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-rw-r--r--DOCS/man/af.rst158
-rwxr-xr-xTOOLS/old-configure4
-rw-r--r--TOOLS/old-makefile12
-rw-r--r--audio/audio.c7
-rw-r--r--audio/audio.h1
-rw-r--r--audio/filter/af.c23
-rw-r--r--audio/filter/af_center.c104
-rw-r--r--audio/filter/af_export.c237
-rw-r--r--audio/filter/af_extrastereo.c132
-rw-r--r--audio/filter/af_hrtf.c670
-rw-r--r--audio/filter/af_hrtf.h510
-rw-r--r--audio/filter/af_karaoke.c86
-rw-r--r--audio/filter/af_ladspa.c851
-rw-r--r--audio/filter/af_sinesuppress.c117
-rw-r--r--audio/filter/af_sub.c148
-rw-r--r--audio/filter/af_surround.c246
-rw-r--r--audio/filter/af_sweep.c92
-rw-r--r--audio/filter/dsp.h31
-rw-r--r--audio/filter/filter.c359
-rw-r--r--audio/filter/filter.h74
-rw-r--r--audio/filter/window.c212
-rw-r--r--audio/filter/window.h42
-rw-r--r--wscript4
-rw-r--r--wscript_build.py12
24 files changed, 0 insertions, 4132 deletions
diff --git a/DOCS/man/af.rst b/DOCS/man/af.rst
index 15bc3050e6..e632820811 100644
--- a/DOCS/man/af.rst
+++ b/DOCS/man/af.rst
@@ -96,25 +96,6 @@ Available filters are:
If the input channel number is less than ``<minch>``, the filter will
detach itself (default: 3).
-``sweep[=speed]``
- Produces a sine sweep.
-
- ``<0.0-1.0>``
- Sine function delta, use very low values to hear the sweep.
-
-``sinesuppress[=freq:decay]``
- Remove a sine at the specified frequency. Useful to get rid of the 50/60 Hz
- noise on low quality audio equipment. It only works on mono input.
-
- ``<freq>``
- The frequency of the sine which should be removed (in Hz) (default:
- 50)
- ``<decay>``
- Controls the adaptivity (a larger value will make the filter adapt to
- amplitude and phase changes quicker, a smaller value will make the
- adaptation slower) (default: 0.0001). Reasonable values are around
- 0.001.
-
``bs2b[=option1:option2:...]``
Bauer stereophonic to binaural transformation using libbs2b. Improves the
headphone listening experience by making the sound similar to that from
@@ -137,18 +118,6 @@ Available filters are:
If ``fcut`` or ``feed`` options are specified together with a profile, they
will be applied on top of the selected profile.
-``hrtf[=flag]``
- Head-related transfer function: Converts multichannel audio to 2-channel
- output for headphones, preserving the spatiality of the sound.
-
- ==== ===================================
- Flag Meaning
- ==== ===================================
- m matrix decoding of the rear channel
- s 2-channel matrix decoding
- 0 no matrix decoding (default)
- ==== ===================================
-
``equalizer=g1:g2:g3:...:g10``
10 octave band graphic equalizer, implemented using 10 IIR band-pass
filters. This means that it works regardless of what type of audio is
@@ -354,64 +323,6 @@ Available filters are:
``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
remixing audio to 5.1 and output it like this.
-``sub[=fc:ch]``
- Adds a subwoofer channel to the audio stream. The audio data used for
- creating the subwoofer channel is an average of the sound in channel 0 and
- channel 1. The resulting sound is then low-pass filtered by a 4th order
- Butterworth filter with a default cutoff frequency of 60Hz and added to a
- separate channel in the audio stream.
-
- .. warning::
-
- Disable this filter when you are playing media with an LFE channel
- (e.g. 5.1 surround sound), otherwise this filter will disrupt the sound
- to the subwoofer.
-
- ``<fc>``
- cutoff frequency in Hz for the low-pass filter (20 Hz to 300 Hz)
- (default: 60 Hz) For the best result try setting the cutoff frequency
- as low as possible. This will improve the stereo or surround sound
- experience.
- ``<ch>``
- Determines the channel number in which to insert the sub-channel
- audio. Channel number can be between 0 and 7 (default: 5). Observe
- that the number of channels will automatically be increased to <ch> if
- necessary.
-
- .. admonition:: Example
-
- ``mpv --af=sub=100:4 --audio-channels=5 media.avi``
- Would add a subwoofer channel with a cutoff frequency of 100 Hz to
- output channel 4.
-
-``center``
- Creates a center channel from the front channels. May currently be low
- quality as it does not implement a high-pass filter for proper extraction
- yet, but averages and halves the channels instead.
-
- ``<ch>``
- Determines the channel number in which to insert the center channel.
- Channel number can be between 0 and 7 (default: 5). Observe that the
- number of channels will automatically be increased to ``<ch>`` if
- necessary.
-
-``surround[=delay]``
- Decoder for matrix encoded surround sound like Dolby Surround. Some files
- with 2-channel audio actually contain matrix encoded surround sound.
-
- ``<delay>``
- delay time in ms for the rear speakers (0 to 1000) (default: 20) This
- delay should be set as follows: If d1 is the distance from the
- listening position to the front speakers and d2 is the distance from
- the listening position to the rear speakers, then the delay should be
- set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
-
- .. admonition:: Example
-
- ``mpv --af=surround=15 --audio-channels=4 media.avi``
- Would add surround sound decoding with 15 ms delay for the sound to
- the rear speakers.
-
``delay[=[ch1,ch2,...]]``
Delays the sound to the loudspeakers such that the sound from the
different channels arrives at the listening position simultaneously. It is
@@ -440,36 +351,6 @@ Available filters are:
Would delay front left and right by 10.5 ms, the two rear channels
and the subwoofer by 0 ms and the center channel by 7 ms.
-``export=mmapped_file:nsamples]``
- Exports the incoming signal to other processes using memory mapping
- (``mmap()``). Memory mapped areas contain a header::
-
- int nch /* number of channels */
- int size /* buffer size */
- unsigned long long counter /* Used to keep sync, updated every time
- new data is exported. */
-
- The rest is payload (non-interleaved) 16-bit data.
-
- ``<mmapped_file>``
- File to map data to (required)
- ``<nsamples>``
- number of samples per channel (default: 512).
-
- .. admonition:: Example
-
- ``mpv --af=export=/tmp/mpv-af_export:1024 media.avi``
- Would export 1024 samples per channel to ``/tmp/mpv-af_export``.
-
-``extrastereo[=mul]``
- (Linearly) increases the difference between left and right channels which
- adds some sort of "live" effect to playback.
-
- ``<mul>``
- Sets the difference coefficient (default: 2.5). 0.0 means mono sound
- (average of both channels), with 1.0 sound will be unchanged, with
- -1.0 left and right channels will be swapped.
-
``drc[=method:target]``
Applies dynamic range compression. This maximizes the volume by compressing
the audio signal's dynamic range. (Formerly called ``volnorm``.)
@@ -493,45 +374,6 @@ Available filters are:
This filter can cause distortion with audio signals that have a very
large dynamic range.
-``ladspa=file:label:[<control0>,<control1>,...]``
- Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This
- filter is reentrant, so multiple LADSPA plugins can be used at once.
-
- ``<file>``
- Specifies the LADSPA plugin library file.
-
- .. note::
-
- See also the note about the ``LADSPA_PATH`` variable in the
- `ENVIRONMENT VARIABLES`_ section.
- ``<label>``
- Specifies the filter within the library. Some libraries contain only
- one filter, but others contain many of them. Entering 'help' here
- will list all available filters within the specified library, which
- eliminates the use of 'listplugins' from the LADSPA SDK.
- ``[<control0>,<control1>,...]``
- Controls are zero or more ``,`` separated floating point values that
- determine the behavior of the loaded plugin (for example delay,
- threshold or gain).
- In verbose mode (add ``-v`` to the mpv command line), all
- available controls and their valid ranges are printed. This eliminates
- the use of 'analyseplugin' from the LADSPA SDK.
- Note that ``,`` is already used by the option parser to separate
- filters, so you must quote the list of values with ``[...]`` or
- similar.
-
- .. admonition:: Example
-
- ``mpv --af=ladspa='/usr/lib/ladspa/delay.so':delay_5s:[0.5,0.2] media.avi``
- Does something.
-
-``karaoke``
- Simple voice removal filter exploiting the fact that voice is usually
- recorded with mono gear and later 'center' mixed onto the final audio
- stream. Beware that this filter will turn your signal into mono. Works
- well for 2 channel tracks; do not bother trying it on anything but 2
- channel stereo.
-
``scaletempo[=option1:option2:...]``
Scales audio tempo without altering pitch, optionally synced to playback
speed (default).
diff --git a/TOOLS/old-configure b/TOOLS/old-configure
index 60575a8ec1..e967a657eb 100755
--- a/TOOLS/old-configure
+++ b/TOOLS/old-configure
@@ -191,7 +191,6 @@ options_state_machine() {
opt_yes_no _libcdio "libcdio support"
opt_yes_no _librubberband "librubberband support"
opt_yes_no _ffmpeg "skip FFmpeg/Libav autodetection"
- opt_yes_no _ladspa "LADSPA plugin support"
opt_yes_no _libbs2b "libbs2b audio filter support"
opt_yes_no _libavresample "libavresample (preferred over libswresample)"
opt_yes_no _libswresample "libswresample"
@@ -749,9 +748,6 @@ check_pkg_config "uchardet" $_uchardet UCHARDET 'uchardet'
check_pkg_config "zlib" auto ZLIB 'zlib'
test $(defretval) = no && die "Unable to find development files for zlib."
-test "$_dl" = no && _ladspa=no
-check_statement_libs "LADSPA plugin support" $_ladspa LADSPA ladspa.h 'LADSPA_Descriptor ld = {0}'
-
check_pkg_config "libbs2b audio filter support" $_libbs2b LIBBS2B 'libbs2b'
check_pkg_config "LCMS2 support" $_lcms2 LCMS2 'lcms2 >= 2.6'
diff --git a/TOOLS/old-makefile b/TOOLS/old-makefile
index 4f5737595f..f3e4d7ea86 100644
--- a/TOOLS/old-makefile
+++ b/TOOLS/old-makefile
@@ -35,7 +35,6 @@ SOURCES-$(DVDREAD) += stream/stream_dvd.c \
SOURCES-$(DVDNAV) += stream/stream_dvdnav.c \
stream/stream_dvd_common.c
-SOURCES-$(LADSPA) += audio/filter/af_ladspa.c
SOURCES-$(RUBBERBAND) += audio/filter/af_rubberband.c
SOURCES-$(LIBASS) += sub/ass_mp.c sub/sd_ass.c \
demux/demux_libass.c
@@ -123,28 +122,17 @@ SOURCES = audio/audio.c \
audio/decode/ad_spdif.c \
audio/decode/dec_audio.c \
audio/filter/af.c \
- audio/filter/af_center.c \
audio/filter/af_channels.c \
audio/filter/af_delay.c \
audio/filter/af_equalizer.c \
- audio/filter/af_export.c \
- audio/filter/af_extrastereo.c \
audio/filter/af_format.c \
- audio/filter/af_hrtf.c \
- audio/filter/af_karaoke.c \
audio/filter/af_lavcac3enc.c \
audio/filter/af_lavrresample.c \
audio/filter/af_pan.c \
audio/filter/af_scaletempo.c \
- audio/filter/af_sinesuppress.c \
- audio/filter/af_sub.c \
- audio/filter/af_surround.c \
- audio/filter/af_sweep.c \
audio/filter/af_drc.c \
audio/filter/af_volume.c \
- audio/filter/filter.c \
audio/filter/tools.c \
- audio/filter/window.c \
audio/out/ao.c \
audio/out/ao_null.c \
audio/out/ao_pcm.c \
diff --git a/audio/audio.c b/audio/audio.c
index f84d6054bc..4b12992879 100644
--- a/audio/audio.c
+++ b/audio/audio.c
@@ -58,13 +58,6 @@ void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels)
update_redundant_info(mpa);
}
-// Use old MPlayer/ALSA channel layout.
-void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels)
-{
- mp_chmap_from_channels_alsa(&mpa->channels, num_channels);
- update_redundant_info(mpa);
-}
-
void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap)
{
mpa->channels = *chmap;
diff --git a/audio/audio.h b/audio/audio.h
index a0ecb2d7bf..bf5358274a 100644
--- a/audio/audio.h
+++ b/audio/audio.h
@@ -46,7 +46,6 @@ struct mp_audio {
void mp_audio_set_format(struct mp_audio *mpa, int format);
void mp_audio_set_num_channels(struct mp_audio *mpa, int num_channels);
-void mp_audio_set_channels_old(struct mp_audio *mpa, int num_channels);
void mp_audio_set_channels(struct mp_audio *mpa, const struct mp_chmap *chmap);
void mp_audio_copy_config(struct mp_audio *dst, const struct mp_audio *src);
bool mp_audio_config_equals(const struct mp_audio *a, const struct mp_audio *b);
diff --git a/audio/filter/af.c b/audio/filter/af.c
index 6a5b1f42a5..b877ba7661 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -34,23 +34,12 @@
extern const struct af_info af_info_delay;
extern const struct af_info af_info_channels;
extern const struct af_info af_info_format;
-extern const struct af_info af_info_force;
extern const struct af_info af_info_volume;
extern const struct af_info af_info_equalizer;
extern const struct af_info af_info_pan;
-extern const struct af_info af_info_surround;
-extern const struct af_info af_info_sub;
-extern const struct af_info af_info_export;
extern const struct af_info af_info_drc;
-extern const struct af_info af_info_extrastereo;
extern const struct af_info af_info_lavcac3enc;
extern const struct af_info af_info_lavrresample;
-extern const struct af_info af_info_sweep;
-extern const struct af_info af_info_hrtf;
-extern const struct af_info af_info_ladspa;
-extern const struct af_info af_info_center;
-extern const struct af_info af_info_sinesuppress;
-extern const struct af_info af_info_karaoke;
extern const struct af_info af_info_scaletempo;
extern const struct af_info af_info_bs2b;
extern const struct af_info af_info_lavfi;
@@ -63,24 +52,12 @@ static const struct af_info *const filter_list[] = {
&af_info_volume,
&af_info_equalizer,
&af_info_pan,
- &af_info_surround,
- &af_info_sub,
- &af_info_export,
&af_info_drc,
- &af_info_extrastereo,
&af_info_lavcac3enc,
&af_info_lavrresample,
- &af_info_sweep,
- &af_info_hrtf,
-#if HAVE_LADSPA
- &af_info_ladspa,
-#endif
#if HAVE_RUBBERBAND
&af_info_rubberband,
#endif
- &af_info_center,
- &af_info_sinesuppress,
- &af_info_karaoke,
&af_info_scaletempo,
#if HAVE_LIBBS2B
&af_info_bs2b,
diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c
deleted file mode 100644
index 69e54e81c6..0000000000
--- a/audio/filter/af_center.c
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * This filter adds a center channel to the audio stream by
- * averaging the left and right channel.
- * There are two runtime controls one for setting which channel
- * to insert the center-audio into called AF_CONTROL_SUB_CH.
- *
- * FIXME: implement a high-pass filter for better results.
- *
- * copyright (c) 2005 Alex Beregszaszi
- *
- * This file is part of mpv.
- *
- * mpv 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.
- *
- * mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "common/common.h"
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_center_s
-{
- int ch; // Channel number which to insert the filtered data
-}af_center_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_center_t* s = af->priv;
-
- switch(cmd){
- case AF_CONTROL_REINIT:{
- // Sanity check
- if(!arg) return AF_ERROR;
-
- af->data->rate = ((struct mp_audio*)arg)->rate;
- mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
- mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-
- return af_test_output(af,(struct mp_audio*)arg);
- }
- }
- return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance* af, struct mp_audio* data)
-{
- if (!data)
- return 0;
- if (af_make_writeable(af, data) < 0) {
- talloc_free(data);
- return -1;
- }
- struct mp_audio* c = data; // Current working data
- af_center_t* s = af->priv; // Setup for this instance
- float* a = c->planes[0]; // Audio data
- int nch = c->nch; // Number of channels
- int len = c->samples*c->nch; // Number of samples in current audio block
- int ch = s->ch; // Channel in which to insert the center audio
- register int i;
-
- // Run filter
- for(i=0;i<len;i+=nch){
- // Average left and right
- a[i+ch] = (a[i]/2) + (a[i+1]/2);
- }
-
- af_add_output_frame(af, data);
- return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
- af->control=control;
- af->filter_frame = filter_frame;
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_center_t
-const struct af_info af_info_center = {
- .info = "Audio filter for adding a center channel",
- .name = "center",
- .flags = AF_FLAGS_NOT_REENTRANT,
- .open = af_open,
- .priv_size = sizeof(af_center_t),
- .options = (const struct m_option[]) {
- OPT_INTRANGE("channel", ch, 0, 0, AF_NCH - 1, OPTDEF_INT(1)),
- {0}
- },
-};
diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c
deleted file mode 100644
index 6020d9d98e..0000000000
--- a/audio/filter/af_export.c
+++ /dev/null
@@ -1,237 +0,0 @@
-/*
- * This audio filter exports the incoming signal to other processes
- * using memory mapping. The memory mapped area contains a header:
- * int nch,
- * int size,
- * unsigned long long counter (updated every time the contents of
- * the area changes),
- * the rest is payload (non-interleaved).
- *
- * Authors: Anders; Gustavo Sverzut Barbieri <gustavo.barbieri@ic.unicamp.br>
- *
- * This file is part of mpv.
- *
- * mpv 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.
- *
- * mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-#include <unistd.h>
-#include "config.h"
-
-#include <sys/types.h>
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-
-#include "osdep/io.h"
-
-#include "talloc.h"
-#include "af.h"
-#include "options/path.h"
-
-#define DEF_SZ 512 // default buffer size (in samples)
-#define SHARED_FILE "mpv-af_export" /* default file name
- (relative to ~/.mpv/ */
-
-#define SIZE_HEADER (2 * sizeof(int) + sizeof(unsigned long long))
-
-// Data for specific instances of this filter
-typedef struct af_export_s
-{
- unsigned long long count; // Used for sync
- void* buf[AF_NCH]; // Buffers for storing the data before it is exported
- int sz; // Size of buffer in samples
- int wi; // Write index
- int fd; // File descriptor to shared memory area
- char* filename; // File to export data
- uint8_t *mmap_area; // MMap shared area
-} af_export_t;
-
-
-/* Initialization and runtime control
- af audio filter instance
- cmd control command
- arg argument
-*/
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_export_t* s = af->priv;
- switch (cmd){
- case AF_CONTROL_REINIT:{
- int i=0;
- int mapsize;
-
- // Free previous buffers
- free(s->buf[0]);
-
- // unmap previous area
- if(s->mmap_area)
- munmap(s->mmap_area, SIZE_HEADER + (af->data->bps*s->sz*af->data->nch));
- // close previous file descriptor
- if(s->fd)
- close(s->fd);
-
- // Accept only int16_t as input format (which sucks)
- mp_audio_copy_config(af->data, (struct mp_audio*)arg);
- mp_audio_set_format(af->data, AF_FORMAT_S16);
-
- // Allocate new buffers (as one continuous block)
- s->buf[0] = calloc(s->sz*af->data->nch, af->data->bps);
- if(NULL == s->buf[0]) {
- MP_FATAL(af, "Out of memory\n");
- return AF_ERROR;
- }
- for(i = 1; i < af->data->nch; i++)
- s->buf[i] = (uint8_t *)s->buf[0] + i*s->sz*af->data->bps;
-
- if (!s->filename) {
- MP_FATAL(af, "No filename set.\n");
- return AF_ERROR;
- }
-
- // Init memory mapping
- s->fd = open(s->filename, O_RDWR | O_CREAT | O_TRUNC | O_CLOEXEC, 0640);
- MP_INFO(af, "Exporting to file: %s\n", s->filename);
- if(s->fd < 0) {
- MP_FATAL(af, "Could not open/create file: %s\n",
- s->filename);
- return AF_ERROR;
- }
-
- // header + buffer
- mapsize = (SIZE_HEADER + (af->data->bps * s->sz * af->data->nch));
-
- // grow file to needed size
- for(i = 0; i < mapsize; i++){
- char null = 0;
- write(s->fd, (void*) &null, 1);
- }
-
- // mmap size
- s->mmap_area = mmap(0, mapsize, PROT_READ|PROT_WRITE,MAP_SHARED, s->fd, 0);
- if(s->mmap_area == NULL)
- MP_FATAL(af, "Could not mmap file %s\n", s->filename);
- MP_INFO(af, "Memory mapped to file: %s (%p)\n",
- s->filename, s->mmap_area);
-
- // Initialize header
- *((int*)s->mmap_area) = af->data->nch;
- *((int*)s->mmap_area + 1) = s->sz * af->data->bps * af->data->nch;
- msync(s->mmap_area, mapsize, MS_ASYNC);
-
- // Use test_output to return FALSE if necessary
- return af_test_output(af, (struct mp_audio*)arg);
- }
- }
- return AF_UNKNOWN;
-}
-
-/* Free allocated memory and clean up other stuff too.
- af audio filter instance
-*/
-static void uninit( struct af_instance* af )
-{
- af_export_t* s = af->priv;
-
- free(s->buf[0]);
-
- // Free mmaped area
- if(s->mmap_area)
- munmap(s->mmap_area, sizeof(af_export_t));
-
- if(s->fd > -1)
- close(s->fd);
-}
-
-/* Filter data through filter
- af audio filter instance
- data audio data
-*/
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
- if (!data)
- return 0;
- struct mp_audio* c = data; // Current working data
- af_export_t* s = af->priv; // Setup for this instance
- int16_t* a = c->planes[0]; // Incoming sound
- int nch = c->nch; // Number of channels
- int len = c->samples*c->nch; // Number of sample in data chunk
- int sz = s->sz; // buffer size (in samples)
- int flag = 0; // Set to 1 if buffer is filled
-
- int ch, i;
-
- // Fill all buffers
- for(ch = 0; ch < nch; ch++){
- int wi = s->wi; // Reset write index
- int16_t* b = s->buf[ch]; // Current buffer
-
- // Copy data to export buffers
- for(i = ch; i < len; i += nch){
- b[wi++] = a[i];
- if(wi >= sz){ // Don't write outside the end of the buffer
- flag = 1;
- break;
- }
- }
- s->wi = wi % s->sz;
- }
-
- // Export buffer to mmaped area
- if(flag){
- // update buffer in mapped area
- memcpy(s->mmap_area + SIZE_HEADER, s->buf[0], sz * c->bps * nch);
- s->count++; // increment counter (to sync)
- memcpy(s->mmap_area + SIZE_HEADER - sizeof(s->count),
- &(s->count), sizeof(s->count));
- }
-
- af_add_output_frame(af, data);
- return 0;
-}
-
-/* Allocate memory and set function pointers
- af audio filter instance
- returns AF_OK or AF_ERROR
-*/
-static int af_open( struct af_instance* af )
-{
- af->control = control;
- af->uninit = uninit;
- af->filter_frame = filter;
- af_export_t *priv = af->priv;
-
- if (!priv->filename || !priv->filename[0]) {
- MP_FATAL(af, "no export filename given");
- return AF_ERROR;
- }
-
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_export_t
-const struct af_info af_info_export = {
- .info = "Sound export filter",
- .name = "export",
- .open = af_open,
- .priv_size = sizeof(af_export_t),
- .options = (const struct m_option[]) {
- OPT_STRING("filename", filename, 0),
- OPT_INTRANGE("buffersamples", sz, 0, 1, 2048, OPTDEF_INT(DEF_SZ)),
- {0}
- },
-};
diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c
deleted file mode 100644
index 49222ebfdc..0000000000
--- a/audio/filter/af_extrastereo.c
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
- *
- * This file is part of mpv.
- *
- * mpv 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.
- *
- * mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <inttypes.h>
-#include <math.h>
-#include <limits.h>
-
-#include "common/common.h"
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_extrastereo_s
-{
- float mul;
-}af_extrastereo_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- switch(cmd){
- case AF_CONTROL_REINIT:{
- // Sanity check
- if(!arg) return AF_ERROR;
-
- mp_audio_copy_config(af->data, (struct mp_audio*)arg);
- mp_audio_force_interleaved_format(af->data);
- mp_audio_set_num_channels(af->data, 2);
- if (af->data->format != AF_FORMAT_FLOAT)
- mp_audio_set_format(af->data, AF_FORMAT_S16);
-
- return af_test_output(af,(struct mp_audio*)arg);
- }
- }
- return AF_UNKNOWN;
-}
-
-// Filter data through filter
-static void play_s16(af_extrastereo_t *s, struct mp_audio* data)
-{
- register int i = 0;
- int16_t *a = (int16_t*)data->planes[0]; // Audio data
- int len = data->samples*data->nch; // Number of samples
- int avg, l, r;
-
- for (i = 0; i < len; i+=2)
- {
- avg = (a[i] + a[i + 1]) / 2;
-
- l = avg + (int)(s->mul * (a[i] - avg));
- r = avg + (int)(s->mul * (a[i + 1] - avg));
-
- a[i] = MPCLAMP(l, SHRT_MIN, SHRT_MAX);
- a[i + 1] = MPCLAMP(r, SHRT_MIN, SHRT_MAX);
- }
-}
-
-static void play_float(af_extrastereo_t *s, struct mp_audio* data)
-{
- register int i = 0;
- float *a = (float*)data->planes[0]; // Audio data
- int len = data->samples * data->nch; // Number of samples
- float avg, l, r;
-
- for (i = 0; i < len; i+=2)
- {
- avg = (a[i] + a[i + 1]) / 2;
-
- l = avg + (s->mul * (a[i] - avg));
- r = avg + (s->mul * (a[i + 1] - avg));
-
- a[i] = af_softclip(l);
- a[i + 1] = af_softclip(r);
- }
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *data)
-{
- if (!data)
- return 0;
- if (af_make_writeable(af, data) < 0) {
- talloc_free(data);
- return -1;
- }
- if (data->format == AF_FORMAT_FLOAT) {
- play_float(af->priv, data);
- } else {
- play_s16(af->priv, data);
- }
- af_add_output_frame(af, data);
- return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
- af->control=control;
- af->filter_frame = filter_frame;
-
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_extrastereo_t
-const struct af_info af_info_extrastereo = {
- .info = "Increase difference between audio channels",
- .name = "extrastereo",
- .flags = AF_FLAGS_NOT_REENTRANT,
- .open = af_open,
- .priv_size = sizeof(af_extrastereo_t),
- .options = (const struct m_option[]) {
- OPT_FLOAT("mul", mul, 0, OPTDEF_FLOAT(2.5)),
- {0}
- },
-};
diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c
deleted file mode 100644
index 3c8a89c665..0000000000
--- a/audio/filter/af_hrtf.c
+++ /dev/null
@@ -1,670 +0,0 @@
-/*
- * Experimental audio filter that mixes 5.1 and 5.1 with matrix
- * encoded rear channels into headphone signal using FIR filtering
- * with HRTF.
- *
- * Author: ylai
- *
- * This file is part of mpv.
- *
- * mpv 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.
- *
- * mpv 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.