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 */