEND}
+	abi_x86_32? (
+		!app-emulation/emul-linux-x86-gstplugins[-abi_x86_32(-)]
+	)"
+
+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}/${PV}-ffmpeg2.patch"
+	fi
+}
+
+multilib_src_configure() {
+	# always use system ffmpeg if possible
+	ECONF_SOURCE=${S} \
+	econf \
+		--with-system-ffmpeg \
+		$(use_enable orc)
+}
+
+multilib_src_install_all() {
+	einstalldocs
+	prune_libtool_files --modules
+}
+
+pkg_postinst() {
+	if has_version "media-video/ffmpeg"; then
+		elog "Please note that upstream uses media-video/libav"
+		elog "rather than media-video/ffmpeg. If you encounter any"
+		elog "issues try to move from ffmpeg to libav."
+	fi
+}
diff --git a/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r3.ebuild b/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r3.ebuild
new file mode 100644
index 000000000000..8c283d099682
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r3.ebuild
@@ -0,0 +1,77 @@
+# Copyright 1999-2015 Gentoo Foundation
+# Distributed under the terms of the GNU General Public License v2
+# $Id$
+
+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]}.${PVP[1]}
+
+DESCRIPTION="FFmpeg based gstreamer plugin"
+HOMEPAGE="http://gstreamer.freedesktop.org/modules/gst-ffmpeg.html"
+#SRC_URI="http://gstreamer.freedesktop.org/src/${MY_PN}/${MY_P}.tar.bz2"
+SRC_URI="http://dev.gentoo.org/~tetromino/distfiles/${PN}/${MY_P}.tar.xz
+	http://dev.gentoo.org/~tetromino/distfiles/${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[${MULTILIB_USEDEP}]
+
+	!libav? ( >=media-video/ffmpeg-1.2.6-r1:0=[${MULTILIB_USEDEP}] )
+	libav? (
+		>=media-video/libav-9.12:0=[${MULTILIB_USEDEP}]
+		<media-video/libav-10:0=[${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}/${PV}-ffmpeg2.patch"
+	fi
+}
+
+multilib_src_configure() {
+	# always use system ffmpeg if possible
+	ECONF_SOURCE=${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/gst-plugins-ffmpeg-0.10.13_p201211-r4.ebuild b/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r4.ebuild
new file mode 100644
index 000000000000..63ced4f67d33
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/gst-plugins-ffmpeg-0.10.13_p201211-r4.ebuild
@@ -0,0 +1,80 @@
+# Copyright 1999-2015 Gentoo Foundation
+# Distributed under the terms of the GNU General Public License v2
+# $Id$
+
+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]}.${PVP[1]}
+
+DESCRIPTION="FFmpeg based gstreamer plugin"
+HOMEPAGE="http://gstreamer.freedesktop.org/modules/gst-ffmpeg.html"
+#SRC_URI="http://gstreamer.freedesktop.org/src/${MY_PN}/${MY_P}.tar.bz2"
+SRC_URI="http://dev.gentoo.org/~tetromino/distfiles/${PN}/${MY_P}.tar.xz
+	http://dev.gentoo.org/~tetromino/distfiles/${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[${MULTILIB_USEDEP}]
+
+	!libav? ( >=media-video/ffmpeg-1.2.6-r1:0=[${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
new file mode 100644
index 000000000000..d1096fb54bd8
--- /dev/null
+++ b/media-plugins/gst-plugins-ffmpeg/metadata.xml
@@ -0,0 +1,5 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE pkgmetadata SYSTEM "http://www.gentoo.org/dtd/metadata.dtd">
+<pkgmetadata>
+	<herd>gstreamer</herd>
+</pkgmetadata>