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, 349 deletions

diff --git a/libaf/af.h b/libaf/af.h
deleted file mode 100644
index edce49a978..0000000000
--- a/libaf/af.h
+++ /dev/null
@@ -1,349 +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.
- */
-
-#ifndef MPLAYER_AF_H
-#define MPLAYER_AF_H
-
-#include <stdio.h>
-
-#include "config.h"
-
-#include "options.h"
-#include "libaf/format.h"
-#include "control.h"
-#include "cpudetect.h"
-#include "mp_msg.h"
-
-struct af_instance;
-
-// Number of channels
-#ifndef AF_NCH
-#define AF_NCH 8
-#endif
-
-// Audio data chunk
-struct mp_audio {
-    void *audio; // data buffer
-    int len;    // buffer length
-    int rate;   // sample rate
-    int nch;    // number of channels
-    int format; // format
-    int bps;    // bytes per sample
-};
-
-
-// Flags used for defining the behavior of an audio filter
-#define AF_FLAGS_REENTRANT      0x00000000
-#define AF_FLAGS_NOT_REENTRANT  0x00000001
-
-/* Audio filter information not specific for current instance, but for
-   a specific filter */
-struct af_info {
-    const char *info;
-    const char *name;
-    const char *author;
-    const char *comment;
-    const int flags;
-    int (*open)(struct af_instance *vf);
-};
-
-// Linked list of audio filters
-struct af_instance {
-    struct af_info *info;
-    int (*control)(struct af_instance *af, int cmd, void *arg);
-    void (*uninit)(struct af_instance *af);
-    struct mp_audio * (*play)(struct af_instance *af, struct mp_audio *data);
-    void *setup;  // setup data for this specific instance and filter
-    struct mp_audio *data; // configuration for outgoing data stream
-    struct af_instance *next;
-    struct af_instance *prev;
-    double delay; /* Delay caused by the filter, in units of bytes read without
-                   * corresponding output */
-    double mul; /* length multiplier: how much does this instance change
-                   the length of the buffer. */
-};
-
-// Initialization flags
-extern int *af_cpu_speed;
-
-#define AF_INIT_AUTO            0x00000000
-#define AF_INIT_SLOW            0x00000001
-#define AF_INIT_FAST            0x00000002
-#define AF_INIT_FORCE           0x00000003
-#define AF_INIT_TYPE_MASK       0x00000003
-
-#define AF_INIT_INT             0x00000000
-#define AF_INIT_FLOAT           0x00000004
-#define AF_INIT_FORMAT_MASK     0x00000004
-
-// Default init type
-#ifndef AF_INIT_TYPE
-#define AF_INIT_TYPE (af_cpu_speed ? *af_cpu_speed : AF_INIT_SLOW)
-#endif
-
-// Configuration switches
-struct af_cfg {
-    int force;  // Initialization type
-    char **list; /* list of names of filters that are added to filter
-                    list during first initialization of stream */
-};
-
-// Current audio stream
-struct af_stream {
-    // The first and last filter in the list
-    struct af_instance *first;
-    struct af_instance *last;
-    // Storage for input and output data formats
-    struct mp_audio input;
-    struct mp_audio output;
-    // Configuration for this stream
-    struct af_cfg cfg;
-    struct MPOpts *opts;
-};
-
-/*********************************************
-   // Return values
- */
-
-#define AF_DETACH   2
-#define AF_OK       1
-#define AF_TRUE     1
-#define AF_FALSE    0
-#define AF_UNKNOWN -1
-#define AF_ERROR   -2
-#define AF_FATAL   -3
-
-
-
-/*********************************************
-   // Export functions
- */
-
-/**
- * \defgroup af_chain Audio filter chain functions
- * \{
- * \param s filter chain
- */
-
-/**
- * \brief Initialize the stream "s".
- * \return 0 on success, -1 on failure
- *
- * This function creates a new filter list if necessary, according
- * to the values set in input and output. Input and output should contain
- * the format of the current movie and the format of the preferred output
- * respectively.
- * Filters to convert to the preferred output format are inserted
- * automatically, except when they are set to 0.
- * The function is reentrant i.e. if called with an already initialized
- * stream the stream will be reinitialized.
- */
-int af_init(struct af_stream *s);
-
-/**
- * \brief Uninit and remove all filters from audio filter chain
- */
-void af_uninit(struct af_stream *s);
-
-/**
- * \brief  Reinit the filter list from the given filter on downwards
- * \param  Filter instance to begin the reinit from
- * \return AF_OK on success or AF_ERROR on failure
- */
-int af_reinit(struct af_stream *s, struct af_instance *af);
-
-/**
- * \brief This function adds the filter "name" to the stream s.
- * \param name name of filter to add
- * \return pointer to the new filter, NULL if insert failed
- *
- * The filter will be inserted somewhere nice in the
- * list of filters (i.e. at the beginning unless the
- * first filter is the format filter (why??).
- */
-struct af_instance *af_add(struct af_stream *s, char *name);
-
-/**
- * \brief Uninit and remove the filter "af"
- * \param af filter to remove
- */
-void af_remove(struct af_stream *s, struct af_instance *af);
-
-/**
- * \brief find filter in chain by name
- * \param name name of the filter to find
- * \return first filter with right name or NULL if not found
- *
- * This function is used for finding already initialized filters
- */
-struct af_instance *af_get(struct af_stream *s, char *name);
-
-/**
- * \brief filter data chunk through the filters in the list
- * \param data data to play
- * \return resulting data
- * \ingroup af_chain
- */
-struct mp_audio *af_play(struct af_stream *s, struct mp_audio *data);
-
-/**
- * \brief send control to all filters, starting with the last until
- *        one accepts the command with AF_OK.
- * \param cmd filter control command
- * \param arg argument for filter command
- * \return the accepting filter or NULL if none was found
- */
-struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg);
-
-/**
- * \brief calculate average ratio of filter output lenth to input length
- * \return the ratio
- */
-double af_calc_filter_multiplier(struct af_stream *s);
-
-/**
- * \brief Calculate the total delay caused by the filters
- * \return delay in bytes of "missing" output
- */
-double af_calc_delay(struct af_stream *s);
-
-/** \} */ // end of af_chain group
-
-// Helper functions and macros used inside the audio filters
-
-/**
- * \defgroup af_filter Audio filter helper functions
- * \{
- */
-
-/* Helper function called by the macro with the same name only to be
-   called from inside filters */
-int af_resize_local_buffer(struct af_instance *af, struct mp_audio *data);
-
-/* Helper function used to calculate the exact buffer length needed
-   when buffers are resized. The returned length is >= than what is
-   needed */
-int af_lencalc(double mul, struct mp_audio *data);
-
-/**
- * \brief convert dB to gain value
- * \param n number of values to convert
- * \param in [in] values in dB, <= -200 will become 0 gain
- * \param out [out] gain values
- * \param k input values are divided by this
- * \param mi minimum dB value, input will be clamped to this
- * \param ma maximum dB value, input will be clamped to this
- * \return AF_ERROR on error, AF_OK otherwise
- */
-int af_from_dB(int n, float *in, float *out, float k, float mi, float ma);
-
-/**
- * \brief convert gain value to dB
- * \param n number of values to convert
- * \param in [in] gain values, 0 wil become -200 dB
- * \param out [out] values in dB
- * \param k output values will be multiplied by this
- * \return AF_ERROR on error, AF_OK otherwise
- */
-int af_to_dB(int n, float *in, float *out, float k);
-
-/**
- * \brief convert milliseconds to sample time
- * \param n number of values to convert
- * \param in [in] values in milliseconds
- * \param out [out] sample time values
- * \param rate sample rate
- * \param mi minimum ms value, input will be clamped to this
- * \param ma maximum ms value, input will be clamped to this
- * \return AF_ERROR on error, AF_OK otherwise
- */
-int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
-
-/**
- * \brief convert sample time to milliseconds
- * \param n number of values to convert
- * \param in [in] sample time values
- * \param out [out] values in milliseconds
- * \param rate sample rate
- * \return AF_ERROR on error, AF_OK otherwise
- */
-int af_to_ms(int n, int *in, float *out, int rate);
-
-/**
- * \brief test if output format matches
- * \param af audio filter
- * \param out needed format, will be overwritten by available
- *            format if they do not match
- * \return AF_FALSE if formats do not match, AF_OK if they match
- *
- * compares the format, bps, rate and nch values of af->data with out
- */
-int af_test_output(struct af_instance *af, struct mp_audio *out);
-
-/**
- * \brief soft clipping function using sin()
- * \param a input value
- * \return clipped value
- */
-float af_softclip(float a);
-
-/** \} */ // end of af_filter group, but more functions of this group below
-
-/** Print a list of all available audio filters */
-void af_help(void);
-
-/**
- * \brief fill the missing parameters in the struct mp_audio structure
- * \param data structure to fill
- * \ingroup af_filter
- *
- * Currently only sets bps based on format
- */
-void af_fix_parameters(struct mp_audio *data);
-
-/** Memory reallocation macro: if a local buffer is used (i.e. if the
-   filter doesn't operate on the incoming buffer this macro must be
-   called to ensure the buffer is big enough.
- * \ingroup af_filter
- */
-#define RESIZE_LOCAL_BUFFER(a, d) \
-    ((a->data->len < \
-      af_lencalc(a->mul, d)) ? af_resize_local_buffer(a, d) : AF_OK)
-
-/* Some other useful macro definitions*/
-#ifndef min
-#define min(a, b)(((a) > (b)) ? (b) : (a))
-#endif
-
-#ifndef max
-#define max(a, b)(((a) > (b)) ? (a) : (b))
-#endif
-
-#ifndef clamp
-#define clamp(a, min, max) (((a) > (max)) ? (max) : (((a) < (min)) ? (min) : (a)))
-#endif
-
-#ifndef sign
-#define sign(a) (((a) > 0) ? (1) : (-1))
-#endif
-
-#ifndef lrnd
-#define lrnd(a, b) ((b)((a) >= 0.0 ? (a) + 0.5 : (a) - 0.5))
-#endif
-
-#endif /* MPLAYER_AF_H */