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.

1 files changed, 0 insertions, 413 deletions

diff --git a/libmpcodecs/ad_ffmpeg.c b/libmpcodecs/ad_ffmpeg.c
deleted file mode 100644
index 2eacfadb8f..0000000000
--- a/libmpcodecs/ad_ffmpeg.c
+++ /dev/null
@@ -1,413 +0,0 @@
-/*
- * 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;
-}