diff options
| -rw-r--r-- | DOCS/man/af.rst | 38 | ||||
| -rw-r--r-- | audio/filter/af_lavrresample.c | 112 | ||||
| -rw-r--r-- | wscript_build.py | 1 |
3 files changed, 0 insertions, 151 deletions
diff --git a/DOCS/man/af.rst b/DOCS/man/af.rst index 4e806d482e..aebf76431a 100644 --- a/DOCS/man/af.rst +++ b/DOCS/man/af.rst @@ -24,44 +24,6 @@ See ``--vf`` group of options for info on how ``--af-defaults``, ``--af-add``, Available filters are: -``lavrresample[=option1:option2:...]`` - This filter uses libavresample (or libswresample, depending on the build) - to change sample rate, sample format, or channel layout of the audio stream. - This filter is automatically enabled if the audio output does not support - the audio configuration of the file being played. - - .. warning:: - - Deprecated. Either use the ``--audio-resample-...`` options to customize - resampling, or the libavfilter ``--af=aresample`` filter, which has its - own options. - - It supports only the following sample formats: u8, s16, s32, float. - - ``filter-size=<length>`` - Length of the filter with respect to the lower sampling rate. (default: - 16) - ``phase-shift=<count>`` - Log2 of the number of polyphase entries. (..., 10->1024, 11->2048, - 12->4096, ...) (default: 10->1024) - ``cutoff=<cutoff>`` - Cutoff frequency (0.0-1.0), default set depending upon filter length. - ``linear`` - If set then filters will be linearly interpolated between polyphase - entries. (default: no) - ``no-detach`` - Do not detach if input and output audio format/rate/channels match. - (If you just want to set defaults for this filter that will be used - even by automatically inserted lavrresample instances, you should - prefer setting them with the ``--audio-resample-...`` options.) This - does not do anything anymore and the filter will never detach. - ``normalize=<yes|no|auto>`` - Whether to normalize when remixing channel layouts (default: auto). - ``auto`` uses the value set by ``--audio-normalize-downmix``. - ``o=<string>`` - Set AVOptions on the SwrContext or AVAudioResampleContext. These should - be documented by FFmpeg or Libav. - ``lavcac3enc[=options]`` Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports 16-bit native-endian input format, maximum 6 channels. The output is diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c deleted file mode 100644 index baa78acb6e..0000000000 --- a/audio/filter/af_lavrresample.c +++ /dev/null @@ -1,112 +0,0 @@ -/* - * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> - * Copyright (c) 2013 Stefano Pigozzi <stefano.pigozzi@gmail.com> - * - * Based on Michael Niedermayer's lavcresample. - * - * 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 <assert.h> - -#include "common/common.h" -#include "config.h" - -#include "common/av_common.h" -#include "common/msg.h" -#include "filters/f_swresample.h" -#include "filters/filter_internal.h" -#include "filters/user_filters.h" -#include "options/m_config.h" -#include "options/m_option.h" -#include "options/options.h" - -struct af_resample { - int allow_detach; - struct mp_resample_opts opts; - int global_normalize; -}; - -static void set_defaults(struct mpv_global *global, void *p) -{ - struct af_resample *s = p; - - struct mp_resample_opts *opts = &s->opts; - - struct mp_resample_opts *src_opts = - mp_get_config_group(s, global, &resample_conf); - - s->global_normalize = src_opts->normalize; - - assert(!opts->avopts); // we don't set a default value, so it must be NULL - - *opts = *src_opts; - - opts->avopts = NULL; - struct m_option dummy = {.type = &m_option_type_keyvalue_list}; - m_option_copy(&dummy, &opts->avopts, &src_opts->avopts); -} - -#define OPT_BASE_STRUCT struct af_resample - -static struct mp_filter *af_lavrresample_create(struct mp_filter *parent, - void *options) -{ - struct af_resample *s = options; - - if (s->opts.normalize < 0) - s->opts.normalize = s->global_normalize; - - struct mp_swresample *swr = mp_swresample_create(parent, &s->opts); - if (!swr) - abort(); - - MP_WARN(swr->f, "This filter is deprecated! Use the --audio-resample- options" - " to customize resampling, or the --af=aresample filter.\n"); - - talloc_free(s); - return swr->f; -} - -const struct mp_user_filter_entry af_lavrresample = { - .desc = { - .description = "Sample frequency conversion using libavresample", - .name = "lavrresample", - .priv_size = sizeof(struct af_resample), - .priv_defaults = &(const struct af_resample) { - .opts = MP_RESAMPLE_OPTS_DEF, - .allow_detach = 1, - }, - .options = (const struct m_option[]) { - OPT_INTRANGE("filter-size", opts.filter_size, 0, 0, 32), - OPT_INTRANGE("phase-shift", opts.phase_shift, 0, 0, 30), - OPT_FLAG("linear", opts.linear, 0), - OPT_DOUBLE("cutoff", opts.cutoff, M_OPT_RANGE, .min = 0, .max = 1), - OPT_FLAG("detach", allow_detach, 0), // does nothing - OPT_CHOICE("normalize", opts.normalize, 0, - ({"no", 0}, {"yes", 1}, {"auto", -1})), - OPT_KEYVALUELIST("o", opts.avopts, 0), - {0} - }, - .set_defaults = set_defaults, - }, - .create = af_lavrresample_create, -}; diff --git a/wscript_build.py b/wscript_build.py index 2c60a8dcf6..8f2f804a39 100644 --- a/wscript_build.py +++ b/wscript_build.py @@ -228,7 +228,6 @@ def build(ctx): ( "audio/decode/ad_spdif.c" ), ( "audio/filter/af_format.c" ), ( "audio/filter/af_lavcac3enc.c" ), - ( "audio/filter/af_lavrresample.c" ), ( "audio/filter/af_rubberband.c", "rubberband" ), ( "audio/filter/af_scaletempo.c" ), ( "audio/fmt-conversion.c" ), |