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author | Michael Niedermayer <michaelni@gmx.at> | 2012-04-25 22:01:59 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-04-25 23:17:41 +0200 |
commit | 3ead79eaa3f77451bc93cb842ed7b38c94858045 (patch) | |
tree | 4c90cceb47c62f96eaa518cd9ad109b539e9f1ad /libavresample/options.c | |
parent | cab15f9db4ba6e390b25dd80d7305bb51b5583c4 (diff) | |
parent | 394dbde5484507f213768019623d016196ddad5f (diff) | |
download | ffmpeg-3ead79eaa3f77451bc93cb842ed7b38c94858045.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
FATE: use updated reference for aac-latm_stereo_to_51
avconv: use libavresample
Add libavresample
FATE: avoid channel mixing in lavf-dv_fmt
Conflicts:
Changelog
Makefile
cmdutils.c
configure
doc/APIchanges
ffmpeg.c
tests/lavf-regression.sh
tests/ref/lavf/dv_fmt
tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavresample/options.c')
-rw-r--r-- | libavresample/options.c | 89 |
1 files changed, 89 insertions, 0 deletions
diff --git a/libavresample/options.c b/libavresample/options.c new file mode 100644 index 0000000000..0be1a26117 --- /dev/null +++ b/libavresample/options.c @@ -0,0 +1,89 @@ +/* + * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> + * + * This file is part of Libav. + * + * Libav is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * Libav 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 + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "libavutil/mathematics.h" +#include "libavutil/opt.h" +#include "avresample.h" +#include "internal.h" +#include "audio_mix.h" + +/** + * @file + * Options definition for AVAudioResampleContext. + */ + +#define OFFSET(x) offsetof(AVAudioResampleContext, x) +#define PARAM AV_OPT_FLAG_AUDIO_PARAM + +static const AVOption options[] = { + { "in_channel_layout", "Input Channel Layout", OFFSET(in_channel_layout), AV_OPT_TYPE_INT64, { 0 }, INT64_MIN, INT64_MAX, PARAM }, + { "in_sample_fmt", "Input Sample Format", OFFSET(in_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM }, + { "in_sample_rate", "Input Sample Rate", OFFSET(in_sample_rate), AV_OPT_TYPE_INT, { 48000 }, 1, INT_MAX, PARAM }, + { "out_channel_layout", "Output Channel Layout", OFFSET(out_channel_layout), AV_OPT_TYPE_INT64, { 0 }, INT64_MIN, INT64_MAX, PARAM }, + { "out_sample_fmt", "Output Sample Format", OFFSET(out_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM }, + { "out_sample_rate", "Output Sample Rate", OFFSET(out_sample_rate), AV_OPT_TYPE_INT, { 48000 }, 1, INT_MAX, PARAM }, + { "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_FLTP }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM }, + { "mix_coeff_type", "Mixing Coefficient Type", OFFSET(mix_coeff_type), AV_OPT_TYPE_INT, { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q6, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" }, + { "q6", "16-bit 10.6 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q6 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" }, + { "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" }, + { "flt", "Floating-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" }, + { "center_mix_level", "Center Mix Level", OFFSET(center_mix_level), AV_OPT_TYPE_DOUBLE, { M_SQRT1_2 }, -32.0, 32.0, PARAM }, + { "surround_mix_level", "Surround Mix Level", OFFSET(surround_mix_level), AV_OPT_TYPE_DOUBLE, { M_SQRT1_2 }, -32.0, 32.0, PARAM }, + { "lfe_mix_level", "LFE Mix Level", OFFSET(lfe_mix_level), AV_OPT_TYPE_DOUBLE, { 0.0 }, -32.0, 32.0, PARAM }, + { "force_resampling", "Force Resampling", OFFSET(force_resampling), AV_OPT_TYPE_INT, { 0 }, 0, 1, PARAM }, + { "filter_size", "Resampling Filter Size", OFFSET(filter_size), AV_OPT_TYPE_INT, { 16 }, 0, 32, /* ??? */ PARAM }, + { "phase_shift", "Resampling Phase Shift", OFFSET(phase_shift), AV_OPT_TYPE_INT, { 10 }, 0, 30, /* ??? */ PARAM }, + { "linear_interp", "Use Linear Interpolation", OFFSET(linear_interp), AV_OPT_TYPE_INT, { 0 }, 0, 1, PARAM }, + { "cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { 0.8 }, 0.0, 1.0, PARAM }, + { NULL }, +}; + +static const AVClass av_resample_context_class = { + .class_name = "AVAudioResampleContext", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; + +AVAudioResampleContext *avresample_alloc_context(void) +{ + AVAudioResampleContext *avr; + + avr = av_mallocz(sizeof(*avr)); + if (!avr) + return NULL; + + avr->av_class = &av_resample_context_class; + av_opt_set_defaults(avr); + + avr->am = av_mallocz(sizeof(*avr->am)); + if (!avr->am) { + av_free(avr); + return NULL; + } + avr->am->avr = avr; + + return avr; +} + +const AVClass *avresample_get_class(void) +{ + return &av_resample_context_class; +} |