Rename directories, move files (step 1 of 2) (does not compile)

Tis drops the silly lib prefixes, and attempts to organize the tree in
a more logical way. Make the top-level directory less cluttered as
well.

Renames the following directories:
    libaf -> audio/filter
    libao2 -> audio/out
    libvo -> video/out
    libmpdemux -> demux

Split libmpcodecs:
    vf* -> video/filter
    vd*, dec_video.* -> video/decode
    mp_image*, img_format*, ... -> video/
    ad*, dec_audio.* -> audio/decode

libaf/format.* is moved to audio/ - this is similar to how mp_image.*
is located in video/.

Move most top-level .c/.h files to core. (talloc.c/.h is left on top-
level, because it's external.) Park some of the more annoying files
in compat/. Some of these are relicts from the time mplayer used
ffmpeg internals.

sub/ is not split, because it's too much of a mess (subtitle code is
mixed with OSD display and rendering).

Maybe the organization of core is not ideal: it mixes playback core
(like mplayer.c) and utility helpers (like bstr.c/h). Should the need
arise, the playback core will be moved somewhere else, while core
contains all helper and common code.

10 files changed, 2244 insertions, 0 deletions

diff --git a/audio/decode/ad.c b/audio/decode/ad.c
new file mode 100644
index 0000000000..93cebed86d
--- /dev/null
+++ b/audio/decode/ad.c
@@ -0,0 +1,50 @@
+/*
+ * audio decoder interface
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include "config.h"
+
+#include "stream/stream.h"
+#include "libmpdemux/demuxer.h"
+#include "libmpdemux/stheader.h"
+#include "ad.h"
+
+/* Missed vorbis, mad, dshow */
+
+extern const ad_functions_t mpcodecs_ad_mpg123;
+extern const ad_functions_t mpcodecs_ad_ffmpeg;
+extern const ad_functions_t mpcodecs_ad_pcm;
+extern const ad_functions_t mpcodecs_ad_dvdpcm;
+extern const ad_functions_t mpcodecs_ad_spdif;
+
+const ad_functions_t * const mpcodecs_ad_drivers[] =
+{
+#ifdef CONFIG_MPG123
+  &mpcodecs_ad_mpg123,
+#endif
+  &mpcodecs_ad_ffmpeg,
+  &mpcodecs_ad_pcm,
+  &mpcodecs_ad_dvdpcm,
+  &mpcodecs_ad_spdif,
+  NULL
+};
diff --git a/audio/decode/ad.h b/audio/decode/ad.h
new file mode 100644
index 0000000000..5396085d04
--- /dev/null
+++ b/audio/decode/ad.h
@@ -0,0 +1,54 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPLAYER_AD_H
+#define MPLAYER_AD_H
+
+#include "mpc_info.h"
+#include "libmpdemux/stheader.h"
+
+typedef struct mp_codec_info ad_info_t;
+
+/* interface of video decoder drivers */
+typedef struct ad_functions
+{
+    const ad_info_t *info;
+    int (*preinit)(sh_audio_t *sh);
+    int (*init)(sh_audio_t *sh);
+    void (*uninit)(sh_audio_t *sh);
+    int (*control)(sh_audio_t *sh,int cmd,void* arg, ...);
+    int (*decode_audio)(sh_audio_t *sh, unsigned char *buffer, int minlen,
+                        int maxlen);
+} ad_functions_t;
+
+// NULL terminated array of all drivers
+extern const ad_functions_t * const mpcodecs_ad_drivers[];
+
+// fallback if ADCTRL_RESYNC not implemented: sh_audio->a_in_buffer_len=0;
+#define ADCTRL_RESYNC_STREAM 1   // resync, called after seeking
+
+// fallback if ADCTRL_SKIP not implemented: ds_fill_buffer(sh_audio->ds);
+#define ADCTRL_SKIP_FRAME 2      // skip block/frame, called while seeking
+
+// fallback if ADCTRL_QUERY_FORMAT not implemented: sh_audio->sample_format
+#define ADCTRL_QUERY_FORMAT 3    // test for availabilty of a format
+
+// fallback: use hw mixer in libao
+#define ADCTRL_SET_VOLUME 4      // not used at the moment
+
+#endif /* MPLAYER_AD_H */
diff --git a/audio/decode/ad_dvdpcm.c b/audio/decode/ad_dvdpcm.c
new file mode 100644
index 0000000000..41f6a1426d
--- /dev/null
+++ b/audio/decode/ad_dvdpcm.c
@@ -0,0 +1,162 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "ad_internal.h"
+
+static const ad_info_t info =
+{
+	"Uncompressed DVD/VOB LPCM audio decoder",
+	"dvdpcm",
+	"Nick Kurshev",
+	"A'rpi",
+	""
+};
+
+LIBAD_EXTERN(dvdpcm)
+
+static int init(sh_audio_t *sh)
+{
+/* DVD PCM Audio:*/
+    sh->i_bps = 0;
+    if(sh->codecdata_len==3){
+	// we have LPCM header:
+	unsigned char h=sh->codecdata[1];
+	sh->channels=1+(h&7);
+	switch((h>>4)&3){
+	case 0: sh->samplerate=48000;break;
+	case 1: sh->samplerate=96000;break;
+	case 2: sh->samplerate=44100;break;
+	case 3: sh->samplerate=32000;break;
+	}
+	switch ((h >> 6) & 3) {
+	  case 0:
+	    sh->sample_format = AF_FORMAT_S16_BE;
+	    sh->samplesize = 2;
+	    break;
+	  case 1:
+	    mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Samples of this format are needed to improve support. Please contact the developers.\n");
+	    sh->i_bps = sh->channels * sh->samplerate * 5 / 2;
+	  case 2:
+	    sh->sample_format = AF_FORMAT_S24_BE;
+	    sh->samplesize = 3;
+	    break;
+	  default:
+	    sh->sample_format = AF_FORMAT_S16_BE;
+	    sh->samplesize = 2;
+	}
+    } else {
+	// use defaults:
+	sh->channels=2;
+	sh->samplerate=48000;
+	sh->sample_format = AF_FORMAT_S16_BE;
+	sh->samplesize = 2;
+    }
+    if (!sh->i_bps)
+    sh->i_bps = sh->samplesize * sh->channels * sh->samplerate;
+    return 1;
+}
+
+static int preinit(sh_audio_t *sh)
+{
+  sh->audio_out_minsize=2048;
+  return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+}
+
+static int control(sh_audio_t *sh,int cmd,void* arg, ...)
+{
+  int skip;
+    switch(cmd)
+    {
+      case ADCTRL_SKIP_FRAME:
+	skip=sh->i_bps/16;
+	skip=skip&(~3);
+	demux_read_data(sh->ds,NULL,skip);
+	return CONTROL_TRUE;
+    }
+  return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
+{
+  int j,len;
+  if (sh_audio->samplesize == 3) {
+    if (((sh_audio->codecdata[1] >> 6) & 3) == 1) {
+      // 20 bit
+      // not sure if the "& 0xf0" and "<< 4" are the right way around
+      // can somebody clarify?
+      for (j = 0; j < minlen; j += 12) {
+        char tmp[10];
+        len = demux_read_data(sh_audio->ds, tmp, 10);
+        if (len < 10) break;
+        // first sample
+        buf[j + 0] = tmp[0];
+        buf[j + 1] = tmp[1];
+        buf[j + 2] = tmp[8] & 0xf0;
+        // second sample
+        buf[j + 3] = tmp[2];
+        buf[j + 4] = tmp[3];
+        buf[j + 5] = tmp[8] << 4;
+        // third sample
+        buf[j + 6] = tmp[4];
+        buf[j + 7] = tmp[5];
+        buf[j + 8] = tmp[9] & 0xf0;
+        // fourth sample
+        buf[j + 9] = tmp[6];
+        buf[j + 10] = tmp[7];
+        buf[j + 11] = tmp[9] << 4;
+      }
+      len = j;
+    } else {
+      // 24 bit
+      for (j = 0; j < minlen; j += 12) {
+        char tmp[12];
+        len = demux_read_data(sh_audio->ds, tmp, 12);
+        if (len < 12) break;
+        // first sample
+        buf[j + 0] = tmp[0];
+        buf[j + 1] = tmp[1];
+        buf[j + 2] = tmp[8];
+        // second sample
+        buf[j + 3] = tmp[2];
+        buf[j + 4] = tmp[3];
+        buf[j + 5] = tmp[9];
+        // third sample
+        buf[j + 6] = tmp[4];
+        buf[j + 7] = tmp[5];
+        buf[j + 8] = tmp[10];
+        // fourth sample
+        buf[j + 9] = tmp[6];
+        buf[j + 10] = tmp[7];
+        buf[j + 11] = tmp[11];
+      }
+      len = j;
+    }
+  } else
+  len=demux_read_data(sh_audio->ds,buf,(minlen+3)&(~3));
+  return len;
+}
diff --git a/audio/decode/ad_internal.h b/audio/decode/ad_internal.h
new file mode 100644
index 0000000000..4cffc95126
--- /dev/null
+++ b/audio/decode/ad_internal.h
@@ -0,0 +1,46 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPLAYER_AD_INTERNAL_H
+#define MPLAYER_AD_INTERNAL_H
+
+#include "codec-cfg.h"
+#include "libaf/format.h"
+
+#include "stream/stream.h"
+#include "libmpdemux/demuxer.h"
+#include "libmpdemux/stheader.h"
+
+#include "ad.h"
+
+static int init(sh_audio_t *sh);
+static int preinit(sh_audio_t *sh);
+static void uninit(sh_audio_t *sh);
+static int control(sh_audio_t *sh,int cmd,void* arg, ...);
+static int decode_audio(sh_audio_t *sh,unsigned char *buffer,int minlen,int maxlen);
+
+#define LIBAD_EXTERN(x) const ad_functions_t mpcodecs_ad_##x = {\
+	&info,\
+	preinit,\
+	init,\
+        uninit,\
+	control,\
+	decode_audio\
+};
+
+#endif /* MPLAYER_AD_INTERNAL_H */
diff --git a/audio/decode/ad_lavc.c b/audio/decode/ad_lavc.c
new file mode 100644
index 0000000000..2eacfadb8f
--- /dev/null
+++ b/audio/decode/ad_lavc.c
@@ -0,0 +1,413 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdbool.h>
+#include <assert.h>
+
+#include <libavcodec/avcodec.h>
+#include <libavutil/opt.h>
+
+#include "talloc.h"
+
+#include "config.h"
+#include "mp_msg.h"
+#include "options.h"
+
+#include "ad_internal.h"
+#include "libaf/reorder_ch.h"
+
+#include "mpbswap.h"
+
+static const ad_info_t info =
+{
+    "libavcodec audio decoders",
+    "ffmpeg",
+    "",
+    "",
+    "",
+    .print_name = "libavcodec",
+};
+
+LIBAD_EXTERN(ffmpeg)
+
+struct priv {
+    AVCodecContext *avctx;
+    AVFrame *avframe;
+    char *output;
+    char *output_packed; // used by deplanarize to store packed audio samples
+    int output_left;
+    int unitsize;
+    int previous_data_left;  // input demuxer packet data
+};
+
+static int preinit(sh_audio_t *sh)
+{
+    return 1;
+}
+
+/* Prefer playing audio with the samplerate given in container data
+ * if available, but take number the number of channels and sample format
+ * from the codec, since if the codec isn't using the correct values for
+ * those everything breaks anyway.
+ */
+static int setup_format(sh_audio_t *sh_audio,
+                        const AVCodecContext *lavc_context)
+{
+    int sample_format = sh_audio->sample_format;
+    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
+    case AV_SAMPLE_FMT_U8:  sample_format = AF_FORMAT_U8;       break;
+    case AV_SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE;   break;
+    case AV_SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE;   break;
+    case AV_SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
+    default:
+        mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
+        sample_format = AF_FORMAT_UNKNOWN;
+    }
+
+    bool broken_srate        = false;
+    int samplerate           = lavc_context->sample_rate;
+    int container_samplerate = sh_audio->container_out_samplerate;
+    if (!container_samplerate && sh_audio->wf)
+        container_samplerate = sh_audio->wf->nSamplesPerSec;
+    if (lavc_context->codec_id == CODEC_ID_AAC
+        && samplerate == 2 * container_samplerate)
+        broken_srate = true;
+    else if (container_samplerate)
+        samplerate = container_samplerate;
+
+    if (lavc_context->channels != sh_audio->channels ||
+        samplerate != sh_audio->samplerate ||
+        sample_format != sh_audio->sample_format) {
+        sh_audio->channels = lavc_context->channels;
+        sh_audio->samplerate = samplerate;
+        sh_audio->sample_format = sample_format;
+        sh_audio->samplesize = af_fmt2bits(sh_audio->sample_format) / 8;
+        if (broken_srate)
+            mp_msg(MSGT_DECAUDIO, MSGL_WARN,
+                   "Ignoring broken container sample rate for AAC with SBR\n");
+        return 1;
+    }
+    return 0;
+}
+
+static int init(sh_audio_t *sh_audio)
+{
+    struct MPOpts *opts = sh_audio->opts;
+    AVCodecContext *lavc_context;
+    AVCodec *lavc_codec;
+
+    if (sh_audio->codec->dll) {
+        lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll);
+        if (!lavc_codec) {
+            mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
+                    "Cannot find codec '%s' in libavcodec...\n",
+                    sh_audio->codec->dll);
+            return 0;
+        }
+    } else if (!sh_audio->libav_codec_id) {
+        mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "No Libav codec ID known. "
+                "Generic lavc decoder is not applicable.\n");
+        return 0;
+    } else {
+        lavc_codec = avcodec_find_decoder(sh_audio->libav_codec_id);
+        if (!lavc_codec) {
+            mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Libavcodec has no decoder "
+                   "for this codec\n");
+            return 0;
+        }
+    }
+
+    sh_audio->codecname = lavc_codec->long_name;
+    if (!sh_audio->codecname)
+        sh_audio->codecname = lavc_codec->name;
+
+    struct priv *ctx = talloc_zero(NULL, struct priv);
+    sh_audio->context = ctx;
+    lavc_context = avcodec_alloc_context3(lavc_codec);
+    ctx->avctx = lavc_context;
+    ctx->avframe = avcodec_alloc_frame();
+
+    // Always try to set - option only exists for AC3 at the moment
+    av_opt_set_double(lavc_context, "drc_scale", opts->drc_level,
+                      AV_OPT_SEARCH_CHILDREN);
+    lavc_context->sample_rate = sh_audio->samplerate;
+    lavc_context->bit_rate = sh_audio->i_bps * 8;
+    if (sh_audio->wf) {
+        lavc_context->channels = sh_audio->wf->nChannels;
+        lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
+        lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
+        lavc_context->block_align = sh_audio->wf->nBlockAlign;
+        lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample;
+    }
+    lavc_context->request_channels = opts->audio_output_channels;
+    lavc_context->codec_tag = sh_audio->format; //FOURCC
+    if (sh_audio->gsh->lavf_codec_tag)
+        lavc_context->codec_tag = sh_audio->gsh->lavf_codec_tag;
+    lavc_context->codec_type = AVMEDIA_TYPE_AUDIO;
+    lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
+
+    /* alloc extra data */
+    if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
+        lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
+        lavc_context->extradata_size = sh_audio->wf->cbSize;
+        memcpy(lavc_context->extradata, sh_audio->wf + 1,
+               lavc_context->extradata_size);
+    }
+
+    // for QDM2
+    if (sh_audio->codecdata_len && sh_audio->codecdata &&
+            !lavc_context->extradata) {
+        lavc_context->extradata = av_malloc(sh_audio->codecdata_len +
+                                            FF_INPUT_BUFFER_PADDING_SIZE);
+        lavc_context->extradata_size = sh_audio->codecdata_len;
+        memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
+               lavc_context->extradata_size);
+    }
+
+    /* open it */
+    if (avcodec_open2(lavc_context, lavc_codec, NULL) < 0) {
+        mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Could not open codec.\n");
+        uninit(sh_audio);
+        return 0;
+    }
+    mp_msg(MSGT_DECAUDIO, MSGL_V, "INFO: libavcodec \"%s\" init OK!\n",
+           lavc_codec->name);
+
+    if (sh_audio->format == 0x3343414D) {
+        // MACE 3:1
+        sh_audio->ds->ss_div = 2 * 3; // 1 samples/packet
+        sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
+    } else if (sh_audio->format == 0x3643414D) {
+        // MACE 6:1
+        sh_audio->ds->ss_div = 2 * 6; // 1 samples/packet
+        sh_audio->ds->ss_mul = 2 * sh_audio->wf->nChannels; // 1 byte*ch/packet
+    }
+
+    // Decode at least 1 byte:  (to get header filled)
+    for (int tries = 0;;) {
+        int x = decode_audio(sh_audio, sh_audio->a_buffer, 1,
+                             sh_audio->a_buffer_size);
+        if (x > 0) {
+            sh_audio->a_buffer_len = x;
+            break;
+        }
+        if (++tries >= 5) {
+            mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+                   "ad_ffmpeg: initial decode failed\n");
+            uninit(sh_audio);
+            return 0;
+        }
+    }
+
+    sh_audio->i_bps = lavc_context->bit_rate / 8;
+    if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
+        sh_audio->i_bps = sh_audio->wf->nAvgBytesPerSec;
+
+    switch (av_get_packed_sample_fmt(lavc_context->sample_fmt)) {
+    case AV_SAMPLE_FMT_U8:
+    case AV_SAMPLE_FMT_S16:
+    case AV_SAMPLE_FMT_S32:
+    case AV_SAMPLE_FMT_FLT:
+        break;
+    default:
+        uninit(sh_audio);
+        return 0;
+    }
+    return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+    sh->codecname = NULL;
+    struct priv *ctx = sh->context;
+    if (!ctx)
+        return;
+    AVCodecContext *lavc_context = ctx->avctx;
+
+    if (lavc_context) {
+        if (avcodec_close(lavc_context) < 0)
+            mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
+        av_freep(&lavc_context->extradata);
+        av_freep(&lavc_context);
+    }
+    avcodec_free_frame(&ctx->avframe);
+    talloc_free(ctx);
+    sh->context = NULL;
+}
+
+static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+{
+    struct priv *ctx = sh->context;
+    switch (cmd) {
+    case ADCTRL_RESYNC_STREAM:
+        avcodec_flush_buffers(ctx->avctx);
+        ds_clear_parser(sh->ds);
+        ctx->previous_data_left = 0;
+        ctx->output_left = 0;
+        return CONTROL_TRUE;
+    }
+    return CONTROL_UNKNOWN;
+}
+
+static av_always_inline void deplanarize(struct sh_audio *sh)
+{
+    struct priv *priv = sh->context;
+
+    size_t bps        = av_get_bytes_per_sample(priv->avctx->sample_fmt);
+    size_t nb_samples = priv->avframe->nb_samples;
+    size_t channels   = priv->avctx->channels;
+    size_t size       = bps * nb_samples * channels;
+
+    if (talloc_get_size(priv->output_packed) != size)
+        priv->output_packed =
+            talloc_realloc_size(priv, priv->output_packed, size);
+
+    size_t offset = 0;
+    unsigned char *output_ptr = priv->output_packed;
+    unsigned char **src = priv->avframe->data;
+
+    for (size_t s = 0; s < nb_samples; s++) {
+        for (size_t c = 0; c < channels; c++) {
+            memcpy(output_ptr, src[c] + offset, bps);
+            output_ptr += bps;
+        }
+        offset += bps;
+    }
+
+    priv->output = priv->output_packed;
+}
+
+static int decode_new_packet(struct sh_audio *sh)
+{
+    struct priv *priv = sh->context;
+    AVCodecContext *avctx = priv->avctx;
+    double pts = MP_NOPTS_VALUE;
+    int insize;
+    bool packet_already_used = priv->previous_data_left;
+    struct demux_packet *mpkt = ds_get_packet2(sh->ds,
+                                               priv->previous_data_left);
+    unsigned char *start;
+    if (!mpkt) {
+        assert(!priv->previous_data_left);
+        start = NULL;
+        insize = 0;
+        ds_parse(sh->ds, &start, &insize, pts, 0);
+        if (insize <= 0)
+            return -1;  // error or EOF
+    } else {
+        assert(mpkt->len >= priv->previous_data_left);
+        if (!priv->previous_data_left) {
+            priv->previous_data_left = mpkt->len;
+            pts = mpkt->pts;
+        }
+        insize = priv->previous_data_left;
+        start = mpkt->buffer + mpkt->len - priv->previous_data_left;
+        int consumed = ds_parse(sh->ds, &start, &insize, pts, 0);
+        priv->previous_data_left -= consumed;
+        priv->previous_data_left = FFMAX(priv->previous_data_left, 0);
+    }
+
+    AVPacket pkt;
+    av_init_packet(&pkt);
+    pkt.data = start;
+    pkt.size = insize;
+    if (mpkt && mpkt->avpacket) {
+        pkt.side_data = mpkt->avpacket->side_data;
+        pkt.side_data_elems = mpkt->avpacket->side_data_elems;
+    }
+    if (pts != MP_NOPTS_VALUE && !packet_already_used) {
+        sh->pts = pts;
+        sh->pts_bytes = 0;
+    }
+    int got_frame = 0;
+    int ret = avcodec_decode_audio4(avctx, priv->avframe, &got_frame, &pkt);
+    // LATM may need many packets to find mux info
+    if (ret == AVERROR(EAGAIN))
+        return 0;
+    if (ret < 0) {
+        mp_msg(MSGT_DECAUDIO, MSGL_V, "lavc_audio: error\n");
+        return -1;
+    }
+    // The "insize >= ret" test is sanity check against decoder overreads
+    if (!sh->parser && insize >= ret)
+        priv->previous_data_left = insize - ret;
+    if (!got_frame)
+        return 0;
+    uint64_t unitsize = (uint64_t)av_get_bytes_per_sample(avctx->sample_fmt) *
+                        avctx->channels;
+    if (unitsize > 100000)
+        abort();
+    priv->unitsize = unitsize;
+    uint64_t output_left = unitsize * priv->avframe->nb_samples;
+    if (output_left > 500000000)
+        abort();
+    priv->output_left = output_left;
+    if (av_sample_fmt_is_planar(avctx->sample_fmt) && avctx->channels > 1) {
+        deplanarize(sh);
+    } else {
+        priv->output = priv->avframe->data[0];
+    }
+    mp_dbg(MSGT_DECAUDIO, MSGL_DBG2, "Decoded %d -> %d  \n", insize,
+           priv->output_left);
+    return 0;
+}
+
+
+static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
+                        int maxlen)
+{
+    struct priv *priv = sh_audio->context;
+    AVCodecContext *avctx = priv->avctx;
+
+    int len = -1;
+    while (len < minlen) {
+        if (!priv->output_left) {
+            if (decode_new_packet(sh_audio) < 0)
+                break;
+            continue;
+        }
+        if (setup_format(sh_audio, avctx))
+            return len;
+        int size = (minlen - len + priv->unitsize - 1);
+        size -= size % priv->unitsize;
+        size = FFMIN(size, priv->output_left);
+        if (size > maxlen)
+            abort();
+        memcpy(buf, priv->output, size);
+        priv->output += size;
+        priv->output_left -= size;
+        if (avctx->channels >= 5) {
+            int samplesize = av_get_bytes_per_sample(avctx->sample_fmt);
+            reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
+                                AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
+                                avctx->channels,
+                                size / samplesize, samplesize);
+        }
+        if (len < 0)
+            len = size;
+        else
+            len += size;
+        buf += size;
+        maxlen -= size;
+        sh_audio->pts_bytes += size;
+    }
+    return len;
+}
diff --git a/audio/decode/ad_mpg123.c b/audio/decode/ad_mpg123.c
new file mode 100644
index 0000000000..a3ce2cdcf6
--- /dev/null
+++ b/audio/decode/ad_mpg123.c
@@ -0,0 +1,489 @@
+/*
+ * MPEG 1.0/2.0/2.5 audio layer I, II, III decoding with libmpg123
+ *
+ * Copyright (C) 2010-2012 Thomas Orgis <thomas@orgis.org>
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include "config.h"
+
+#include "ad_internal.h"
+
+static const ad_info_t info = {
+    "MPEG 1.0/2.0/2.5 layers I, II, III",
+    "mpg123",
+    "Thomas Orgis",
+    "mpg123.org",
+    "High-performance decoder using libmpg123."
+};
+
+LIBAD_EXTERN(mpg123)
+
+/* Reducing the ifdeffery to two main variants:
+ *   1. most compatible to any libmpg123 version
+ *   2. fastest variant with recent libmpg123 (>=1.14)
+ * Running variant 2 on older libmpg123 versions may work in
+ * principle, but is not supported.
+ * So, please leave the check for MPG123_API_VERSION there, m-kay?
+ */
+#include <mpg123.h>
+
+/* Enable faster mode of operation with newer libmpg123, avoiding
+ * unnecessary memcpy() calls. */
+#if (defined MPG123_API_VERSION) && (MPG123_API_VERSION >= 33)
+#define AD_MPG123_FRAMEWISE
+#endif
+
+/* Switch for updating bitrate info of VBR files. Not essential. */
+#define AD_MPG123_MEAN_BITRATE
+
+/* Funny thing, that. I assume I shall use it for selecting mpg123 channels.
+ * Please correct me if I guessed wrong. */
+extern int fakemono;
+
+struct ad_mpg123_context {
+    mpg123_handle *handle;
+#ifdef AD_MPG123_MEAN_BITRATE
+    /* Running mean for bit rate, stream length estimation. */
+    float mean_rate;
+    unsigned int mean_count;
+    /* Time delay for updates. */
+    short delay;
+#endif
+    /* If the stream is actually VBR. */
+    char vbr;
+};
+
+/* This initializes libmpg123 and prepares the handle, including funky
+ * parameters. */
+static int preinit(sh_audio_t *sh)
+{
+    int err, flag;
+    struct ad_mpg123_context *con;
+    /* Assumption: You always call preinit + init + uninit, on every file.
+     * But you stop at preinit in case it fails.
+     * If that is not true, one must ensure not to call mpg123_init / exit
+     * twice in a row. */
+    if (mpg123_init() != MPG123_OK)
+        return 0;
+
+    sh->context = malloc(sizeof(struct ad_mpg123_context));
+    con = sh->context;
+    /* Auto-choice of optimized decoder (first argument NULL). */
+    con->handle = mpg123_new(NULL, &err);
+    if (!con->handle)
+        goto bad_end;
+
+    /* Guessing here: Default value triggers forced upmix of mono to stereo. */
+    flag = fakemono == 0 ? MPG123_FORCE_STEREO :
+           fakemono == 1 ? MPG123_MONO_LEFT    :
+           fakemono == 2 ? MPG123_MONO_RIGHT   : 0;
+    if (mpg123_param(con->handle, MPG123_ADD_FLAGS, flag, 0.0) != MPG123_OK)
+        goto bad_end;
+
+    /* Basic settings.
+     * Don't spill messages, enable better resync with non-seekable streams.
+     * Give both flags individually without error checking to keep going with
+     * old libmpg123. Generally, it is not fatal if the flags are not
+     * honored */
+    mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_QUIET, 0.0);
+    /* Do not bail out on malformed streams at all.
+     * MPlayer does not handle a decoder throwing the towel on crappy input. */
+    mpg123_param(con->handle, MPG123_RESYNC_LIMIT, -1, 0.0);
+
+    /* Open decisions: Configure libmpg123 to force encoding (or stay open about
+     * library builds that support only float or int32 output), (de)configure
+     * gapless decoding (won't work with seeking in MPlayer, though).
+     * Don't forget to eventually enable ReplayGain/RVA support, too.
+     * Let's try to run with the default for now. */
+
+    /* That would produce floating point output.
+     * You can get 32 and 24 bit ints, even 8 bit via format matrix. */
+    /* mpg123_param(con->handle, MPG123_ADD_FLAGS, MPG123_FORCE_FLOAT, 0.); */
+
+    /* Example for RVA choice (available since libmpg123 1.0.0):
+    mpg123_param(con->handle, MPG123_RVA, MPG123_RVA_MIX, 0.0) */
+
+#ifdef AD_MPG123_FRAMEWISE
+    /* Prevent funky automatic resampling.
+     * This way, we can be sure that one frame will never produce
+     * more than 1152 stereo samples. */
+    mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_AUTO_RESAMPLE, 0.);
+#else
+    /* Older mpg123 is vulnerable to concatenated streams when gapless cutting
+     * is enabled (will only play the jingle of a badly constructed radio
+     * stream). The versions using framewise decoding are fine with that. */
+    mpg123_param(con->handle, MPG123_REMOVE_FLAGS, MPG123_GAPLESS, 0.);
+#endif
+
+    return 1;
+
+  bad_end:
+    if (!con->handle)
+        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
+               mpg123_plain_strerror(err));
+    else
+        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 preinit error: %s\n",
+               mpg123_strerror(con->handle));
+
+    if (con->handle)
+        mpg123_delete(con->handle);
+    mpg123_exit();
+    free(sh->context);
+    sh->context = NULL;
+    return 0;
+}
+
+/* Compute bitrate from frame size. */
+static int compute_bitrate(struct mpg123_frameinfo *i)
+{
+    static const int samples_per_frame[4][4] = {
+        {-1, 384, 1152, 1152},  /* MPEG 1 */
+        {-1, 384, 1152,  576},  /* MPEG 2 */
+        {-1, 384, 1152,  576},  /* MPEG 2.5 */
+        {-1,  -1,   -1,   -1},  /* Unknown */
+    };
+    return (int) ((i->framesize + 4) * 8 * i->rate * 0.001 /
+                  samples_per_frame[i->version][i->layer] + 0.5);
+}
+
+/* Opted against the header printout from old mp3lib, too much
+ * irrelevant info. This is modelled after the mpg123 app's
+ * standard output line.
+ * If more verbosity is demanded, one can add more detail and
+ * also throw in ID3v2 info which libmpg123 collects anyway. */
+static void print_header_compact(struct mpg123_frameinfo *i)
+{
+    static const char *smodes[5] = {
+        "stereo", "joint-stereo", "dual-channel", "mono", "invalid"
+    };
+    static const char *layers[4] = {
+        "Unknown", "I", "II", "III"
+    };
+    static const char *versions[4] = {
+        "1.0", "2.0", "2.5", "x.x"
+    };
+
+    mp_msg(MSGT_DECAUDIO, MSGL_V, "MPEG %s layer %s, ",
+           versions[i->version], layers[i->layer]);
+    switch (i->vbr) {
+    case MPG123_CBR:
+        if (i->bitrate)
+            mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s", i->bitrate);
+        else
+            mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s (free format)",
+                   compute_bitrate(i));
+        break;
+    case MPG123_VBR:
+        mp_msg(MSGT_DECAUDIO, MSGL_V, "VBR");
+        break;
+    case MPG123_ABR:
+        mp_msg(MSGT_DECAUDIO, MSGL_V, "%d kbit/s ABR", i->abr_rate);
+        break;
+    default:
+        mp_msg(MSGT_DECAUDIO, MSGL_V, "???");
+    }
+    mp_msg(MSGT_DECAUDIO, MSGL_V, ", %ld Hz %s\n", i->rate,
+           smodes[i->mode]);
+}
+
+/* This tries to extract a requested amount of decoded data.
+ * Even when you request 0 bytes, it will feed enough input so that
+ * the decoder _could_ have delivered something.
+ * Returns byte count >= 0, -1 on error.
+ *
+ * Thoughts on exact pts keeping:
+ * We have to assume that MPEG frames are cut in pieces by packet boundaries.
+ * Also, it might be possible that the first packet does not contain enough
+ * data to ensure initial stream sync... or re-sync on erroneous streams.
+ * So we need something robust to relate the decoded byte count to the correct
+ * time stamp. This is tricky, though. From the outside, you cannot tell if,
+ * after having fed two packets until the first output arrives, one should
+ * start counting from the first packet's pts or the second packet's.
+ * So, let's just count from the last fed package's pts. If the packets are
+ * exactly cut to MPEG frames, this will cause one frame mismatch in the
+ * beginning (when mpg123 peeks ahead for the following header), but will
+ * be corrected with the third frame already. One might add special code to
+ * not increment the base pts past the first packet's after a resync before
+ * the first decoded bytes arrived. */
+static int decode_a_bit(sh_audio_t *sh, unsigned char *buf, int count)
+{
+    int ret = MPG123_OK;
+    int got = 0;
+    struct ad_mpg123_context *con = sh->context;
+
+    /* There will be one MPG123_NEW_FORMAT message on first open.
+     * This will be handled in init(). */
+    do {
+        size_t got_now = 0;
+
+        /* Feed the decoder. This will only fire from the second round on. */
+        if (ret == MPG123_NEED_MORE) {
+            int incount;
+            double pts;
+            unsigned char *inbuf;
+            /* Feed more input data. */
+            incount = ds_get_packet_pts(sh->ds, &inbuf, &pts);
+            if (incount <= 0)
+                break;          /* Apparently that's it. EOF. */
+
+            /* Next bytes from that presentation time. */
+            if (pts != MP_NOPTS_VALUE) {
+                sh->pts       = pts;
+                sh->pts_bytes = 0;
+            }
+
+#ifdef AD_MPG123_FRAMEWISE
+            /* Have to use mpg123_feed() to avoid decoding here. */
+            ret = mpg123_feed(con->handle, inbuf, incount);
+#else
+            /* Do not use mpg123_feed(), added in later libmpg123 versions. */
+            ret = mpg123_decode(con->handle, inbuf, incount, NULL, 0, NULL);
+#endif
+            if (ret == MPG123_ERR)
+                break;
+        }
+        /* Theoretically, mpg123 could return MPG123_DONE, so be prepared.
+         * Should not happen in our usage, but it is a valid return code. */
+        else if (ret == MPG123_ERR || ret == MPG123_DONE)
+            break;
+
+        /* Try to decode a bit. This is the return value that counts
+         * for the loop condition. */
+#ifdef AD_MPG123_FRAMEWISE
+        if (!buf) { /* fake call just for feeding to get format */
+            ret = mpg123_getformat(con->handle, NULL, NULL, NULL);
+        } else { /* This is the decoding. One frame at a time. */
+            ret = mpg123_replace_buffer(con->handle, buf, count);
+            if (ret == MPG123_OK)
+                ret = mpg123_decode_frame(con->handle, NULL, NULL, &got_now);
+        }
+#else
+        ret = mpg123_decode(con->handle, NULL, 0, buf + got, count - got,
+                            &got_now);
+#endif
+
+        got += got_now;
+        sh->pts_bytes += got_now;
+
+#ifdef AD_MPG123_FRAMEWISE
+    } while (ret == MPG123_NEED_MORE || (got == 0 && count != 0));
+#else
+    } while (ret == MPG123_NEED_MORE || got < count);
+#endif
+
+    if (ret == MPG123_ERR) {
+        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 decoding failed: %s\n",
+               mpg123_strerror(con->handle));
+        mpg123_close(con->handle);
+        return -1;
+    }
+
+    return got;
+}
+
+/* Close, reopen stream. Feed data until we know the format of the stream.
+ * 1 on success, 0 on error */
+static int reopen_stream(sh_audio_t *sh)
+{
+    struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
+
+    mpg123_close(con->handle);
+    /* No resetting of the context:
+     * We do not want to loose the mean bitrate data. */
+
+    /* Open and make sure we have fed enough data to get stream properties. */
+    if (MPG123_OK == mpg123_open_feed(con->handle) &&
+        /* Feed data until mpg123 is ready (has found stream beginning). */
+        !decode_a_bit(sh, NULL, 0)) {
+        return 1;
+    } else {
+        mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+               "mpg123 failed to reopen stream: %s\n",
+               mpg123_strerror(con->handle));
+        mpg123_close(con->handle);
+        return 0;
+    }
+}
+
+/* Now we really start accessing some data and determining file format.
+ * Paranoia note: The mpg123_close() on errors is not really necessary,
+ * But it ensures that we don't accidentally continue decoding with a
+ * bad state (possibly interpreting the format badly or whatnot). */
+static int init(sh_audio_t *sh)
+{
+    long rate    = 0;
+    int channels = 0;
+    int encoding = 0;
+    mpg123_id3v2 *v2;
+    struct mpg123_frameinfo finfo;
+    struct ad_mpg123_context *con = sh->context;
+
+    /* We're open about any output format that libmpg123 will suggest.
+     * Note that a standard build will always default to 16 bit signed and
+     * the native sample rate of the file. */
+    if (MPG123_OK == mpg123_format_all(con->handle) &&
+        reopen_stream(sh) &&
+        MPG123_OK == mpg123_getformat(con->handle, &rate, &channels, &encoding) &&
+        /* Forbid the format to change later on. */
+        MPG123_OK == mpg123_format_none(con->handle) &&
+        MPG123_OK == mpg123_format(con->handle, rate, channels, encoding) &&
+        /* Get MPEG header info. */
+        MPG123_OK == mpg123_info(con->handle, &finfo) &&
+        /* Since we queried format, mpg123 should have read past ID3v2 tags.
+         * We need to decide if printing of UTF-8 encoded text info is wanted. */
+        MPG123_OK == mpg123_id3(con->handle, NULL, &v2)) {
+        /* If we are here, we passed all hurdles. Yay! Extract the info. */
+        print_header_compact(&finfo);
+        /* Do we want to print out the UTF-8 Id3v2 info?
+        if (v2)
+            print_id3v2(v2); */
+
+        /* Have kb/s, want B/s
+         * For VBR, the first frame will be a bad estimate. */
+        sh->i_bps = (finfo.bitrate ? finfo.bitrate : compute_bitrate(&finfo))
+                    * 1000 / 8;
+#ifdef AD_MPG123_MEAN_BITRATE
+        con->delay      = 1;
+        con->mean_rate  = 0.;
+        con->mean_count = 0;
+#endif
+        con->vbr = (finfo.vbr != MPG123_CBR);
+        sh->channels   = channels;
+        sh->samplerate = rate;
+        /* Without external force, mpg123 will always choose signed encoding,
+         * and non-16-bit only on builds that don't support it.
+         * Be reminded that it doesn't matter to the MPEG file what encoding
+         * is produced from it. */
+        switch (encoding) {
+        case MPG123_ENC_SIGNED_8:
+            sh->sample_format = AF_FORMAT_S8;
+            sh->samplesize    = 1;
+            break;
+        case MPG123_ENC_SIGNED_16:
+            sh->sample_format = AF_FORMAT_S16_NE;
+            sh->samplesize    = 2;
+            break;
+        /* To stay compatible with the oldest libmpg123 headers, do not rely
+         * on float and 32 bit encoding symbols being defined.
+         * Those formats came later */
+        case 0x1180: /* MPG123_ENC_SIGNED_32 */
+            sh->sample_format = AF_FORMAT_S32_NE;
+            sh->samplesize    = 4;
+            break;
+        case 0x200: /* MPG123_ENC_FLOAT_32 */
+            sh->sample_format = AF_FORMAT_FLOAT_NE;
+            sh->samplesize    = 4;
+            break;
+        default:
+            mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+                   "Bad encoding from mpg123: %i.\n", encoding);
+            mpg123_close(con->handle);
+            return 0;
+        }
+#ifdef AD_MPG123_FRAMEWISE
+        /* Going to decode directly to MPlayer's memory. It is important
+         * to have MPG123_AUTO_RESAMPLE disabled for the buffer size
+         * being an all-time limit. */
+        sh->audio_out_minsize = 1152 * 2 * sh->samplesize;
+#endif
+
+        return 1;
+    } else {
+        mp_msg(MSGT_DECAUDIO, MSGL_ERR, "mpg123 init error: %s\n",
+               mpg123_strerror(con->handle));
+        mpg123_close(con->handle);
+        return 0;
+    }
+}
+
+static void uninit(sh_audio_t *sh)
+{
+    struct ad_mpg123_context *con = (struct ad_mpg123_context*) sh->context;
+
+    mpg123_close(con->handle);
+    mpg123_delete(con->handle);
+    free(sh->context);
+    sh->context = NULL;
+    mpg123_exit();
+}
+
+#ifdef AD_MPG123_MEAN_BITRATE
+/* Update mean bitrate. This could be dropped if accurate time display
+ * on audio file playback is not desired. */
+static void update_info(sh_audio_t *sh)
+{
+    struct ad_mpg123_context *con = sh->context;
+    if (con->vbr && --con->delay < 1) {
+        struct mpg123_frameinfo finfo;
+        if (MPG123_OK == mpg123_info(con->handle, &finfo)) {
+            if (++con->mean_count > ((unsigned int) -1) / 2)
+                con->mean_count = ((unsigned int) -1) / 4;
+
+            /* Might not be numerically optimal, but works fine enough. */
+            con->mean_rate = ((con->mean_count - 1) * con->mean_rate +
+                              finfo.bitrate) / con->mean_count;
+            sh->i_bps = (int) (con->mean_rate * 1000 / 8);
+
+            con->delay = 10;
+        }
+    }
+}
+#endif
+
+static int decode_audio(sh_audio_t *sh, unsigned char *buf, int minlen,
+                        int maxlen)
+{
+    int bytes;
+
+    bytes = decode_a_bit(sh, buf, maxlen);
+    if (bytes == 0)
+        return -1;              /* EOF */
+
+#ifdef AD_MPG123_MEAN_BITRATE
+    update_info(sh);
+#endif
+    return bytes;
+}
+
+static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+{
+    switch (cmd) {
+    case ADCTRL_RESYNC_STREAM:
+        /* Close/reopen the stream for mpg123 to make sure it doesn't
+         * think that it still knows the exact stream position.
+         * Otherwise, we would have funny effects from the gapless code.
+         * Oh, and it helps to minimize artifacts from jumping in the stream. */
+        if (reopen_stream(sh)) {
+#ifdef AD_MPG123_MEAN_BITRATE
+            update_info(sh);
+#endif
+            return CONTROL_TRUE;
+        } else {
+            /* MPlayer ignores this case! It just keeps on decoding.
+             * So we have to make sure resync never fails ... */
+            mp_msg(MSGT_DECAUDIO, MSGL_ERR,
+                   "mpg123 cannot reopen stream for resync.\n");
+            return CONTROL_FALSE;
+        }
+        break;
+    }
+    return CONTROL_UNKNOWN;
+}
diff --git a/audio/decode/ad_pcm.c b/audio/decode/ad_pcm.c
new file mode 100644
index 0000000000..c265dfcd56
--- /dev/null
+++ b/audio/decode/ad_pcm.c
@@ -0,0 +1,220 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <stdbool.h>
+
+#include <libavutil/common.h>
+
+#include "talloc.h"
+#include "config.h"
+#include "ad_internal.h"
+#include "libaf/format.h"
+#include "libaf/reorder_ch.h"
+
+static const ad_info_t info = {
+    "Uncompressed PCM audio decoder",
+    "pcm",
+    "Nick Kurshev",
+    "A'rpi",
+    ""
+};
+
+struct ad_pcm_context {
+    unsigned char *buffer;
+    int buffer_pos;
+    int buffer_len;
+    int buffer_size;
+};
+
+LIBAD_EXTERN(pcm)
+
+static int init(sh_audio_t * sh_audio)
+{
+    WAVEFORMATEX *h = sh_audio->wf;
+    if (!h)
+        return 0;
+    sh_audio->i_bps = h->nAvgBytesPerSec;
+    sh_audio->channels = h->nChannels;
+    sh_audio->samplerate = h->nSamplesPerSec;
+    sh_audio->samplesize = (h->wBitsPerSample + 7) / 8;
+    sh_audio->sample_format = AF_FORMAT_S16_LE; // default
+    switch (sh_audio->format) { /* hardware formats: */
+    case 0x0:
+    case 0x1:                  // Microsoft PCM
+    case 0xfffe:               // Extended
+        switch (sh_audio->samplesize) {
+        case 1: sh_audio->sample_format = AF_FORMAT_U8;     break;
+        case 2: sh_audio->sample_format = AF_FORMAT_S16_LE; break;
+        case 3: sh_audio->sample_format = AF_FORMAT_S24_LE; break;
+        case 4: sh_audio->sample_format = AF_FORMAT_S32_LE; break;
+        }
+        break;
+    case 0x3:                  // IEEE float
+        sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
+        break;
+    case 0x6:  sh_audio->sample_format = AF_FORMAT_A_LAW;      break;
+    case 0x7:  sh_audio->sample_format = AF_FORMAT_MU_LAW;     break;
+    case 0x11: sh_audio->sample_format = AF_FORMAT_IMA_ADPCM;  break;
+    case 0x50: sh_audio->sample_format = AF_FORMAT_MPEG2;      break;
+/*    case 0x2000: sh_audio->sample_format=AFMT_AC3; */
+    case 0x20776172: // 'raw '
+        sh_audio->sample_format = AF_FORMAT_S16_BE;
+        if (sh_audio->samplesize == 1)
+            sh_audio->sample_format = AF_FORMAT_U8;
+        break;
+    case 0x736F7774: // 'twos'
+        sh_audio->sample_format = AF_FORMAT_S16_BE;
+        // intended fall-through
+    case 0x74776F73: // 'sowt'
+        if (sh_audio->samplesize == 1)
+            sh_audio->sample_format = AF_FORMAT_S8;
+        break;
+    case 0x32336c66: // 'fl32', bigendian float32
+    case 0x32334C46: // 'FL32', bigendian float32 in aiff
+        sh_audio->sample_format = AF_FORMAT_FLOAT_BE;
+        sh_audio->samplesize = 4;
+        break;
+    case 0x666c3332: // '23lf', little endian float32, MPlayer internal fourCC
+    case 0x6D63706C: // 'lpcm'
+        sh_audio->sample_format = AF_FORMAT_FLOAT_LE;
+        sh_audio->samplesize = 4;
+        break;
+/*    case 0x34366c66: // 'fl64', bigendian float64
+       sh_audio->sample_format=AF_FORMAT_FLOAT_BE;
+       sh_audio->samplesize=8;
+       break;
+    case 0x666c3634: // '46lf', little endian float64, MPlayer internal fourCC
+       sh_audio->sample_format=AF_FORMAT_FLOAT_LE;
+       sh_audio->samplesize=8;
+       break;*/
+    case 0x34326e69: // 'in24', bigendian int24
+        sh_audio->sample_format = AF_FORMAT_S24_BE;
+        sh_audio->samplesize = 3;
+        break;
+    case 0x696e3234: // '42ni', little endian int24, MPlayer internal fourCC
+        sh_audio->sample_format = AF_FORMAT_S24_LE;
+        sh_audio->samplesize = 3;
+        break;
+    case 0x32336e69: // 'in32', bigendian int32
+        sh_audio->sample_format = AF_FORMAT_S32_BE;
+        sh_audio->samplesize = 4;
+        break;
+    case 0x696e3332: // '23ni', little endian int32, MPlayer internal fourCC
+        sh_audio->sample_format = AF_FORMAT_S32_LE;
+        sh_audio->samplesize = 4;
+        break;
+    case MKTAG('M', 'P', 'a', 'f'):
+        sh_audio->sample_format = h->wFormatTag;
+        sh_audio->samplesize = (af_fmt2bits(sh_audio->sample_format) + 7) / 8;
+        break;
+    default:
+        if (sh_audio->samplesize != 2)
+            sh_audio->sample_format = AF_FORMAT_U8;
+    }
+    if (!sh_audio->samplesize)  // this would cause MPlayer to hang later
+        sh_audio->samplesize = 2;
+    sh_audio->context = talloc_zero(NULL, struct ad_pcm_context);
+    return 1;
+}
+
+static int preinit(sh_audio_t *sh)
+{
+    sh->audio_out_minsize = 2048;
+    return 1;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+    talloc_free(sh->context);
+}
+
+static int control(sh_audio_t *sh, int cmd, void *arg, ...)
+{
+    struct ad_pcm_context *ctx = sh->context;
+    int skip;
+    switch (cmd) {
+    case ADCTRL_RESYNC_STREAM:
+        ctx->buffer_len = 0;
+        return true;
+    case ADCTRL_SKIP_FRAME:
+        skip = sh->i_bps / 16;
+        skip = skip & (~3);
+        demux_read_data(sh->ds, NULL, skip);
+        return CONTROL_TRUE;
+    }
+    return CONTROL_UNKNOWN;
+}
+
+static int decode_audio(sh_audio_t *sh_audio, unsigned char *buf, int minlen,
+                        int maxlen)
+{
+    int unitsize = sh_audio->channels * sh_audio->samplesize;
+    minlen = (minlen + unitsize - 1) / unitsize * unitsize;
+    if (minlen > maxlen)
+        // if someone needs hundreds of channels adjust audio_out_minsize
+        // based on channels in preinit()
+        return -1;
+
+    int len = 0;
+    struct ad_pcm_context *ctx = sh_audio->context;
+    while (len < minlen) {
+        if (ctx->buffer_len - ctx->buffer_pos <= 0) {
+            double pts;
+            unsigned char *ptr;
+            int plen = ds_get_packet_pts(sh_audio->ds, &ptr, &pts);
+            if (plen < 0)
+                break;
+            if (ctx->buffer_size < plen) {
+                talloc_free(ctx->buffer);
+                ctx->buffer = talloc_size(ctx, plen);
+                ctx->buffer_size = plen;
+            }
+            memcpy(ctx->buffer, ptr, plen);
+            ctx->buffer_len = plen;
+            ctx->buffer_pos = 0;
+            if (pts != MP_NOPTS_VALUE) {
+                sh_audio->pts = pts;
+                sh_audio->pts_bytes = 0;
+            }
+        }
+        int from_stored = ctx->buffer_len - ctx->buffer_pos;
+        if (from_stored > minlen - len)
+            from_stored = minlen - len;
+        memcpy(buf + len, ctx->buffer + ctx->buffer_pos, from_stored);
+        ctx->buffer_pos += from_stored;
+        sh_audio->pts_bytes += from_stored;
+        len += from_stored;
+    }
+    if (len % unitsize) {
+        mp_msg(MSGT_DECAUDIO, MSGL_WARN, "[ad_pcm] discarding partial sample "
+               "at end\n");
+        len -= len % unitsize;
+    }
+    if (len == 0)
+        len = -1;               // The loop above only exits at error/EOF
+    if (len > 0 && sh_audio->channels >= 5) {
+        reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_WAVEEX_DEFAULT,
+                            AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
+                            sh_audio->channels, len / sh_audio->samplesize,
+                            sh_audio->samplesize);
+    }
+    return len;
+}
diff --git a/audio/decode/ad_spdif.c b/audio/decode/ad_spdif.c
new file mode 100644
index 0000000000..877bc99317
--- /dev/null
+++ b/audio/decode/ad_spdif.c
@@ -0,0 +1,310 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <string.h>
+
+#include <libavformat/avformat.h>
+#include <libavcodec/avcodec.h>
+#include <libavutil/opt.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "ad_internal.h"
+
+static const ad_info_t info = {
+    "libavformat/spdifenc audio pass-through decoder.",
+    "spdif",
+    "Naoya OYAMA",
+    "Naoya OYAMA",
+    "For ALL hardware decoders"
+};
+
+LIBAD_EXTERN(spdif)
+
+#define FILENAME_SPDIFENC "spdif"
+#define OUTBUF_SIZE 65536
+struct spdifContext {
+    AVFormatContext *lavf_ctx;
+    int              iec61937_packet_size;
+    int              out_buffer_len;
+    int              out_buffer_size;
+    uint8_t         *out_buffer;
+    uint8_t          pb_buffer[OUTBUF_SIZE];
+};
+
+static int read_packet(void *p, uint8_t *buf, int buf_size)
+{
+    // spdifenc does not use read callback.
+    return 0;
+}
+
+static int write_packet(void *p, uint8_t *buf, int buf_size)
+{
+    int len;
+    struct spdifContext *ctx = p;
+
+    len = FFMIN(buf_size, ctx->out_buffer_size -ctx->out_buffer_len);
+    memcpy(&ctx->out_buffer[ctx->out_buffer_len], buf, len);
+    ctx->out_buffer_len += len;
+    return len;
+}
+
+static int64_t seek(void *p, int64_t offset, int whence)
+{
+    // spdifenc does not use seek callback.
+    return 0;
+}
+
+static int preinit(sh_audio_t *sh)
+{
+    sh->samplesize = 2;
+    return 1;
+}
+
+static int init(sh_audio_t *sh)
+{
+    int i, x, in_size, srate, bps, *dtshd_rate;
+    unsigned char *start;
+    double pts;
+    static const struct {
+        const char *name; enum CodecID id;
+    } fmt_id_type[] = {
+        { "aac" , CODEC_ID_AAC    },
+        { "ac3" , CODEC_ID_AC3    },
+        { "dca" , CODEC_ID_DTS    },
+        { "eac3", CODEC_ID_EAC3   },
+        { "mpa" , CODEC_ID_MP3    },
+        { "thd" , CODEC_ID_TRUEHD },
+        { NULL  , 0 }
+    };
+    AVFormatContext     *lavf_ctx  = NULL;
+    AVStream            *stream    = NULL;
+    const AVOption      *opt       = NULL;
+    struct spdifContext *spdif_ctx = NULL;
+
+    spdif_ctx = av_mallocz(sizeof(*spdif_ctx));
+    if (!spdif_ctx)
+        goto fail;
+    spdif_ctx->lavf_ctx = avformat_alloc_context();
+    if (!spdif_ctx->lavf_ctx)
+        goto fail;
+
+    sh->context = spdif_ctx;
+    lavf_ctx    = spdif_ctx->lavf_ctx;
+
+    lavf_ctx->oformat = av_guess_format(FILENAME_SPDIFENC, NULL, NULL);
+    if (!lavf_ctx->oformat)
+        goto fail;
+    lavf_ctx->priv_data = av_mallocz(lavf_ctx->oformat->priv_data_size);
+    if (!lavf_ctx->priv_data)
+        goto fail;
+    lavf_ctx->pb = avio_alloc_context(spdif_ctx->pb_buffer, OUTBUF_SIZE, 1, spdif_ctx,
+                            read_packet, write_packet, seek);
+    if (!lavf_ctx->pb)
+        goto fail;
+    stream = avformat_new_stream(lavf_ctx, 0);
+    if (!stream)
+        goto fail;
+    lavf_ctx->duration   = AV_NOPTS_VALUE;
+    lavf_ctx->start_time = AV_NOPTS_VALUE;
+    for (i = 0; fmt_id_type[i].name; i++) {
+        if (!strcmp(sh->codec->dll, fmt_id_type[i].name)) {
+            lavf_ctx->streams[0]->codec->codec_id = fmt_id_type[i].id;
+            break;
+        }
+    }
+    lavf_ctx->raw_packet_buffer_remaining_size = RAW_PACKET_BUFFER_SIZE;
+    if (AVERROR_PATCHWELCOME == lavf_ctx->oformat->write_header(lavf_ctx)) {
+        mp_msg(MSGT_DECAUDIO,MSGL_INFO,
+               "This codec is not supported by spdifenc.\n");
+        goto fail;
+    }
+
+    // get sample_rate & bitrate from parser
+    bps = srate = 0;
+    x = ds_get_packet_pts(sh->ds, &start, &pts);
+    in_size = x;
+    if (x <= 0) {
+        pts = MP_NOPTS_VALUE;
+        x = 0;
+    }
+    ds_parse(sh->ds, &start, &x, pts, 0);
+    if (x == 0) { // not enough buffer
+        srate = 48000;    //fake value
+        bps   = 768000/8; //fake value
+    } else  if (sh->avctx) {
+        if (sh->avctx->sample_rate < 44100) {
+            mp_msg(MSGT_DECAUDIO,MSGL_INFO,
+                   "This stream sample_rate[%d Hz] may be broken. "
+                   "Force reset 48000Hz.\n",
+                   sh->avctx->sample_rate);
+            srate = 48000; //fake value
+        } else
+            srate = sh->avctx->sample_rate;
+        bps = sh->avctx->bit_rate/8;
+    }
+    sh->ds->buffer_pos -= in_size;
+
+    switch (lavf_ctx->streams[0]->codec->codec_id) {
+    case CODEC_ID_AAC:
+        spdif_ctx->iec61937_packet_size = 16384;
+        sh->sample_format               = AF_FORMAT_IEC61937_LE;
+        sh->samplerate                  = srate;
+        sh->channels                    = 2;
+        sh->i_bps                       = bps;
+        break;
+    case CODEC_ID_AC3:
+        spdif_ctx->iec61937_packet_size = 6144;
+        sh->sample_format               = AF_FORMAT_IEC61937_LE;
+        sh->samplerate                  = srate;
+        sh->channels                    = 2;
+        sh->i_bps                       = bps;
+        break;
+    case CODEC_ID_DTS: // FORCE USE DTS-HD
+        opt = av_opt_find(&lavf_ctx->oformat->priv_class,
+                          "dtshd_rate", NULL, 0, 0);
+        if (!opt)
+            goto fail;
+        dtshd_rate                      = (int*)(((uint8_t*)lavf_ctx->priv_data) +
+                                          opt->offset);
+        *dtshd_rate                     = 192000*4;
+        spdif_ctx->iec61937_packet_size = 32768;
+        sh->sample_format               = AF_FORMAT_IEC61937_LE;
+        sh->samplerate                  = 192000; // DTS core require 48000
+        sh->channels                    = 2*4;
+        sh->i_bps                       = bps;
+        break;
+    case CODEC_ID_EAC3:
+        spdif_ctx->iec61937_packet_size = 24576;
+        sh->sample_format               = AF_FORMAT_IEC61937_LE;
+        sh->samplerate                  = 192000;
+        sh->channels                    = 2;
+        sh->i_bps                       = bps;
+        break;
+    case CODEC_ID_MP3:
+        spdif_ctx->iec61937_packet_size = 4608;
+        sh->sample_format               = AF_FORMAT_MPEG2;
+        sh->samplerate                  = srate;
+        sh->channels                    = 2;
+        sh->i_bps                       = bps;
+        break;
+    case CODEC_ID_TRUEHD:
+        spdif_ctx->iec61937_packet_size = 61440;
+        sh->sample_format               = AF_FORMAT_IEC61937_LE;
+        sh->samplerate                  = 192000;
+        sh->channels                    = 8;
+        sh->i_bps                       = bps;
+        break;
+    default:
+        break;
+    }
+
+    return 1;
+
+fail:
+    uninit(sh);
+    return 0;
+}
+
+static int decode_audio(sh_audio_t *sh, unsigned char *buf,
+                        int minlen, int maxlen)
+{
+    struct spdifContext *spdif_ctx = sh->context;
+    AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
+    AVPacket            pkt;
+    double              pts;
+    int                 ret, in_size, consumed, x;
+    unsigned char       *start = NULL;
+
+    consumed = spdif_ctx->out_buffer_len  = 0;
+    spdif_ctx->out_buffer_size = maxlen;
+    spdif_ctx->out_buffer      = buf;
+    while (spdif_ctx->out_buffer_len + spdif_ctx->iec61937_packet_size < maxlen
+           && spdif_ctx->out_buffer_len < minlen) {
+        if (sh->ds->eof)
+            break;
+        x = ds_get_packet_pts(sh->ds, &start, &pts);
+        if (x <= 0) {
+            x = 0;
+            ds_parse(sh->ds, &start, &x, MP_NOPTS_VALUE, 0);
+            if (x == 0)
+                continue; // END_NOT_FOUND
+            in_size = x;
+        } else {
+            in_size = x;
+            consumed = ds_parse(sh->ds, &start, &x, pts, 0);
+            if (x == 0) {
+                mp_msg(MSGT_DECAUDIO,MSGL_V,
+                       "start[%p] in_size[%d] consumed[%d] x[%d].\n",
+                       start, in_size, consumed, x);
+                continue; // END_NOT_FOUND
+            }
+            sh->ds->buffer_pos -= in_size - consumed;
+        }
+        av_init_packet(&pkt);
+        pkt.data = start;
+        pkt.size = x;
+        mp_msg(MSGT_DECAUDIO,MSGL_V,
+               "start[%p] pkt.size[%d] in_size[%d] consumed[%d] x[%d].\n",
+               start, pkt.size, in_size, consumed, x);
+        if (pts != MP_NOPTS_VALUE) {
+            sh->pts       = pts;
+            sh->pts_bytes = 0;
+        }
+        ret = lavf_ctx->oformat->write_packet(lavf_ctx, &pkt);
+        if (ret < 0)
+            break;
+    }
+    sh->pts_bytes += spdif_ctx->out_buffer_len;
+    return spdif_ctx->out_buffer_len;
+}
+
+static int control(sh_audio_t *sh, int cmd, void* arg, ...)
+{
+    unsigned char *start;
+    double pts;
+
+    switch (cmd) {
+    case ADCTRL_RESYNC_STREAM:
+    case ADCTRL_SKIP_FRAME:
+        ds_get_packet_pts(sh->ds, &start, &pts);
+        return CONTROL_TRUE;
+    }
+    return CONTROL_UNKNOWN;
+}
+
+static void uninit(sh_audio_t *sh)
+{
+    struct spdifContext *spdif_ctx = sh->context;
+    AVFormatContext     *lavf_ctx  = spdif_ctx->lavf_ctx;
+
+    if (lavf_ctx) {
+        if (lavf_ctx->oformat)
+            lavf_ctx->oformat->write_trailer(lavf_ctx);
+        av_freep(&lavf_ctx->pb);
+        if (lavf_ctx->streams) {
+            av_freep(&lavf_ctx->streams[0]->codec);
+            av_freep(&lavf_ctx->streams[0]->info);
+            av_freep(&lavf_ctx->streams[0]);
+        }
+        av_freep(&lavf_ctx->streams);
+        av_freep(&lavf_ctx->priv_data);
+    }
+    av_freep(&lavf_ctx);
+    av_freep(&spdif_ctx);
+}
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c
new file mode 100644
index 0000000000..2602352e52
--- /dev/null
+++ b/audio/decode/dec_audio.c
@@ -0,0 +1,462 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <assert.h>
+
+#include "config.h"
+#include "mp_msg.h"
+#include "bstr.h"
+
+#include "stream/stream.h"
+#include "libmpdemux/demuxer.h"
+
+#include "codec-cfg.h"
+#include "libmpdemux/stheader.h"
+
+#include "dec_audio.h"
+#include "ad.h"
+#include "libaf/format.h"
+
+#include "libaf/af.h"
+
+int fakemono = 0;
+
+struct af_cfg af_cfg = { 1, NULL };	// Configuration for audio filters
+
+void afm_help(void)
+{
+    int i;
+    mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Available (compiled-in) audio codec families/drivers:\n");
+    mp_msg(MSGT_IDENTIFY, MSGL_INFO, "ID_AUDIO_DRIVERS\n");
+    mp_msg(MSGT_DECAUDIO, MSGL_INFO, "    afm:    info:  (comment)\n");
+    for (i = 0; mpcodecs_ad_drivers[i] != NULL; i++)
+	if (mpcodecs_ad_drivers[i]->info->comment
+	    && mpcodecs_ad_drivers[i]->info->comment[0])
+	    mp_msg(MSGT_DECAUDIO, MSGL_INFO, "%9s  %s (%s)\n",
+		   mpcodecs_ad_drivers[i]->info->short_name,
+		   mpcodecs_ad_drivers[i]->info->name,
+		   mpcodecs_ad_drivers[i]->info->comment);
+	else
+	    mp_msg(MSGT_DECAUDIO, MSGL_INFO, "%9s  %s\n",
+		   mpcodecs_ad_drivers[i]->info->short_name,
+		   mpcodecs_ad_drivers[i]->info->name);
+}
+
+static int init_audio_codec(sh_audio_t *sh_audio)
+{
+    assert(!sh_audio->initialized);
+    resync_audio_stream(sh_audio);
+    if ((af_cfg.force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT) {
+	int fmt = AF_FORMAT_FLOAT_NE;
+	if (sh_audio->ad_driver->control(sh_audio, ADCTRL_QUERY_FORMAT,
+					 &fmt) == CONTROL_TRUE) {
+	    sh_audio->sample_format = fmt;
+	    sh_audio->samplesize = 4;
+	}
+    }
+    sh_audio->audio_out_minsize = 8192; // default, preinit() may change it
+    if (!sh_audio->ad_driver->preinit(sh_audio)) {
+	mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "ADecoder preinit failed :(\n");
+	return 0;
+    }
+
+    /* allocate audio in buffer: */
+    if (sh_audio->audio_in_minsize > 0) {
+	sh_audio->a_in_buffer_size = sh_audio->audio_in_minsize;
+	mp_tmsg(MSGT_DECAUDIO, MSGL_V, "dec_audio: Allocating %d bytes for input buffer.\n",
+	       sh_audio->a_in_buffer_size);
+	sh_audio->a_in_buffer = av_mallocz(sh_audio->a_in_buffer_size);
+    }
+
+    const int base_size = 65536;
+    // At least 64 KiB plus rounding up to next decodable unit size
+    sh_audio->a_buffer_size = base_size + sh_audio->audio_out_minsize;
+
+    mp_tmsg(MSGT_DECAUDIO, MSGL_V, "dec_audio: Allocating %d + %d = %d bytes for output buffer.\n",
+	   sh_audio->audio_out_minsize, base_size, sh_audio->a_buffer_size);
+
+    sh_audio->a_buffer = av_mallocz(sh_audio->a_buffer_size);
+    if (!sh_audio->a_buffer)
+        abort();
+    sh_audio->a_buffer_len = 0;
+
+    if (!sh_audio->ad_driver->init(sh_audio)) {
+	mp_tmsg(MSGT_DECAUDIO, MSGL_V, "ADecoder init failed :(\n");
+	uninit_audio(sh_audio);	// free buffers
+	return 0;
+    }
+
+    sh_audio->initialized = 1;
+
+    if (!sh_audio->channels || !sh_audio->samplerate) {
+	mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Audio decoder did not specify "
+                "audio format!\n");
+	uninit_audio(sh_audio);	// free buffers
+	return 0;
+    }
+
+    if (!sh_audio->o_bps)
+	sh_audio->o_bps = sh_audio->channels * sh_audio->samplerate
+	                  * sh_audio->samplesize;
+    return 1;
+}
+
+static int init_audio(sh_audio_t *sh_audio, char *codecname, char *afm,
+		      int status, stringset_t *selected)
+{
+    int force = 0;
+    if (codecname && codecname[0] == '+') {
+	codecname = &codecname[1];
+	force = 1;
+    }
+    sh_audio->codec = NULL;
+    while (1) {
+	const ad_functions_t *mpadec;
+	sh_audio->ad_driver = 0;
+	if (!(sh_audio->codec = find_audio_codec(sh_audio->format,
+						 NULL,
+						 sh_audio->codec, force)))
+	    break;
+	// ok we found one codec
+	if (stringset_test(selected, sh_audio->codec->name))
+	    continue;	// already tried & failed
+	if (codecname && strcmp(sh_audio->codec->name, codecname))
+	    continue;	// -ac
+	if (afm && strcmp(sh_audio->codec->drv, afm))
+	    continue;	// afm doesn't match
+	if (!force && sh_audio->codec->status < status)
+	    continue;	// too unstable
+	stringset_add(selected, sh_audio->codec->name);	// tagging it
+	// ok, it matches all rules, let's find the driver!
+        int i;
+	for (i = 0; mpcodecs_ad_drivers[i] != NULL; i++)
+	    if (!strcmp(mpcodecs_ad_drivers[i]->info->short_name,
+		 sh_audio->codec->drv))
+		break;
+	mpadec = mpcodecs_ad_drivers[i];
+	if (!mpadec) {		// driver not available (==compiled in)
+	    mp_tmsg(MSGT_DECAUDIO, MSGL_ERR,
+		   "Requested audio codec family [%s] (afm=%s) not available.\nEnable it at compilation.\n",
+		   sh_audio->codec->name, sh_audio->codec->drv);
+	    continue;
+	}
+	// it's available, let's try to init!
+	// init()
+	mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Opening audio decoder: [%s] %s\n",
+	       mpadec->info->short_name, mpadec->info->name);
+	sh_audio->ad_driver = mpadec;
+	if (!init_audio_codec(sh_audio)) {
+            mp_tmsg(MSGT_DECAUDIO, MSGL_WARN, "Audio decoder init failed for "
+                    "codecs.conf entry \"%s\".\n", sh_audio->codec->name);
+	    continue;		// try next...
+	}
+	// Yeah! We got it!
+	return 1;
+    }
+    return 0;
+}
+
+int init_best_audio_codec(sh_audio_t *sh_audio, char **audio_codec_list,
+			  char **audio_fm_list)
+{
+    stringset_t selected;
+    char *ac_l_default[2] = { "", (char *) NULL };
+    // hack:
+    if (!audio_codec_list)
+	audio_codec_list = ac_l_default;
+    // Go through the codec.conf and find the best codec...
+    sh_audio->initialized = 0;
+    stringset_init(&selected);
+    while (!sh_audio->initialized && *audio_codec_list) {
+	char *audio_codec = *(audio_codec_list++);
+	if (audio_codec[0]) {
+	    if (audio_codec[0] == '-') {
+		// disable this codec:
+		stringset_add(&selected, audio_codec + 1);
+	    } else {
+		// forced codec by name:
+		mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Forced audio codec: %s\n",
+		       audio_codec);
+		init_audio(sh_audio, audio_codec, NULL, -1, &selected);
+	    }
+	} else {
+	    int status;
+	    // try in stability order: UNTESTED, WORKING, BUGGY.
+	    // never try CRASHING.
+	    if (audio_fm_list) {
+		char **fmlist = audio_fm_list;
+		// try first the preferred codec families:
+		while (!sh_audio->initialized && *fmlist) {
+		    char *audio_fm = *(fmlist++);
+		    mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Trying to force audio codec driver family %s...\n",
+			   audio_fm);
+		    for (status = CODECS_STATUS__MAX;
+			 status >= CODECS_STATUS__MIN; --status)
+			if (init_audio(sh_audio, NULL, audio_fm, status, &selected))
+			    break;
+		}
+	    }
+	    if (!sh_audio->initialized)
+		for (status = CODECS_STATUS__MAX; status >= CODECS_STATUS__MIN;
+		     --status)
+		    if (init_audio(sh_audio, NULL, NULL, status, &selected))
+			break;
+	}
+    }
+    stringset_free(&selected);
+
+    if (!sh_audio->initialized) {
+	mp_tmsg(MSGT_DECAUDIO, MSGL_ERR, "Cannot find codec for audio format 0x%X.\n",
+	       sh_audio->format);
+	return 0;   // failed
+    }
+
+    mp_tmsg(MSGT_DECAUDIO, MSGL_INFO, "Selected audio codec: %s [%s]\n",
+            sh_audio->codecname ? sh_audio->codecname : sh_audio->codec->info,
+            sh_audio->ad_driver->info->print_name ?
+            sh_audio->ad_driver->info->print_name :
+            sh_audio->ad_driver->info->short_name);
+    mp_tmsg(MSGT_DECAUDIO, MSGL_V,
+            "Audio codecs.conf entry: %s (%s)  afm: %s\n",
+	   sh_audio->codec->name, sh_audio->codec->info, sh_audio->codec->drv);
+    mp_msg(MSGT_DECAUDIO, MSGL_V,
+	   "AUDIO: %d Hz, %d ch, %s, %3.1f kbit/%3.2f%% (ratio: %d->%d)\n",
+	   sh_audio->samplerate, sh_audio->channels,
+	   af_fmt2str_short(sh_audio->sample_format),
+	   sh_audio->i_bps * 8 * 0.001,
+	   ((float) sh_audio->i_bps / sh_audio->o_bps) * 100.0,
+	   sh_audio->i_bps, sh_audio->o_bps);
+    mp_msg(MSGT_IDENTIFY, MSGL_INFO,
+	   "ID_AUDIO_BITRATE=%d\nID_AUDIO_RATE=%d\n" "ID_AUDIO_NCH=%d\n",
+	   sh_audio->i_bps * 8, sh_audio->samplerate, sh_audio->channels);
+
+    return 1;   // success
+}
+
+void uninit_audio(sh_audio_t *sh_audio)
+{
+    if (sh_audio->afilter) {
+	mp_msg(MSGT_DECAUDIO, MSGL_V, "Uninit audio filters...\n");
+	af_uninit(sh_audio->afilter);
+	free(sh_audio->afilter);
+	sh_audio->afilter = NULL;
+    }
+    if (sh_audio->initialized) {
+	mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Uninit audio: %s\n",
+	       sh_audio->codec->drv);
+	sh_audio->ad_driver->uninit(sh_audio);
+	sh_audio->initialized = 0;
+    }
+    av_freep(&sh_audio->a_buffer);
+    av_freep(&sh_audio->a_in_buffer);
+}
+
+
+int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
+		       int *out_samplerate, int *out_channels, int *out_format)
+{
+    struct af_stream *afs = sh_audio->afilter;
+    if (!afs) {
+	afs = calloc(1, sizeof(struct af_stream));
+	afs->opts = sh_audio->opts;
+    }
+    // input format: same as codec's output format:
+    afs->input.rate   = in_samplerate;
+    afs->input.nch    = sh_audio->channels;
+    afs->input.format = sh_audio->sample_format;
+    af_fix_parameters(&(afs->input));
+
+    // output format: same as ao driver's input format (if missing, fallback to input)
+    afs->output.rate   = *out_samplerate;
+    afs->output.nch    = *out_channels;
+    afs->output.format = *out_format;
+    af_fix_parameters(&(afs->output));
+
+    // filter config:
+    memcpy(&afs->cfg, &af_cfg, sizeof(struct af_cfg));
+
+    mp_tmsg(MSGT_DECAUDIO, MSGL_V, "Building audio filter chain for %dHz/%dch/%s -> %dHz/%dch/%s...\n",
+	   afs->input.rate, afs->input.nch,
+	   af_fmt2str_short(afs->input.format), afs->output.rate,
+	   afs->output.nch, af_fmt2str_short(afs->output.format));
+
+    // let's autoprobe it!
+    if (0 != af_init(afs)) {
+	sh_audio->afilter = NULL;
+	free(afs);
+	return 0;   // failed :(
+    }
+
+    *out_samplerate = afs->output.rate;
+    *out_channels = afs->output.nch;
+    *out_format = afs->output.format;
+
+    // ok!
+    sh_audio->afilter = (void *) afs;
+    return 1;
+}
+
+static void set_min_out_buffer_size(struct bstr *outbuf, int len)
+{
+    size_t oldlen = talloc_get_size(outbuf->start);
+    if (oldlen < len) {
+        assert(outbuf->start);  // talloc context should be already set
+	mp_msg(MSGT_DECAUDIO, MSGL_V, "Increasing filtered audio buffer size "
+	       "from %zd to %d\n", oldlen, len);
+        outbuf->start = talloc_realloc_size(NULL, outbuf->start, len);
+    }
+}
+
+static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
+{
+    assert(len-1 + sh->audio_out_minsize <= sh->a_buffer_size);
+
+    int error = 0;
+
+    // Decode more bytes if needed
+    int old_samplerate = sh->samplerate;
+    int old_channels = sh->channels;
+    int old_sample_format = sh->sample_format;
+    while (sh->a_buffer_len < len) {
+	unsigned char *buf = sh->a_buffer + sh->a_buffer_len;
+	int minlen = len - sh->a_buffer_len;
+	int maxlen = sh->a_buffer_size - sh->a_buffer_len;
+	int ret = sh->ad_driver->decode_audio(sh, buf, minlen, maxlen);
+	int format_change = sh->samplerate != old_samplerate
+                            || sh->channels != old_channels
+                            || sh->sample_format != old_sample_format;
+	if (ret <= 0 || format_change) {
+	    error = format_change ? -2 : -1;
+            // samples from format-changing call get discarded too
+	    len = sh->a_buffer_len;
+	    break;
+	}
+	sh->a_buffer_len += ret;
+    }
+
+    // Filter
+    struct mp_audio filter_input = {
+	.audio = sh->a_buffer,
+	.len = len,
+	.rate = sh->samplerate,
+	.nch = sh->channels,
+	.format = sh->sample_format
+    };
+    af_fix_parameters(&filter_input);
+    struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
+    if (!filter_output)
+	return -1;
+    set_min_out_buffer_size(outbuf, outbuf->len + filter_output->len);
+    memcpy(outbuf->start + outbuf->len, filter_output->audio,
+           filter_output->len);
+    outbuf->len += filter_output->len;
+
+    // remove processed data from decoder buffer:
+    sh->a_buffer_len -= len;
+    memmove(sh->a_buffer, sh->a_buffer + len, sh->a_buffer_len);
+
+    return error;
+}
+
+/* Try to get at least minlen decoded+filtered bytes in outbuf
+ * (total length including possible existing data).
+ * Return 0 on success, -1 on error/EOF (not distinguished).
+ * In the former case outbuf->len is always >= minlen on return.
+ * In case of EOF/error it might or might not be.
+ * Outbuf.start must be talloc-allocated, and will be reallocated
+ * if needed to fit all filter output. */
+int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen)
+{
+    // Indicates that a filter seems to be buffering large amounts of data
+    int huge_filter_buffer = 0;
+    // Decoded audio must be cut at boundaries of this many bytes
+    int unitsize = sh_audio->channels * sh_audio->samplesize * 16;
+
+    /* Filter output size will be about filter_multiplier times input size.
+     * If some filter buffers audio in big blocks this might only hold
+     * as average over time. */
+    double filter_multiplier = af_calc_filter_multiplier(sh_audio->afilter);
+
+    /* If the decoder set audio_out_minsize then it can do the equivalent of
+     * "while (output_len < target_len) output_len += audio_out_minsize;",
+     * so we must guarantee there is at least audio_out_minsize-1 bytes
+     * more space in the output buffer than the minimum length we try to
+     * decode. */
+    int max_decode_len = sh_audio->a_buffer_size - sh_audio->audio_out_minsize;
+    if (!unitsize)
+        return -1;
+    max_decode_len -= max_decode_len % unitsize;
+
+    while (outbuf->len < minlen) {
+	int declen = (minlen - outbuf->len) / filter_multiplier
+	    + (unitsize << 5); // some extra for possible filter buffering
+	if (huge_filter_buffer)
+	/* Some filter must be doing significant buffering if the estimated
+	 * input length didn't produce enough output from filters.
+	 * Feed the filters 2k bytes at a time until we have enough output.
+	 * Very small amounts could make filtering inefficient while large
+	 * amounts can make MPlayer demux the file unnecessarily far ahead
+	 * to get audio data and buffer video frames in memory while doing
+	 * so. However the performance impact of either is probably not too
+	 * significant as long as the value is not completely insane. */
+	    declen = 2000;
+	declen -= declen % unitsize;
+	if (declen > max_decode_len)
+	    declen = max_decode_len;
+	else
+	    /* if this iteration does not fill buffer, we must have lots
+	     * of buffering in filters */
+	    huge_filter_buffer = 1;
+	int res = filter_n_bytes(sh_audio, outbuf, declen);
+	if (res < 0)
+	    return res;
+    }
+    return 0;
+}
+
+void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte)
+{
+    set_min_out_buffer_size(outbuf, outbuf->len + count);
+    memmove(outbuf->start + count, outbuf->start, outbuf->len);
+    memset(outbuf->start, byte, count);
+    outbuf->len += count;
+}
+
+
+void resync_audio_stream(sh_audio_t *sh_audio)
+{
+    sh_audio->a_in_buffer_len = 0;	// clear audio input buffer
+    sh_audio->pts = MP_NOPTS_VALUE;
+    if (!sh_audio->initialized)
+	return;
+    sh_audio->ad_driver->control(sh_audio, ADCTRL_RESYNC_STREAM, NULL);
+}
+
+void skip_audio_frame(sh_audio_t *sh_audio)
+{
+    if (!sh_audio->initialized)
+	return;
+    if (sh_audio->ad_driver->control(sh_audio, ADCTRL_SKIP_FRAME, NULL) ==
+	CONTROL_TRUE)
+	return;
+    // default skip code:
+    ds_fill_buffer(sh_audio->ds);	// skip block
+}
diff --git a/audio/decode/dec_audio.h b/audio/decode/dec_audio.h
new file mode 100644
index 0000000000..0d4baf0666
--- /dev/null
+++ b/audio/decode/dec_audio.h
@@ -0,0 +1,38 @@
+/*
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPLAYER_DEC_AUDIO_H
+#define MPLAYER_DEC_AUDIO_H
+
+#include "libmpdemux/stheader.h"
+
+struct bstr;
+
+// dec_audio.c:
+void afm_help(void);
+int init_best_audio_codec(sh_audio_t *sh_audio, char** audio_codec_list, char** audio_fm_list);
+int decode_audio(sh_audio_t *sh_audio, struct bstr *outbuf, int minlen);
+void decode_audio_prepend_bytes(struct bstr *outbuf, int count, int byte);
+void resync_audio_stream(sh_audio_t *sh_audio);
+void skip_audio_frame(sh_audio_t *sh_audio);
+void uninit_audio(sh_audio_t *sh_audio);
+
+int init_audio_filters(sh_audio_t *sh_audio, int in_samplerate,
+                       int *out_samplerate, int *out_channels, int *out_format);
+
+#endif /* MPLAYER_DEC_AUDIO_H */