media-sound/xmms2: support stable API for libav, bug #540890

Latest stable media-video/ffmpeg-2.8.6 provides both functions:
  avcodec_decode_audio3
  avcodec_decode_audio4

While latest stable media-video/libav-11.3 provides only
  avcodec_decode_audio4

Pulled large patchset from upstream to support audio4 API.

Builds fine on both stable virtual/ffmpeg implementations
and unstable libav.

Reported-by: Toralf Förster
Bug: https://bugs.gentoo.org/540890

Package-Manager: portage-2.2.28

8 files changed, 1392 insertions, 0 deletions

diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p1.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p1.patch
new file mode 100644
index 000000000000..21ed6492a44d
--- /dev/null
+++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p1.patch
@@ -0,0 +1,123 @@
+commit 8831bc77d705c03b3f8081de0520dd10afa85c69
+Author: Uli Franke <cls@nebadje.org>
+Date:   Tue Jan 17 23:23:46 2012 +0100
+
+    BUG(2509): Avoid unaligned reads in avcodec xform.
+
+diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c
+index fe58fc5..1b4a659 100644
+--- a/src/plugins/avcodec/avcodec.c
++++ b/src/plugins/avcodec/avcodec.c
+@@ -36,6 +36,9 @@ typedef struct {
+ 	guint buffer_size;
+ 	gboolean no_demuxer;
+ 
++	gchar *read_out_buffer;
++	gint read_out_buffer_size;
++
+ 	guint channels;
+ 	guint samplerate;
+ 	xmms_sample_format_t sampleformat;
+@@ -107,6 +110,7 @@ xmms_avcodec_destroy (xmms_xform_t *xform)
+ 
+ 	avcodec_close (data->codecctx);
+ 	av_free (data->codecctx);
++	av_free (data->read_out_buffer);
+ 
+ 	g_string_free (data->outbuf, TRUE);
+ 	g_free (data->buffer);
+@@ -132,6 +136,9 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 	data->buffer_size = AVCODEC_BUFFER_SIZE;
+ 	data->codecctx = NULL;
+ 
++	data->read_out_buffer = av_malloc (AVCODEC_MAX_AUDIO_FRAME_SIZE);
++	data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
++
+ 	xmms_xform_private_data_set (xform, data);
+ 
+ 	avcodec_init ();
+@@ -196,7 +203,7 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 		} else {
+ 			/* A demuxer plugin forgot to give decoder config? */
+ 			xmms_log_error ("Decoder config data not found!");
+-			return FALSE;
++			goto err;
+ 		}
+ 	}
+ 
+@@ -220,7 +227,7 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 
+ 		/* some codecs need to have something read before they set
+ 		 * the samplerate and channels correctly, unfortunately... */
+-		if ((ret = xmms_avcodec_read (xform, buf, 42, &error)) > 0) {
++		if ((ret = xmms_avcodec_read (xform, buf, sizeof (buf), &error)) > 0) {
+ 			g_string_insert_len (data->outbuf, 0, buf, ret);
+ 		} else {
+ 			XMMS_DBG ("First read failed, codec is not working...");
+@@ -251,6 +258,9 @@ err:
+ 	if (data->codecctx) {
+ 		av_free (data->codecctx);
+ 	}
++	if (data->read_out_buffer) {
++		av_free (data->read_out_buffer);
++	}
+ 	g_string_free (data->outbuf, TRUE);
+ 	g_free (data->extradata);
+ 	g_free (data);
+@@ -263,8 +273,7 @@ xmms_avcodec_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len,
+                    xmms_error_t *error)
+ {
+ 	xmms_avcodec_data_t *data;
+-	char outbuf[AVCODEC_MAX_AUDIO_FRAME_SIZE];
+-	gint outbufsize, bytes_read = 0;
++	gint bytes_read = 0;
+ 	guint size;
+ 
+ 	data = xmms_xform_private_data_get (xform);
+@@ -330,9 +339,9 @@ xmms_avcodec_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len,
+ 		packet.data = data->buffer;
+ 		packet.size = data->buffer_length;
+ 
+-		outbufsize = sizeof (outbuf);
+-		bytes_read = avcodec_decode_audio3 (data->codecctx, (short *) outbuf,
+-		                                    &outbufsize, &packet);
++		data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
++		bytes_read = avcodec_decode_audio3 (data->codecctx, (short *) data->read_out_buffer,
++		                                    &data->read_out_buffer_size, &packet);
+ 
+ 		/* The DTS decoder of ffmpeg is buggy and always returns
+ 		 * the input buffer length, get frame length from header */
+@@ -354,8 +363,8 @@ xmms_avcodec_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len,
+ 
+ 		data->buffer_length -= bytes_read;
+ 
+-		if (outbufsize > 0) {
+-			g_string_append_len (data->outbuf, outbuf, outbufsize);
++		if (data->read_out_buffer_size > 0) {
++			g_string_append_len (data->outbuf, data->read_out_buffer, data->read_out_buffer_size);
+ 		}
+ 
+ 		size = MIN (data->outbuf->len, len);
+@@ -371,8 +380,7 @@ static gint64
+ xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples, xmms_xform_seek_mode_t whence, xmms_error_t *err)
+ {
+ 	xmms_avcodec_data_t *data;
+-	char outbuf[AVCODEC_MAX_AUDIO_FRAME_SIZE];
+-	gint outbufsize, bytes_read = 0;
++	gint bytes_read = 0;
+ 	gint64 ret = -1;
+ 
+ 	g_return_val_if_fail (xform, -1);
+@@ -396,9 +404,9 @@ xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples, xmms_xform_seek_mode_t w
+ 		packet.data = data->buffer;
+ 		packet.size = data->buffer_length;
+ 
+-		outbufsize = sizeof (outbuf);
+-		bytes_read = avcodec_decode_audio3 (data->codecctx, (short *) outbuf,
+-		                                    &outbufsize, &packet);
++		data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
++		bytes_read = avcodec_decode_audio3 (data->codecctx, (short *) data->read_out_buffer,
++		                                    &data->read_out_buffer_size, &packet);
+ 
+ 		if (bytes_read < 0 || bytes_read > data->buffer_length) {
+ 			XMMS_DBG ("Error decoding data!");
diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch
new file mode 100644
index 000000000000..46b5b1d3b926
--- /dev/null
+++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p2.patch
@@ -0,0 +1,171 @@
+commit b614459dc1ea353d6c24b4a77c7f92a5577d5bc3
+Author: Uli Franke <cls@nebadje.org>
+Date:   Thu Jan 19 11:53:57 2012 +0100
+
+    BUG(2510): Add more bitrates/samplerates to AAC/ALAC.
+
+diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c
+index 1b4a659..b32de4d 100644
+--- a/src/plugins/avcodec/avcodec.c
++++ b/src/plugins/avcodec/avcodec.c
+@@ -60,6 +60,7 @@ static gint xmms_avcodec_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len
+                                xmms_error_t *error);
+ static gint64 xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples,
+                                  xmms_xform_seek_mode_t whence, xmms_error_t *err);
++static xmms_sample_format_t xmms_avcodec_translate_sample_format (enum AVSampleFormat av_sample_format);
+ 
+ /*
+  * Plugin header
+@@ -168,12 +169,12 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 		data->channels = ret;
+ 	}
+ 
+-	/* bitrate required for WMA files */
++	/* Required by WMA xform. */
+ 	xmms_xform_auxdata_get_int (xform,
+ 	                            "bitrate",
+ 	                            &data->bitrate);
+ 
+-	/* ALAC and MAC require bits per sample field to be 16 */
++	/* Required by tta and apefile xforms. */
+ 	xmms_xform_auxdata_get_int (xform,
+ 	                            "samplebits",
+ 	                            &data->samplebits);
+@@ -238,12 +239,17 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 
+ 	data->samplerate = data->codecctx->sample_rate;
+ 	data->channels = data->codecctx->channels;
++	data->sampleformat = xmms_avcodec_translate_sample_format (data->codecctx->sample_fmt);
++	if (data->sampleformat == XMMS_SAMPLE_FORMAT_UNKNOWN) {
++		avcodec_close (data->codecctx);
++		goto err;
++	}
+ 
+ 	xmms_xform_outdata_type_add (xform,
+ 	                             XMMS_STREAM_TYPE_MIMETYPE,
+ 	                             "audio/pcm",
+ 	                             XMMS_STREAM_TYPE_FMT_FORMAT,
+-	                             XMMS_SAMPLE_FORMAT_S16,
++	                             data->sampleformat,
+ 	                             XMMS_STREAM_TYPE_FMT_CHANNELS,
+ 	                             data->channels,
+ 	                             XMMS_STREAM_TYPE_FMT_SAMPLERATE,
+@@ -428,3 +434,23 @@ xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples, xmms_xform_seek_mode_t w
+ 
+ 	return ret;
+ }
++
++static xmms_sample_format_t
++xmms_avcodec_translate_sample_format (enum AVSampleFormat av_sample_format)
++{
++	switch (av_sample_format) {
++	case AV_SAMPLE_FMT_U8:
++		return XMMS_SAMPLE_FORMAT_U8;
++	case AV_SAMPLE_FMT_S16:
++		return XMMS_SAMPLE_FORMAT_S16;
++	case AV_SAMPLE_FMT_S32:
++		return XMMS_SAMPLE_FORMAT_S32;
++	case AV_SAMPLE_FMT_FLT:
++		return XMMS_SAMPLE_FORMAT_FLOAT;
++	case AV_SAMPLE_FMT_DBL:
++		return XMMS_SAMPLE_FORMAT_DOUBLE;
++	default:
++		XMMS_DBG ("AVSampleFormat (%i) not supported.", av_sample_format);
++		return XMMS_SAMPLE_FORMAT_UNKNOWN;
++	}
++}
+diff --git a/src/plugins/mp4/mp4.c b/src/plugins/mp4/mp4.c
+index 7c915c4..3ee9357 100644
+--- a/src/plugins/mp4/mp4.c
++++ b/src/plugins/mp4/mp4.c
+@@ -186,9 +186,6 @@ xmms_mp4_init (xmms_xform_t *xform)
+ 	xmms_xform_auxdata_set_bin (xform, "decoder_config", tmpbuf, tmpbuflen);
+ 	g_free (tmpbuf);
+ 
+-	/* This is only for ALAC to set 16-bit samples, ignored for AAC */
+-	xmms_xform_auxdata_set_int (xform, "samplebits", 16);
+-
+ 	xmms_mp4_get_mediainfo (xform);
+ 
+ 	XMMS_DBG ("MP4 demuxer inited successfully!");
+@@ -288,7 +285,7 @@ xmms_mp4_get_mediainfo (xmms_xform_t *xform)
+ 	data = xmms_xform_private_data_get (xform);
+ 	g_return_if_fail (data);
+ 
+-	if ((temp = mp4ff_get_sample_rate (data->mp4ff, data->track)) >= 0) {
++	if ((temp = mp4ff_get_sample_rate (data->mp4ff, data->track)) > 0) {
+ 		glong srate = temp;
+ 
+ 		if ((temp = mp4ff_get_track_duration_use_offsets (data->mp4ff,
+@@ -492,7 +489,7 @@ xmms_mp4_get_track (xmms_xform_t *xform, mp4ff_t *infile)
+ 		case 0x69: /* MPEG-2 audio */
+ 		case 0x6B: /* MPEG-1 audio */
+ 			continue;
+-		case 0xff:
++		case 0xff: /* ALAC */
+ 			chans = mp4ff_get_channel_count (infile, i);
+ 			rate = mp4ff_get_sample_rate (infile, i);
+ 			if (chans <= 0 || rate <= 0) {
+diff --git a/src/plugins/mp4/mp4ff/README.xmms2 b/src/plugins/mp4/mp4ff/README.xmms2
+index c2737c5..8021618 100644
+--- a/src/plugins/mp4/mp4ff/README.xmms2
++++ b/src/plugins/mp4/mp4ff/README.xmms2
+@@ -12,3 +12,4 @@ Changes:
+  * Add value_length variable to tag type and use it when adding new item-value pairs,
+    necessary for cover art since it's binary data and can't be handled as a string
+  * Add support for Apple Lossless audio files
++ * Add a workaround for supporting higher samplerates.
+diff --git a/src/plugins/mp4/mp4ff/mp4ff.c b/src/plugins/mp4/mp4ff/mp4ff.c
+index ee7f7fb..b6f0a37 100644
+--- a/src/plugins/mp4/mp4ff/mp4ff.c
++++ b/src/plugins/mp4/mp4ff/mp4ff.c
+@@ -32,6 +32,8 @@
+ #include <string.h>
+ #include "mp4ffint.h"
+ 
++static uint32_t mp4ff_normalize_flawed_sample_rate (uint16_t samplerate);
++
+ mp4ff_t *mp4ff_open_read(mp4ff_callback_t *f)
+ {
+     mp4ff_t *ff = malloc(sizeof(mp4ff_t));
+@@ -304,12 +306,39 @@ int32_t mp4ff_num_samples(const mp4ff_t *f, const int32_t track)
+     return total;
+ }
+ 
++static uint32_t
++mp4ff_normalize_flawed_sample_rate (uint16_t samplerate)
++{
++	/* A list of common rates can be found at
++	 * http://en.wikipedia.org/wiki/Sampling_rate */
++	uint32_t rates[] = {8000, 11025, 16000, 22050, 32000, 44056, 44100,
++	                  47250, 48000, 50000, 50400, 88200, 96000, 176400,
++	                  192000, 352800, 384000, 0};
++	uint32_t* rate;
++
++	/* First check standard rates. */
++	for (rate = rates; *rate; rate++) {
++		if (*rate == samplerate) {
++			return *rate;
++		}
++	}
++
++	/* No standard rates matching - check if sample rate got truncated when
++	 * added to MP4 container */
++	for (rate = rates; *rate; rate++) {
++		if ((*rate & 0x0000FFFF) == samplerate) {
++			return *rate;
++		}
++	}
+ 
++	/* Failed to find a standard rate - we give up returning the original rate */
++	return samplerate;
++}
+ 
+ 
+ uint32_t mp4ff_get_sample_rate(const mp4ff_t *f, const int32_t track)
+ {
+-	return f->track[track]->sampleRate;
++	return mp4ff_normalize_flawed_sample_rate (f->track[track]->sampleRate);
+ }
+ 
+ uint32_t mp4ff_get_channel_count(const mp4ff_t * f,const int32_t track)
diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p3.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p3.patch
new file mode 100644
index 000000000000..a9145c0d3d26
--- /dev/null
+++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p3.patch
@@ -0,0 +1,388 @@
+commit 4d0682030e20a8ed218f4ff924554f93d276d9ee
+Author: Anthony Garcia <lagg@lavabit.com>
+Date:   Thu Apr 22 16:59:37 2010 -0700
+
+    OTHER: Cleanup
+    
+    Re-enabled nellymoser (ffmpeg appears to be okay with it now)
+    
+    Fixed possible infinite loop in the code that handles the data (if any)
+    between the header and tag data.
+
+diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c
+index 6c9fea8..5554056 100644
+--- a/src/plugins/avcodec/avcodec.c
++++ b/src/plugins/avcodec/avcodec.c
+@@ -90,7 +90,7 @@ xmms_avcodec_plugin_setup (xmms_xform_plugin_t *xform_plugin)
+ 	xmms_magic_add ("A/52 (AC-3) header", "audio/x-ffmpeg-ac3",
+ 	                "0 beshort 0x0b77", NULL);
+ 	xmms_magic_add ("DTS header", "audio/x-ffmpeg-dca",
+-	                "0 belong 0x7ffe8001", NULL); 
++	                "0 belong 0x7ffe8001", NULL);
+ 
+ 	xmms_xform_plugin_indata_add (xform_plugin,
+ 	                              XMMS_STREAM_TYPE_MIMETYPE,
+@@ -197,7 +197,8 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 		    !strcmp (data->codec_id, "adpcm_swf") ||
+ 		    !strcmp (data->codec_id, "pcm_s16le") ||
+ 		    !strcmp (data->codec_id, "ac3") ||
+-		    !strcmp (data->codec_id, "dca")) {
++		    !strcmp (data->codec_id, "dca") ||
++		    !strcmp (data->codec_id, "nellymoser")) {
+ 			/* number 1024 taken from libavformat raw.c RAW_PACKET_SIZE */
+ 			data->extradata = g_malloc0 (1024);
+ 			data->extradata_size = 1024;
+diff --git a/src/plugins/flv/flv.c b/src/plugins/flv/flv.c
+index 440010c..266fea6 100644
+--- a/src/plugins/flv/flv.c
++++ b/src/plugins/flv/flv.c
+@@ -25,29 +25,41 @@
+  * and other info, then data
+  */
+ #define FLV_TAG_SIZE 11
+-/* random constant */
+ #define FLV_CHUNK_SIZE 4096
+ 
+-/* let libavcodec take care of swapping sample bytes */
+-static const gchar *mime_pcm_s16le = "audio/x-ffmpeg-pcm_s16le";
+-static const gchar *fmt_mime[11] = {
+-	/* Supported when samples are 8 bit
+-	 * (otherwise there's no way of knowing endianness)
+-	 */
+-	"audio/pcm",
+-	"audio/x-ffmpeg-adpcm_swf",
+-	"audio/mpeg",
+-	/* if bps is 8 bit u8
+-	 * if bps is 16 bit sle16
+-	 */
+-	"audio/pcm",
+-	/* libavcodec can't handle nelly without dying yet */
+-	/*"audio/x-ffmpeg-nellymoser",
+-	"audio/x-ffmpeg-nellymoser",
+-	"audio/x-ffmpeg-nellymoser",*/
+-	"", "", "",
+-	"", "", "",
+-	"audio/aac"
++typedef enum {
++	/* Only u8 bit samples since
++	   there's no way to determine endianness
++	*/
++	CODEC_PCM_HOST,
++	CODEC_ADPCM,
++	CODEC_MP3,
++	/* 8 bps: unsigned
++	   16 bps: signed
++	*/
++	CODEC_PCM_LE,
++	CODEC_NELLYMOSER_16K,
++	CODEC_NELLYMOSER_8K,
++	/* Uses the sample rate in
++	   the tag as normal
++	*/
++	CODEC_NELLYMOSER,
++	CODEC_AAC = 10
++} xmms_flv_codec_id;
++
++struct xmms_flv_codec_table {
++	xmms_flv_codec_id id;
++	const gchar *mime;
++} static flv_codecs[] = {
++	{CODEC_PCM_HOST, "audio/pcm"},
++	{CODEC_ADPCM, "audio/x-ffmpeg-adpcm_swf"},
++	{CODEC_MP3, "audio/mpeg"},
++	/* Will be audio/x-ffmpeg-pcm_s16le if bps is 16 */
++	{CODEC_PCM_LE, "audio/pcm"},
++	{CODEC_NELLYMOSER_16K, "audio/x-ffmpeg-nellymoser"},
++	{CODEC_NELLYMOSER_8K, "audio/x-ffmpeg-nellymoser"},
++	{CODEC_NELLYMOSER, "audio/x-ffmpeg-nellymoser"},
++	{CODEC_AAC, "audio/aac"}
+ };
+ 
+ typedef struct {
+@@ -111,23 +123,26 @@ static gboolean
+ xmms_flv_init (xmms_xform_t *xform)
+ {
+ 	xmms_sample_format_t bps;
+-	gint readret;
++	gint readret, i;
+ 	guint8 channels, flags, format;
+-	guint8 header[FLV_TAG_SIZE + 5];
+-	const gchar *mime;
++	guint8 header[FLV_TAG_SIZE + 1];
+ 	guint32 dataoffset, samplerate;
+ 	xmms_error_t err;
+ 	xmms_flv_data_t *flvdata;
++	struct xmms_flv_codec_table *codec = NULL;
++
++	flvdata = g_new0 (xmms_flv_data_t, 1);
++	xmms_xform_private_data_set (xform, flvdata);
+ 
+ 	readret = xmms_xform_read (xform, header, FLV_HDR_SIZE, &err);
+ 	if (readret != FLV_HDR_SIZE) {
+ 		xmms_log_error ("Header read error");
+-		return FALSE;
++		goto init_err;
+ 	}
+ 
+ 	if ((header[4] & HAS_AUDIO) != HAS_AUDIO) {
+ 		xmms_log_error ("FLV has no audio stream");
+-		return FALSE;
++		goto init_err;
+ 	}
+ 
+ 	dataoffset = get_be32 (&header[5]) - FLV_HDR_SIZE;
+@@ -140,7 +155,7 @@ xmms_flv_init (xmms_xform_t *xform)
+ 		                           dataoffset : FLV_HDR_SIZE, &err);
+ 		if (readret <= 0) {
+ 			xmms_log_error ("Error reading header:tag body gap");
+-			return FALSE;
++			goto init_err;
+ 		}
+ 
+ 		dataoffset -= readret;
+@@ -148,86 +163,99 @@ xmms_flv_init (xmms_xform_t *xform)
+ 
+ 	if (next_audio_tag (xform) <= 0) {
+ 		xmms_log_error ("Can't find first audio tag");
+-		return FALSE;
++		goto init_err;
+ 	}
+ 
+-	if (xmms_xform_peek (xform, header, FLV_TAG_SIZE + 5, &err) < FLV_TAG_SIZE + 5) {
++	if (xmms_xform_read (xform, header, FLV_TAG_SIZE + 1, &err) < FLV_TAG_SIZE + 1) {
+ 		xmms_log_error ("Can't read first audio tag");
+-		return FALSE;
++		goto init_err;
+ 	}
+ 
+-	flags = header[FLV_TAG_SIZE + 4];
++	flags = header[11];
+ 	XMMS_DBG ("Audio flags: %X", flags);
+ 
+-	switch (flags&12) {
+-		case 0: samplerate = 5512; break;
+-		case 4: samplerate = 11025; break;
+-		case 8: samplerate = 22050; break;
+-		case 12: samplerate = 44100; break;
+-		default: samplerate = 8000; break;
++	format = flags >> 4;
++	for (i = 0; i < G_N_ELEMENTS (flv_codecs); i++) {
++		if (flv_codecs[i].id == format) {
++			codec = &flv_codecs[i];
++			break;
++		}
+ 	}
+ 
+-	if (flags&2) {
+-		bps = XMMS_SAMPLE_FORMAT_S16;
++	if (flags & 1) {
++		channels = 2;
+ 	} else {
+-		bps = XMMS_SAMPLE_FORMAT_U8;
++		channels = 1;
+ 	}
+ 
+-	if (flags&1) {
+-		channels = 2;
++	if (flags & 2) {
++		bps = XMMS_SAMPLE_FORMAT_S16;
+ 	} else {
+-		channels = 1;
++		bps = XMMS_SAMPLE_FORMAT_U8;
+ 	}
+ 
+-	format = flags >> 4;
+-	mime = (format <= 10)? fmt_mime[format] : NULL;
+-	switch (format) {
+-		case 0:
+-			/* If the flv has an HE PCM audio stream, the
+-			 * samples must be unsigned and 8 bits long
+-			 */
+-			if (bps != XMMS_SAMPLE_FORMAT_U8) {
+-				xmms_log_error ("Only u8 HE PCM is supported");
+-				return FALSE;
+-			}
+-			break;
+-		case 3:
+-			if (bps == XMMS_SAMPLE_FORMAT_S16) {
+-				mime = mime_pcm_s16le;
+-			}
+-			break;
++	switch ((flags & 12) >> 2) {
++		case 0: samplerate = 5512; break;
++		case 1: samplerate = 11025; break;
++		case 2: samplerate = 22050; break;
++		case 3: samplerate = 44100; break;
++		default: samplerate = 8000; break;
+ 	}
+ 
+-	if (mime && *mime) {
+-		flvdata = g_new0 (xmms_flv_data_t, 1);
++	if (codec) {
++		switch (codec->id) {
++			case CODEC_PCM_HOST:
++				if (bps != XMMS_SAMPLE_FORMAT_U8) {
++					xmms_log_error ("Only u8 HE PCM is supported");
++					goto init_err;
++				}
++				break;
++			case CODEC_PCM_LE:
++				if (bps == XMMS_SAMPLE_FORMAT_S16) {
++					codec->mime = "audio/x-ffmpeg-pcm_s16le";
++				}
++				break;
++			case CODEC_NELLYMOSER_16K:
++				samplerate = 16000;
++				break;
++			case CODEC_NELLYMOSER_8K:
++				samplerate = 8000;
++				break;
++			default:
++				break;
++		}
++
+ 		flvdata->format = format;
++		flvdata->last_datasize = get_be24 (&header[1]) - 1;
+ 
+ 		XMMS_DBG ("Rate: %d, bps: %d, channels: %d", samplerate,
+ 		          bps, channels);
+ 
+-		xmms_xform_private_data_set (xform, flvdata);
+ 		xmms_xform_outdata_type_add (xform,
+ 		                             XMMS_STREAM_TYPE_MIMETYPE,
+-					     mime,
+-					     XMMS_STREAM_TYPE_FMT_SAMPLERATE,
+-					     samplerate,
+-					     XMMS_STREAM_TYPE_FMT_FORMAT,
+-					     bps,
+-					     XMMS_STREAM_TYPE_FMT_CHANNELS,
+-					     channels,
+-					     XMMS_STREAM_TYPE_END);
++		                             codec->mime,
++		                             XMMS_STREAM_TYPE_FMT_SAMPLERATE,
++		                             samplerate,
++		                             XMMS_STREAM_TYPE_FMT_FORMAT,
++		                             bps,
++		                             XMMS_STREAM_TYPE_FMT_CHANNELS,
++		                             channels,
++		                             XMMS_STREAM_TYPE_END);
+ 		return TRUE;
+ 	} else {
+ 		xmms_log_error ("Unsupported audio format");
+-		return FALSE;
+ 	}
++
++init_err:
++	g_free (flvdata);
++	return FALSE;
+ }
+ 
+ static gint
+ xmms_flv_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len, xmms_error_t *err)
+ {
+-	gint ret = 0, thismuch = FLV_TAG_SIZE + 5;
+-	guint8 header[FLV_TAG_SIZE + 6], gap = 1;
++	gint ret = 0, thismuch = FLV_TAG_SIZE + 1;
++	guint8 header[FLV_TAG_SIZE + 1];
+ 	xmms_flv_data_t *data = NULL;
+ 
+ 	data = xmms_xform_private_data_get (xform);
+@@ -236,12 +264,8 @@ xmms_flv_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len, xmms_error_t *
+ 		xmms_xform_auxdata_barrier (xform);
+ 		ret = next_audio_tag (xform);
+ 		if (ret > 0) {
+-			if (data->format == 10) {
+-				thismuch++;
+-				gap++;
+-			}
+ 			if (xmms_xform_read (xform, header, thismuch, err) == thismuch) {
+-				data->last_datasize = get_be24 (&header[5]) - gap;
++				data->last_datasize = get_be24 (&header[1]) - 1;
+ 			} else {
+ 				xmms_log_error ("Need %d bytes", thismuch);
+ 				return -1;
+@@ -280,40 +304,51 @@ xmms_flv_destroy (xmms_xform_t *xform)
+ static gint
+ next_audio_tag (xmms_xform_t *xform)
+ {
+-	guint8 header[FLV_TAG_SIZE + 4];
++	guint8 header[FLV_TAG_SIZE];
+ 	guint8 dumb[FLV_CHUNK_SIZE];
+ 	gint ret = 0;
+ 	xmms_error_t err;
+-	guint32 datasize = 0;
++	xmms_flv_data_t *data;
++
++	data = xmms_xform_private_data_get (xform);
+ 
+ 	do {
+-		/* there's a last 4 bytes at the end of an FLV giving the final
+-		 * tag's size, this isn't an error
+-		 */
+-		ret = xmms_xform_peek (xform, header, FLV_TAG_SIZE + 4, &err);
+-		if ((ret < FLV_TAG_SIZE) && (ret > -1)) {
+-			ret = 0;
+-			break;
+-		} else if (ret == -1) {
+-			xmms_log_error ("%s", xmms_error_message_get (&err));
+-			break;
+-		}
++		/* If > 0 assume we're in the middle of a tag's data */
++		if (!data->last_datasize) {
++			/* There are 4 bytes before an actual tag giving
++			   the previous tag's size. The first size in an
++			   flv is always 0.
++			*/
++			if (xmms_xform_read (xform, header, 4, &err) != 4) {
++				xmms_log_error ("Couldn't read last tag size");
++				return -1;
++			}
+ 
+-		if (header[4] == 8) {
+-			/* woo audio tag! */
+-			break;
+-		}
++			ret = xmms_xform_peek (xform, header, FLV_TAG_SIZE, &err);
++			if ((ret < FLV_TAG_SIZE) && (ret > -1)) {
++				return 0;
++			} else if (ret == -1) {
++				xmms_log_error ("%s", xmms_error_message_get (&err));
++				return ret;
++			}
++
++			if (header[0] == 8) {
++				/* woo audio tag! */
++				break;
++			}
+ 
+-		ret = xmms_xform_read (xform, header, FLV_TAG_SIZE + 4, &err);
+-		if (ret <= 0) { return ret; }
++			if ((ret = xmms_xform_read (xform, header, FLV_TAG_SIZE, &err)) <= 0) {
++				return ret;
++			}
+ 
+-		datasize = get_be24 (&header[5]);
++			data->last_datasize = get_be24 (&header[1]);
++		}
+ 
+-		while (datasize) {
++		while (data->last_datasize) {
+ 			ret = xmms_xform_read (xform, dumb,
+-			                         (datasize < FLV_CHUNK_SIZE) ?
+-			                         datasize : FLV_CHUNK_SIZE,
+-			                         &err);
++			                       (data->last_datasize < FLV_CHUNK_SIZE) ?
++			                       data->last_datasize : FLV_CHUNK_SIZE,
++			                       &err);
+ 			if (ret == 0) {
+ 				xmms_log_error ("Data field short!");
+ 				break;
+@@ -323,7 +358,7 @@ next_audio_tag (xmms_xform_t *xform)
+ 				break;
+ 			}
+ 
+-			datasize -= ret;
++			data->last_datasize -= ret;
+ 		}
+ 
+ 	} while (ret);
diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p4.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p4.patch
new file mode 100644
index 000000000000..552f202df19d
--- /dev/null
+++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p4.patch
@@ -0,0 +1,296 @@
+commit 4198d9bf5dff517740ed51b22313367f156107e1
+Author: Erik Massop <e.massop@hccnet.nl>
+Date:   Sun Dec 22 17:19:30 2013 +0100
+
+    OTHER: Split xmms_avcodec_read, remove some duplicate code
+
+diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c
+index 5b9b606..eed7964 100644
+--- a/src/plugins/avcodec/avcodec.c
++++ b/src/plugins/avcodec/avcodec.c
+@@ -57,6 +57,9 @@ typedef struct {
+ static gboolean xmms_avcodec_plugin_setup (xmms_xform_plugin_t *xform_plugin);
+ static gboolean xmms_avcodec_init (xmms_xform_t *xform);
+ static void xmms_avcodec_destroy (xmms_xform_t *xform);
++static gint xmms_avcodec_internal_read_some (xmms_xform_t *xform, xmms_avcodec_data_t *data, xmms_error_t *error);
++static gint xmms_avcodec_internal_decode_some (xmms_avcodec_data_t *data);
++static void xmms_avcodec_internal_append (xmms_avcodec_data_t *data);
+ static gint xmms_avcodec_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len,
+                                xmms_error_t *error);
+ static gint64 xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples,
+@@ -281,101 +284,24 @@ xmms_avcodec_read (xmms_xform_t *xform, xmms_sample_t *buf, gint len,
+                    xmms_error_t *error)
+ {
+ 	xmms_avcodec_data_t *data;
+-	gint bytes_read = 0;
+ 	guint size;
+ 
+ 	data = xmms_xform_private_data_get (xform);
+ 	g_return_val_if_fail (data, -1);
+ 
+-	size = MIN (data->outbuf->len, len);
+-	while (size == 0) {
+-		AVPacket packet;
+-		av_init_packet (&packet);
++	while (0 == (size = MIN (data->outbuf->len, len))) {
++		gint res;
+ 
+ 		if (data->no_demuxer || data->buffer_length == 0) {
+-			gint read_total;
+-
+-			bytes_read = xmms_xform_read (xform,
+-			                              (gchar *) (data->buffer + data->buffer_length),
+-			                              data->buffer_size - data->buffer_length,
+-			                              error);
+-
+-			if (bytes_read < 0) {
+-				XMMS_DBG ("Error while reading data");
+-				return bytes_read;
+-			} else if (bytes_read == 0) {
+-				XMMS_DBG ("EOF");
+-				return 0;
+-			}
+-
+-			read_total = bytes_read;
+-
+-			/* If we have a demuxer plugin, make sure we read the whole packet */
+-			while (read_total == data->buffer_size && !data->no_demuxer) {
+-				/* multiply the buffer size and try to read again */
+-				data->buffer = g_realloc (data->buffer, data->buffer_size * 2);
+-				bytes_read = xmms_xform_read (xform,
+-				                              (gchar *) data->buffer +
+-				                                data->buffer_size,
+-				                              data->buffer_size,
+-				                              error);
+-				data->buffer_size *= 2;
+-
+-				if (bytes_read < 0) {
+-					XMMS_DBG ("Error while reading data");
+-					return bytes_read;
+-				}
+-
+-				read_total += bytes_read;
+-
+-				if (read_total < data->buffer_size) {
+-					/* finally double the buffer size for performance reasons, the
+-					 * hotspot handling likes to fit two frames in the buffer */
+-					data->buffer = g_realloc (data->buffer, data->buffer_size * 2);
+-					data->buffer_size *= 2;
+-					XMMS_DBG ("Reallocated avcodec internal buffer to be %d bytes",
+-					          data->buffer_size);
+-
+-					break;
+-				}
+-			}
+-
+-			/* Update the buffer length */
+-			data->buffer_length += read_total;
+-		}
+-
+-		packet.data = data->buffer;
+-		packet.size = data->buffer_length;
+-
+-		data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+-		bytes_read = avcodec_decode_audio3 (data->codecctx, (short *) data->read_out_buffer,
+-		                                    &data->read_out_buffer_size, &packet);
++			gint bytes_read;
+ 
+-		/* The DTS decoder of ffmpeg is buggy and always returns
+-		 * the input buffer length, get frame length from header */
+-		if (!strcmp (data->codec_id, "dca") && bytes_read > 0) {
+-			bytes_read = ((int)data->buffer[5] << 12) |
+-			             ((int)data->buffer[6] << 4) |
+-			             ((int)data->buffer[7] >> 4);
+-			bytes_read = (bytes_read & 0x3fff) + 1;
++			bytes_read = xmms_avcodec_internal_read_some (xform, data, error);
++			if (bytes_read <= 0) { return bytes_read; }
+ 		}
+ 
+-		if (bytes_read < 0 || bytes_read > data->buffer_length) {
+-			XMMS_DBG ("Error decoding data!");
+-			return -1;
+-		} else if (bytes_read != data->buffer_length) {
+-			g_memmove (data->buffer,
+-			           data->buffer + bytes_read,
+-			           data->buffer_length - bytes_read);
+-		}
+-
+-		data->buffer_length -= bytes_read;
+-
+-		if (data->read_out_buffer_size > 0) {
+-			g_string_append_len (data->outbuf, data->read_out_buffer, data->read_out_buffer_size);
+-		}
+-
+-		size = MIN (data->outbuf->len, len);
++		res = xmms_avcodec_internal_decode_some (data);
++		if (res < 0) { return res; }
++		if (res > 0) { xmms_avcodec_internal_append (data); }
+ 	}
+ 
+ 	memcpy (buf, data->outbuf->str, size);
+@@ -388,7 +314,6 @@ static gint64
+ xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples, xmms_xform_seek_mode_t whence, xmms_error_t *err)
+ {
+ 	xmms_avcodec_data_t *data;
+-	gint bytes_read = 0;
+ 	gint64 ret = -1;
+ 
+ 	g_return_val_if_fail (xform, -1);
+@@ -406,23 +331,11 @@ xmms_avcodec_seek (xmms_xform_t *xform, gint64 samples, xmms_xform_seek_mode_t w
+ 
+ 	/* The buggy ape decoder doesn't flush buffers, so we need to finish decoding
+ 	 * the frame before seeking to avoid segfaults... this hack sucks */
++	/* FIXME: Is ^^^ still true? */
+ 	while (data->buffer_length > 0) {
+-		AVPacket packet;
+-		av_init_packet (&packet);
+-		packet.data = data->buffer;
+-		packet.size = data->buffer_length;
+-
+-		data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+-		bytes_read = avcodec_decode_audio3 (data->codecctx, (short *) data->read_out_buffer,
+-		                                    &data->read_out_buffer_size, &packet);
+-
+-		if (bytes_read < 0 || bytes_read > data->buffer_length) {
+-			XMMS_DBG ("Error decoding data!");
++		if (xmms_avcodec_internal_decode_some (data) < 0) {
+ 			return -1;
+ 		}
+-
+-		data->buffer_length -= bytes_read;
+-		g_memmove (data->buffer, data->buffer + bytes_read, data->buffer_length);
+ 	}
+ 
+ 	ret = xmms_xform_seek (xform, samples, whence, err);
+@@ -456,3 +369,131 @@ xmms_avcodec_translate_sample_format (enum AVSampleFormat av_sample_format)
+ 		return XMMS_SAMPLE_FORMAT_UNKNOWN;
+ 	}
+ }
++
++/*
++Read some data from our source of data to data->buffer, updating buffer_length
++and buffer_size as needed.
++
++Returns: on error: negative
++         on EOF: zero
++         otherwise: number of bytes read.
++*/
++static gint
++xmms_avcodec_internal_read_some (xmms_xform_t *xform,
++                                 xmms_avcodec_data_t *data,
++                                 xmms_error_t *error)
++{
++	gint bytes_read, read_total;
++
++	bytes_read = xmms_xform_read (xform,
++	                              (gchar *) (data->buffer + data->buffer_length),
++	                              data->buffer_size - data->buffer_length,
++	                              error);
++
++	if (bytes_read < 0) {
++		XMMS_DBG ("Error while reading data");
++		return bytes_read;
++	} else if (bytes_read == 0) {
++		XMMS_DBG ("EOF");
++		return 0;
++	}
++
++	read_total = bytes_read;
++
++	/* If we have a demuxer plugin, make sure we read the whole packet */
++	while (read_total == data->buffer_size && !data->no_demuxer) {
++		/* multiply the buffer size and try to read again */
++		data->buffer = g_realloc (data->buffer, data->buffer_size * 2);
++		bytes_read = xmms_xform_read (xform,
++		                              (gchar *) data->buffer +
++		                                data->buffer_size,
++		                              data->buffer_size,
++		                              error);
++		data->buffer_size *= 2;
++
++		if (bytes_read < 0) {
++			XMMS_DBG ("Error while reading data");
++			return bytes_read;
++		}
++
++		read_total += bytes_read;
++
++		if (read_total < data->buffer_size) {
++			/* finally double the buffer size for performance reasons, the
++			 * hotspot handling likes to fit two frames in the buffer */
++			data->buffer = g_realloc (data->buffer, data->buffer_size * 2);
++			data->buffer_size *= 2;
++			XMMS_DBG ("Reallocated avcodec internal buffer to be %d bytes",
++			          data->buffer_size);
++
++			break;
++		}
++	}
++
++	/* Update the buffer length */
++	data->buffer_length += read_total;
++
++	return read_total;
++}
++
++/*
++Decode some data from data->buffer[0..data->buffer_length-1] to
++data->read_out_buffer. Number of bytes in data->read_out_buffer
++is stored in data->read_out_buffer_size.
++
++Returns: on error: negative
++         on no new data produced: zero
++         otherwise: positive
++
++FIXME: data->buffer should be at least data->buffer_length +
++FF_INPUT_BUFFER_PADDING_SIZE long.
++*/
++static gint
++xmms_avcodec_internal_decode_some (xmms_avcodec_data_t *data)
++{
++	gint bytes_read = 0;
++	AVPacket packet;
++
++	av_init_packet (&packet);
++	packet.data = data->buffer;
++	packet.size = data->buffer_length;
++
++	data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
++	bytes_read = avcodec_decode_audio3 (data->codecctx,
++	                                    (short *) data->read_out_buffer,
++	                                    &data->read_out_buffer_size, &packet);
++
++	/* The DTS decoder of ffmpeg is buggy and always returns
++	 * the input buffer length, get frame length from header */
++	/* FIXME: Is ^^^^ still true? */
++	if (!strcmp (data->codec_id, "dca") && bytes_read > 0) {
++		bytes_read = ((int)data->buffer[5] << 12) |
++		             ((int)data->buffer[6] << 4) |
++		             ((int)data->buffer[7] >> 4);
++		bytes_read = (bytes_read & 0x3fff) + 1;
++	}
++
++	if (bytes_read < 0 || bytes_read > data->buffer_length) {
++		XMMS_DBG ("Error decoding data!");
++		return -1;
++	}
++
++	if (bytes_read < data->buffer_length) {
++		data->buffer_length -= bytes_read;
++		g_memmove (data->buffer,
++		           data->buffer + bytes_read,
++		           data->buffer_length);
++	} else {
++		data->buffer_length = 0;
++	}
++
++	return data->read_out_buffer_size;
++}
++
++static void
++xmms_avcodec_internal_append (xmms_avcodec_data_t *data)
++{
++	g_string_append_len (data->outbuf,
++	                     (gchar *) data->read_out_buffer,
++	                     data->read_out_buffer_size);
++}
diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p5.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p5.patch
new file mode 100644
index 000000000000..8ed5bb4a24a5
--- /dev/null
+++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p5.patch
@@ -0,0 +1,154 @@
+commit d44312fb14bde0ab47ee6de1b3fe7435d4a97c99
+Author: Erik Massop <e.massop@hccnet.nl>
+Date:   Sun Dec 22 20:01:18 2013 +0100
+
+    BUG(2572): Use avcodec_decode_audio4
+
+diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c
+index 266a607..a41a675 100644
+--- a/src/plugins/avcodec/avcodec.c
++++ b/src/plugins/avcodec/avcodec.c
+@@ -37,8 +37,7 @@ typedef struct {
+ 	guint buffer_size;
+ 	gboolean no_demuxer;
+ 
+-	gchar *read_out_buffer;
+-	gint read_out_buffer_size;
++	AVFrame *read_out_frame;
+ 
+ 	guint channels;
+ 	guint samplerate;
+@@ -125,7 +124,7 @@ xmms_avcodec_destroy (xmms_xform_t *xform)
+ 
+ 	avcodec_close (data->codecctx);
+ 	av_free (data->codecctx);
+-	av_free (data->read_out_buffer);
++	avcodec_free_frame (&data->read_out_frame);
+ 
+ 	g_string_free (data->outbuf, TRUE);
+ 	g_free (data->buffer);
+@@ -151,8 +150,7 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 	data->buffer_size = AVCODEC_BUFFER_SIZE;
+ 	data->codecctx = NULL;
+ 
+-	data->read_out_buffer = av_malloc (AVCODEC_MAX_AUDIO_FRAME_SIZE);
+-	data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
++	data->read_out_frame = avcodec_alloc_frame ();
+ 
+ 	xmms_xform_private_data_set (xform, data);
+ 
+@@ -233,6 +231,7 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 	data->codecctx->extradata_size = data->extradata_size;
+ 	data->codecctx->codec_id = codec->id;
+ 	data->codecctx->codec_type = codec->type;
++	data->codecctx->refcounted_frames = 0;
+ 
+ 	if (avcodec_open2 (data->codecctx, codec, NULL) < 0) {
+ 		XMMS_DBG ("Opening decoder '%s' failed", codec->name);
+@@ -279,8 +278,8 @@ err:
+ 	if (data->codecctx) {
+ 		av_free (data->codecctx);
+ 	}
+-	if (data->read_out_buffer) {
+-		av_free (data->read_out_buffer);
++	if (data->read_out_frame) {
++		avcodec_free_frame (&data->read_out_frame);
+ 	}
+ 	g_string_free (data->outbuf, TRUE);
+ 	g_free (data->extradata);
+@@ -365,17 +364,23 @@ xmms_avcodec_translate_sample_format (enum AVSampleFormat av_sample_format)
+ {
+ 	switch (av_sample_format) {
+ 	case AV_SAMPLE_FMT_U8:
++	case AV_SAMPLE_FMT_U8P:
+ 		return XMMS_SAMPLE_FORMAT_U8;
+ 	case AV_SAMPLE_FMT_S16:
++	case AV_SAMPLE_FMT_S16P:
+ 		return XMMS_SAMPLE_FORMAT_S16;
+ 	case AV_SAMPLE_FMT_S32:
++	case AV_SAMPLE_FMT_S32P:
+ 		return XMMS_SAMPLE_FORMAT_S32;
+ 	case AV_SAMPLE_FMT_FLT:
++	case AV_SAMPLE_FMT_FLTP:
+ 		return XMMS_SAMPLE_FORMAT_FLOAT;
+ 	case AV_SAMPLE_FMT_DBL:
++	case AV_SAMPLE_FMT_DBLP:
+ 		return XMMS_SAMPLE_FORMAT_DOUBLE;
+ 	default:
+-		XMMS_DBG ("AVSampleFormat (%i) not supported.", av_sample_format);
++		XMMS_DBG ("AVSampleFormat (%i: %s) not supported.", av_sample_format,
++		          av_get_sample_fmt_name (av_sample_format));
+ 		return XMMS_SAMPLE_FORMAT_UNKNOWN;
+ 	}
+ }
+@@ -448,8 +453,7 @@ xmms_avcodec_internal_read_some (xmms_xform_t *xform,
+ 
+ /*
+ Decode some data from data->buffer[0..data->buffer_length-1] to
+-data->read_out_buffer. Number of bytes in data->read_out_buffer
+-is stored in data->read_out_buffer_size.
++data->read_out_frame
+ 
+ Returns: on error: negative
+          on no new data produced: zero
+@@ -461,6 +465,7 @@ FF_INPUT_BUFFER_PADDING_SIZE long.
+ static gint
+ xmms_avcodec_internal_decode_some (xmms_avcodec_data_t *data)
+ {
++	int got_frame = 0;
+ 	gint bytes_read = 0;
+ 	AVPacket packet;
+ 
+@@ -468,10 +473,10 @@ xmms_avcodec_internal_decode_some (xmms_avcodec_data_t *data)
+ 	packet.data = data->buffer;
+ 	packet.size = data->buffer_length;
+ 
+-	data->read_out_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
+-	bytes_read = avcodec_decode_audio3 (data->codecctx,
+-	                                    (short *) data->read_out_buffer,
+-	                                    &data->read_out_buffer_size, &packet);
++	avcodec_get_frame_defaults (data->read_out_frame);
++
++	bytes_read = avcodec_decode_audio4 (
++		data->codecctx, data->read_out_frame, &got_frame, &packet);
+ 
+ 	/* The DTS decoder of ffmpeg is buggy and always returns
+ 	 * the input buffer length, get frame length from header */
+@@ -497,13 +502,33 @@ xmms_avcodec_internal_decode_some (xmms_avcodec_data_t *data)
+ 		data->buffer_length = 0;
+ 	}
+ 
+-	return data->read_out_buffer_size;
++	return got_frame ? 1 : 0;
+ }
+ 
+ static void
+ xmms_avcodec_internal_append (xmms_avcodec_data_t *data)
+ {
+-	g_string_append_len (data->outbuf,
+-	                     (gchar *) data->read_out_buffer,
+-	                     data->read_out_buffer_size);
++	enum AVSampleFormat fmt = (enum AVSampleFormat) data->read_out_frame->format;
++	int samples = data->read_out_frame->nb_samples;
++	int channels = data->codecctx->channels;
++	int bps = av_get_bytes_per_sample (fmt);
++
++	if (av_sample_fmt_is_planar (fmt)) {
++		/* Convert from planar to packed format */
++		gint i, j;
++
++		for (i = 0; i < samples; i++) {
++			for (j = 0; j < channels; j++) {
++				g_string_append_len (
++					data->outbuf,
++					(gchar *) (data->read_out_frame->extended_data[j] + i*bps),
++					bps
++				);
++			}
++		}
++	} else {
++		g_string_append_len (data->outbuf,
++		                     (gchar *) data->read_out_frame->extended_data[0],
++		                     samples * channels * bps);
++	}
+ }
diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p6.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p6.patch
new file mode 100644
index 000000000000..b1bc1c5d609b
--- /dev/null
+++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p6.patch
@@ -0,0 +1,106 @@
+commit fc66249e69f53eef709c5210546fdd92e1c89554
+Author: Erik Massop <e.massop@hccnet.nl>
+Date:   Sun Dec 22 23:04:08 2013 +0100
+
+    OTHER: Some compatibility with different avcodec versions
+
+diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c
+index a41a675..023833d 100644
+--- a/src/plugins/avcodec/avcodec.c
++++ b/src/plugins/avcodec/avcodec.c
+@@ -124,7 +124,7 @@ xmms_avcodec_destroy (xmms_xform_t *xform)
+ 
+ 	avcodec_close (data->codecctx);
+ 	av_free (data->codecctx);
+-	avcodec_free_frame (&data->read_out_frame);
++	av_frame_free (&data->read_out_frame);
+ 
+ 	g_string_free (data->outbuf, TRUE);
+ 	g_free (data->buffer);
+@@ -150,7 +150,7 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 	data->buffer_size = AVCODEC_BUFFER_SIZE;
+ 	data->codecctx = NULL;
+ 
+-	data->read_out_frame = avcodec_alloc_frame ();
++	data->read_out_frame = av_frame_alloc ();
+ 
+ 	xmms_xform_private_data_set (xform, data);
+ 
+@@ -231,7 +231,6 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 	data->codecctx->extradata_size = data->extradata_size;
+ 	data->codecctx->codec_id = codec->id;
+ 	data->codecctx->codec_type = codec->type;
+-	data->codecctx->refcounted_frames = 0;
+ 
+ 	if (avcodec_open2 (data->codecctx, codec, NULL) < 0) {
+ 		XMMS_DBG ("Opening decoder '%s' failed", codec->name);
+@@ -473,7 +472,8 @@ xmms_avcodec_internal_decode_some (xmms_avcodec_data_t *data)
+ 	packet.data = data->buffer;
+ 	packet.size = data->buffer_length;
+ 
+-	avcodec_get_frame_defaults (data->read_out_frame);
++	/* clear buffers and reset fields to defaults */
++	av_frame_unref (data->read_out_frame);
+ 
+ 	bytes_read = avcodec_decode_audio4 (
+ 		data->codecctx, data->read_out_frame, &got_frame, &packet);
+diff --git a/src/plugins/avcodec/avcodec_compat.h b/src/plugins/avcodec/avcodec_compat.h
+index 73ac2ab..e74b3f8 100644
+--- a/src/plugins/avcodec/avcodec_compat.h
++++ b/src/plugins/avcodec/avcodec_compat.h
+@@ -83,3 +83,17 @@ typedef struct AVPacket {
+ # define avcodec_open2(avctx, codec, options) \
+     avcodec_open(avctx, codec)
+ #endif
++
++/* Map avcodec_free_frame to av_freep if the former doesn't exist.
++ * (This is in versions earlier than 54.28.0 (libav) or 54.59.100 (ffmpeg)) */
++#if ! HAVE_AVCODEC_FREE_FRAME
++# define avcodec_free_frame av_freep
++#endif
++
++/* Map av_frame_alloc, av_frame_unref, av_frame_free into their
++ * deprecated versions in versions earlier than 55.28.1 */
++#if LIBAVCODEC_VERSION_INT < 0x371c01
++# define av_frame_alloc avcodec_alloc_frame
++# define av_frame_unref avcodec_get_frame_defaults
++# define av_frame_free avcodec_free_frame
++#endif
+diff --git a/src/plugins/avcodec/wscript b/src/plugins/avcodec/wscript
+index 03ba7d8..d367816 100644
+--- a/src/plugins/avcodec/wscript
++++ b/src/plugins/avcodec/wscript
+@@ -1,10 +1,33 @@
+ from waftools.plugin import plugin
+ 
++## Code fragments for configuration
++avcodec_free_frame_fragment = """
++#ifdef HAVE_LIBAVCODEC_AVCODEC_H
++# include "libavcodec/avcodec.h"
++#else
++# include "avcodec.h"
++#endif
++int main(void) {
++    AVFrame *frame;
++
++    avcodec_free_frame (&frame);
++
++    return 0;
++}
++"""
++
+ def plugin_configure(conf):
+     conf.check_cfg(package="libavcodec", uselib_store="avcodec",
+             args="--cflags --libs")
+     conf.check_cc(header_name="avcodec.h", uselib="avcodec", type="cshlib", mandatory=False)
+     conf.check_cc(header_name="libavcodec/avcodec.h", uselib="avcodec", type="cshlib", mandatory=False)
+ 
++    # non-mandatory function avcodec_free_frame since
++    # * ffmpeg: commit 46a3595, lavc 54.59.100, release 1.0
++    # * libav: commit a42aada, lavc 54.28.0, release 9
++    conf.check_cc(fragment=avcodec_free_frame_fragment, uselib="avcodec",
++                  uselib_store="avcodec_free_frame",
++                  msg="Checking for function avcodec_free_frame", mandatory=False)
++
+ configure, build = plugin('avcodec', configure=plugin_configure,
+                           libs=["avcodec"])
diff --git a/media-sound/xmms2/files/xmms2-0.8-audio4-p7.patch b/media-sound/xmms2/files/xmms2-0.8-audio4-p7.patch
new file mode 100644
index 000000000000..2d4bafd95a7d
--- /dev/null
+++ b/media-sound/xmms2/files/xmms2-0.8-audio4-p7.patch
@@ -0,0 +1,147 @@
+commit f460440b3f2a9db1a9deef3faf7dae6e626dd7b5
+Author: Erik Massop <e.massop@hccnet.nl>
+Date:   Sun Dec 22 23:34:12 2013 +0100
+
+    OTHER: Require avcodec_decode_audio4
+    
+    This was introduced in versions 53.40.0 (ffmpeg) and 53.25.0 (libav) of
+    avcodec. Hence we drop compatibility for earlier versions.
+
+diff --git a/src/plugins/avcodec/avcodec.c b/src/plugins/avcodec/avcodec.c
+index 023833d..6d0b667 100644
+--- a/src/plugins/avcodec/avcodec.c
++++ b/src/plugins/avcodec/avcodec.c
+@@ -154,7 +154,6 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 
+ 	xmms_xform_private_data_set (xform, data);
+ 
+-	avcodec_init ();
+ 	avcodec_register_all ();
+ 
+ 	mimetype = xmms_xform_indata_get_str (xform,
+@@ -225,7 +224,7 @@ xmms_avcodec_init (xmms_xform_t *xform)
+ 	data->codecctx->sample_rate = data->samplerate;
+ 	data->codecctx->channels = data->channels;
+ 	data->codecctx->bit_rate = data->bitrate;
+-	CONTEXT_BPS (data->codecctx) = data->samplebits;
++	data->codecctx->bits_per_coded_sample = data->samplebits;
+ 	data->codecctx->block_align = data->block_align;
+ 	data->codecctx->extradata = data->extradata;
+ 	data->codecctx->extradata_size = data->extradata_size;
+diff --git a/src/plugins/avcodec/avcodec_compat.h b/src/plugins/avcodec/avcodec_compat.h
+index e74b3f8..b50fa4b 100644
+--- a/src/plugins/avcodec/avcodec_compat.h
++++ b/src/plugins/avcodec/avcodec_compat.h
+@@ -21,69 +21,6 @@
+ # include "avcodec.h"
+ #endif
+ 
+-/* Map avcodec_decode_audio2 into the deprecated version
+- * avcodec_decode_audio in versions earlier than 51.28 */
+-#if LIBAVCODEC_VERSION_INT < 0x331c00
+-# define avcodec_decode_audio2 avcodec_decode_audio
+-#endif
+-
+-/* Handle API change that happened in libavcodec 52.00 */
+-#if LIBAVCODEC_VERSION_INT < 0x340000
+-# define CONTEXT_BPS(codecctx) (codecctx)->bits_per_sample
+-#else
+-# define CONTEXT_BPS(codecctx) (codecctx)->bits_per_coded_sample
+-#endif
+-
+-/* Before 52.23 AVPacket was defined in avformat.h which we
+- * do not want to depend on, so we define part of it manually
+- * on versions smaller than 52.23 (this makes me cry) */
+-#if LIBAVCODEC_VERSION_INT < 0x341700
+-typedef struct AVPacket {
+-        uint8_t *data;
+-        int size;
+-} AVPacket;
+-#endif
+-
+-/* Same thing as above for av_init_packet and version 52.25 */
+-#if LIBAVCODEC_VERSION_INT < 0x341900
+-# define av_init_packet(pkt) do { \
+-    (pkt)->data = NULL; \
+-    (pkt)->size = 0; \
+-  } while(0)
+-#endif
+-
+-/* Map avcodec_decode_audio3 into the deprecated version
+- * avcodec_decode_audio2 in versions earlier than 52.26 */
+-#if LIBAVCODEC_VERSION_INT < 0x341a00
+-# define avcodec_decode_audio3(avctx, samples, frame_size_ptr, avpkt) \
+-    avcodec_decode_audio2(avctx, samples, frame_size_ptr, \
+-                          (avpkt)->data, (avpkt)->size)
+-#endif
+-
+-/* Handle API change that happened in libavcodec 52.64 */
+-#if LIBAVCODEC_VERSION_INT < 0x344000
+-# define AVMEDIA_TYPE_AUDIO CODEC_TYPE_AUDIO
+-#endif
+-
+-/* Calling avcodec_init is not necessary after 53.04 (ffmpeg 0.9) */
+-#if LIBAVCODEC_VERSION_INT >= 0x350400
+-# define avcodec_init()
+-#endif
+-
+-/* Map avcodec_alloc_context3 into the deprecated version
+- * avcodec_alloc_context in versions earlier than 53.04 (ffmpeg 0.9) */
+-#if LIBAVCODEC_VERSION_INT < 0x350400
+-# define avcodec_alloc_context3(codec) \
+-    avcodec_alloc_context()
+-#endif
+-
+-/* Map avcodec_open2 into the deprecated version
+- * avcodec_open in versions earlier than 53.04 (ffmpeg 0.9) */
+-#if LIBAVCODEC_VERSION_INT < 0x350400
+-# define avcodec_open2(avctx, codec, options) \
+-    avcodec_open(avctx, codec)
+-#endif
+-
+ /* Map avcodec_free_frame to av_freep if the former doesn't exist.
+  * (This is in versions earlier than 54.28.0 (libav) or 54.59.100 (ffmpeg)) */
+ #if ! HAVE_AVCODEC_FREE_FRAME
+diff --git a/src/plugins/avcodec/wscript b/src/plugins/avcodec/wscript
+index d367816..00b182b 100644
+--- a/src/plugins/avcodec/wscript
++++ b/src/plugins/avcodec/wscript
+@@ -1,6 +1,24 @@
+ from waftools.plugin import plugin
+ 
+ ## Code fragments for configuration
++avcodec_decode_audio4_fragment = """
++#ifdef HAVE_LIBAVCODEC_AVCODEC_H
++# include "libavcodec/avcodec.h"
++#else
++# include "avcodec.h"
++#endif
++int main(void) {
++    AVCodecContext *ctx;
++    AVFrame *frame;
++    int got_frame;
++    AVPacket *pkt;
++
++    avcodec_decode_audio4 (ctx, frame, &got_frame, pkt);
++
++    return 0;
++}
++"""
++
+ avcodec_free_frame_fragment = """
+ #ifdef HAVE_LIBAVCODEC_AVCODEC_H
+ # include "libavcodec/avcodec.h"
+@@ -22,6 +40,13 @@ def plugin_configure(conf):
+     conf.check_cc(header_name="avcodec.h", uselib="avcodec", type="cshlib", mandatory=False)
+     conf.check_cc(header_name="libavcodec/avcodec.h", uselib="avcodec", type="cshlib", mandatory=False)
+ 
++    # mandatory function avcodec_decode_audio4 available since
++    # * ffmpeg: commit e4de716, lavc 53.40.0, release 0.9
++    # * libav: commit 0eea212, lavc 53.25.0, release 0.8
++    conf.check_cc(fragment=avcodec_decode_audio4_fragment, uselib="avcodec",
++                  uselib_store="avcodec_decode_audio4",
++                  msg="Checking for function avcodec_decode_audio4", mandatory=True)
++
+     # non-mandatory function avcodec_free_frame since
+     # * ffmpeg: commit 46a3595, lavc 54.59.100, release 1.0
+     # * libav: commit a42aada, lavc 54.28.0, release 9
diff --git a/media-sound/xmms2/xmms2-0.8-r2.ebuild b/media-sound/xmms2/xmms2-0.8-r2.ebuild
index 4661d2d4af6a..4500c37e84e4 100644
--- a/media-sound/xmms2/xmms2-0.8-r2.ebuild
+++ b/media-sound/xmms2/xmms2-0.8-r2.ebuild
@@ -116,6 +116,13 @@ src_prepare() {
 	epatch "${FILESDIR}/${P}"-ffmpeg2.patch #536232
 	epatch "${FILESDIR}/${P}"-cpython.patch
 	epatch "${FILESDIR}/${P}"-modpug.patch #536046
+	epatch "${FILESDIR}/${P}"-audio4-p1.patch
+	epatch "${FILESDIR}/${P}"-audio4-p2.patch
+	epatch "${FILESDIR}/${P}"-audio4-p3.patch
+	epatch "${FILESDIR}/${P}"-audio4-p4.patch
+	epatch "${FILESDIR}/${P}"-audio4-p5.patch
+	epatch "${FILESDIR}/${P}"-audio4-p6.patch
+	epatch "${FILESDIR}/${P}"-audio4-p7.patch
 
 	if has_version dev-libs/libcdio-paranoia; then
 		sed -i -e 's:cdio/cdda.h:cdio/paranoia/cdda.h:' src/plugins/cdda/cdda.c || die