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![Stefano Sabatini](/assets/img/avatar_default.png)
The function strtok_r() is part of the POSIX.1 specification, but is not available on some platforms. We provide an internal implementation, so we do not need to rely on a platform implementation.
419 lines
17 KiB
C
419 lines
17 KiB
C
/*
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* Copyright (c) 2010 S.N. Hemanth Meenakshisundaram <smeenaks@ucsd.edu>
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* Copyright (c) 2011 Stefano Sabatini
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* Copyright (c) 2011 Mina Nagy Zaki
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* sample format and channel layout conversion audio filter
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* based on code in libavcodec/resample.c by Fabrice Bellard and
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* libavcodec/audioconvert.c by Michael Niedermayer
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*/
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#include "libavutil/audioconvert.h"
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#include "libavutil/avstring.h"
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#include "libavcodec/audioconvert.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct {
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enum AVSampleFormat out_sample_fmt, in_sample_fmt; ///< in/out sample formats
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int64_t out_chlayout, in_chlayout; ///< in/out channel layout
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int out_nb_channels, in_nb_channels; ///< number of in/output channels
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enum AVFilterPacking out_packing_fmt, in_packing_fmt; ///< output packing format
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int max_nb_samples; ///< maximum number of buffered samples
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AVFilterBufferRef *mix_samplesref; ///< rematrixed buffer
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AVFilterBufferRef *out_samplesref; ///< output buffer after required conversions
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uint8_t *in_mix[8], *out_mix[8]; ///< input/output for rematrixing functions
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uint8_t *packed_data[8]; ///< pointers for packing conversion
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int out_strides[8], in_strides[8]; ///< input/output strides for av_audio_convert
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uint8_t **in_conv, **out_conv; ///< input/output for av_audio_convert
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AVAudioConvert *audioconvert_ctx; ///< context for conversion to output sample format
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void (*convert_chlayout)(); ///< function to do the requested rematrixing
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} AConvertContext;
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#define REMATRIX_FUNC_SIG(NAME) static void REMATRIX_FUNC_NAME(NAME) \
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(FMT_TYPE *outp[], FMT_TYPE *inp[], int nb_samples, AConvertContext *aconvert)
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#define FMT_TYPE uint8_t
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#define REMATRIX_FUNC_NAME(NAME) NAME ## _u8
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#include "af_aconvert_rematrix.c"
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#define FMT_TYPE int16_t
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#define REMATRIX_FUNC_NAME(NAME) NAME ## _s16
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#include "af_aconvert_rematrix.c"
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#define FMT_TYPE int32_t
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#define REMATRIX_FUNC_NAME(NAME) NAME ## _s32
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#include "af_aconvert_rematrix.c"
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#define FLOATING
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#define FMT_TYPE float
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#define REMATRIX_FUNC_NAME(NAME) NAME ## _flt
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#include "af_aconvert_rematrix.c"
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#define FMT_TYPE double
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#define REMATRIX_FUNC_NAME(NAME) NAME ## _dbl
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#include "af_aconvert_rematrix.c"
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#define FMT_TYPE uint8_t
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#define REMATRIX_FUNC_NAME(NAME) NAME
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REMATRIX_FUNC_SIG(stereo_remix_planar)
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{
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int size = av_get_bytes_per_sample(aconvert->in_sample_fmt) * nb_samples;
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memcpy(outp[0], inp[0], size);
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memcpy(outp[1], inp[aconvert->in_nb_channels == 1 ? 0 : 1], size);
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}
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#define REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC, PACKING) \
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{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_U8, FUNC##_u8}, \
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{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S16, FUNC##_s16}, \
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{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_S32, FUNC##_s32}, \
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{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_FLT, FUNC##_flt}, \
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{INCHLAYOUT, OUTCHLAYOUT, PACKING, AV_SAMPLE_FMT_DBL, FUNC##_dbl},
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#define REGISTER_FUNC(INCHLAYOUT, OUTCHLAYOUT, FUNC) \
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REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_packed, AVFILTER_PACKED) \
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REGISTER_FUNC_PACKING(INCHLAYOUT, OUTCHLAYOUT, FUNC##_planar, AVFILTER_PLANAR)
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static struct RematrixFunctionInfo {
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int64_t in_chlayout, out_chlayout;
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int planar, sfmt;
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void (*func)();
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} rematrix_funcs[] = {
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REGISTER_FUNC (AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_5POINT1, stereo_to_surround_5p1)
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REGISTER_FUNC (AV_CH_LAYOUT_5POINT1, AV_CH_LAYOUT_STEREO, surround_5p1_to_stereo)
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REGISTER_FUNC_PACKING(AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_MONO, stereo_to_mono_packed, AVFILTER_PACKED)
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REGISTER_FUNC_PACKING(AV_CH_LAYOUT_MONO, AV_CH_LAYOUT_STEREO, mono_to_stereo_packed, AVFILTER_PACKED)
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REGISTER_FUNC (0, AV_CH_LAYOUT_MONO, mono_downmix)
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REGISTER_FUNC_PACKING(0, AV_CH_LAYOUT_STEREO, stereo_downmix_packed, AVFILTER_PACKED)
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// This function works for all sample formats
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{0, AV_CH_LAYOUT_STEREO, AVFILTER_PLANAR, -1, stereo_remix_planar}
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};
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static av_cold int init(AVFilterContext *ctx, const char *args0, void *opaque)
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{
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AConvertContext *aconvert = ctx->priv;
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char *arg, *ptr = NULL;
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int ret = 0;
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char *args = av_strdup(args0);
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aconvert->out_sample_fmt = AV_SAMPLE_FMT_NONE;
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aconvert->out_chlayout = 0;
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aconvert->out_packing_fmt = -1;
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if ((arg = av_strtok(args, ":", &ptr)) && strcmp(arg, "auto")) {
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if ((ret = ff_parse_sample_format(&aconvert->out_sample_fmt, arg, ctx)) < 0)
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goto end;
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}
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if ((arg = av_strtok(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
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if ((ret = ff_parse_channel_layout(&aconvert->out_chlayout, arg, ctx)) < 0)
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goto end;
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}
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if ((arg = av_strtok(NULL, ":", &ptr)) && strcmp(arg, "auto")) {
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if ((ret = ff_parse_packing_format((int *)&aconvert->out_packing_fmt, arg, ctx)) < 0)
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goto end;
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}
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end:
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av_freep(&args);
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return ret;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AConvertContext *aconvert = ctx->priv;
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avfilter_unref_buffer(aconvert->mix_samplesref);
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avfilter_unref_buffer(aconvert->out_samplesref);
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if (aconvert->audioconvert_ctx)
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av_audio_convert_free(aconvert->audioconvert_ctx);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AConvertContext *aconvert = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
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&inlink->out_formats);
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if (aconvert->out_sample_fmt != AV_SAMPLE_FMT_NONE) {
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formats = NULL;
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avfilter_add_format(&formats, aconvert->out_sample_fmt);
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avfilter_formats_ref(formats, &outlink->in_formats);
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} else
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avfilter_formats_ref(avfilter_make_all_formats(AVMEDIA_TYPE_AUDIO),
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&outlink->in_formats);
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avfilter_formats_ref(avfilter_make_all_channel_layouts(),
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&inlink->out_chlayouts);
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if (aconvert->out_chlayout != 0) {
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formats = NULL;
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avfilter_add_format(&formats, aconvert->out_chlayout);
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avfilter_formats_ref(formats, &outlink->in_chlayouts);
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} else
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avfilter_formats_ref(avfilter_make_all_channel_layouts(),
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&outlink->in_chlayouts);
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avfilter_formats_ref(avfilter_make_all_packing_formats(),
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&inlink->out_packing);
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if (aconvert->out_packing_fmt != -1) {
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formats = NULL;
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avfilter_add_format(&formats, aconvert->out_packing_fmt);
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avfilter_formats_ref(formats, &outlink->in_packing);
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} else
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avfilter_formats_ref(avfilter_make_all_packing_formats(),
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&outlink->in_packing);
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterLink *inlink = outlink->src->inputs[0];
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AConvertContext *aconvert = outlink->src->priv;
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char buf1[64], buf2[64];
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aconvert->in_sample_fmt = inlink->format;
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aconvert->in_packing_fmt = inlink->planar;
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if (aconvert->out_packing_fmt == -1)
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aconvert->out_packing_fmt = outlink->planar;
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aconvert->in_chlayout = inlink->channel_layout;
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aconvert->in_nb_channels =
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av_get_channel_layout_nb_channels(inlink->channel_layout);
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/* if not specified in args, use the format and layout of the output */
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if (aconvert->out_sample_fmt == AV_SAMPLE_FMT_NONE)
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aconvert->out_sample_fmt = outlink->format;
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if (aconvert->out_chlayout == 0)
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aconvert->out_chlayout = outlink->channel_layout;
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aconvert->out_nb_channels =
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av_get_channel_layout_nb_channels(outlink->channel_layout);
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av_get_channel_layout_string(buf1, sizeof(buf1),
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-1, inlink ->channel_layout);
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av_get_channel_layout_string(buf2, sizeof(buf2),
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-1, outlink->channel_layout);
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av_log(outlink->src, AV_LOG_INFO,
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"fmt:%s cl:%s planar:%i -> fmt:%s cl:%s planar:%i\n",
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av_get_sample_fmt_name(inlink ->format), buf1, inlink ->planar,
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av_get_sample_fmt_name(outlink->format), buf2, outlink->planar);
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/* compute which channel layout conversion to use */
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if (inlink->channel_layout != outlink->channel_layout) {
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int i;
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for (i = 0; i < sizeof(rematrix_funcs); i++) {
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const struct RematrixFunctionInfo *f = &rematrix_funcs[i];
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if ((f->in_chlayout == 0 || f->in_chlayout == inlink ->channel_layout) &&
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(f->out_chlayout == 0 || f->out_chlayout == outlink->channel_layout) &&
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(f->planar == -1 || f->planar == inlink->planar) &&
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(f->sfmt == -1 || f->sfmt == inlink->format)
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) {
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aconvert->convert_chlayout = f->func;
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break;
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}
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}
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if (!aconvert->convert_chlayout) {
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av_log(outlink->src, AV_LOG_ERROR,
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"Unsupported channel layout conversion '%s -> %s' requested!\n",
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buf1, buf2);
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return AVERROR(EINVAL);
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}
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}
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return 0;
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}
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static int init_buffers(AVFilterLink *inlink, int nb_samples)
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{
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AConvertContext *aconvert = inlink->dst->priv;
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AVFilterLink * const outlink = inlink->dst->outputs[0];
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int i, packed_stride = 0;
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const unsigned
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packing_conv = inlink->planar != outlink->planar &&
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aconvert->out_nb_channels != 1,
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format_conv = inlink->format != outlink->format;
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int nb_channels = aconvert->out_nb_channels;
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uninit(inlink->dst);
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aconvert->max_nb_samples = nb_samples;
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if (aconvert->convert_chlayout) {
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/* allocate buffer for storing intermediary mixing samplesref */
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uint8_t *data[8];
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int linesize[8];
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int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
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if (av_samples_alloc(data, linesize, nb_channels, nb_samples,
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inlink->format, inlink->planar, 16) < 0)
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goto fail_no_mem;
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aconvert->mix_samplesref =
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avfilter_get_audio_buffer_ref_from_arrays(data, linesize, AV_PERM_WRITE,
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nb_samples, inlink->format,
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outlink->channel_layout,
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inlink->planar);
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if (!aconvert->mix_samplesref)
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goto fail_no_mem;
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}
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// if there's a format/packing conversion we need an audio_convert context
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if (format_conv || packing_conv) {
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aconvert->out_samplesref =
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avfilter_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
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if (!aconvert->out_samplesref)
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goto fail_no_mem;
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aconvert->in_strides [0] = av_get_bytes_per_sample(inlink ->format);
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aconvert->out_strides[0] = av_get_bytes_per_sample(outlink->format);
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aconvert->out_conv = aconvert->out_samplesref->data;
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if (aconvert->mix_samplesref)
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aconvert->in_conv = aconvert->mix_samplesref->data;
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if (packing_conv) {
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// packed -> planar
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if (outlink->planar == AVFILTER_PLANAR) {
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if (aconvert->mix_samplesref)
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aconvert->packed_data[0] = aconvert->mix_samplesref->data[0];
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aconvert->in_conv = aconvert->packed_data;
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packed_stride = aconvert->in_strides[0];
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aconvert->in_strides[0] *= nb_channels;
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// planar -> packed
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} else {
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aconvert->packed_data[0] = aconvert->out_samplesref->data[0];
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aconvert->out_conv = aconvert->packed_data;
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packed_stride = aconvert->out_strides[0];
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aconvert->out_strides[0] *= nb_channels;
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}
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} else if (outlink->planar == AVFILTER_PACKED) {
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/* If there's no packing conversion, and the stream is packed
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* then we treat the entire stream as one big channel
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*/
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nb_channels = 1;
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}
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for (i = 1; i < nb_channels; i++) {
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aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
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aconvert->in_strides[i] = aconvert->in_strides[0];
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aconvert->out_strides[i] = aconvert->out_strides[0];
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}
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aconvert->audioconvert_ctx =
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av_audio_convert_alloc(outlink->format, nb_channels,
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inlink->format, nb_channels, NULL, 0);
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if (!aconvert->audioconvert_ctx)
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goto fail_no_mem;
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}
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return 0;
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fail_no_mem:
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av_log(inlink->dst, AV_LOG_ERROR, "Could not allocate memory.\n");
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return AVERROR(ENOMEM);
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}
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static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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{
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AConvertContext *aconvert = inlink->dst->priv;
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AVFilterBufferRef *curbuf = insamplesref;
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AVFilterLink * const outlink = inlink->dst->outputs[0];
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int chan_mult;
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/* in/reinint the internal buffers if this is the first buffer
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* provided or it is needed to use a bigger one */
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if (!aconvert->max_nb_samples ||
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(curbuf->audio->nb_samples > aconvert->max_nb_samples))
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if (init_buffers(inlink, curbuf->audio->nb_samples) < 0) {
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av_log(inlink->dst, AV_LOG_ERROR, "Could not initialize buffers.\n");
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return;
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}
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/* if channel mixing is required */
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if (aconvert->mix_samplesref) {
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memcpy(aconvert->in_mix, curbuf->data, sizeof(aconvert->in_mix));
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memcpy(aconvert->out_mix, aconvert->mix_samplesref->data, sizeof(aconvert->out_mix));
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aconvert->convert_chlayout(aconvert->out_mix,
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aconvert->in_mix,
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curbuf->audio->nb_samples,
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aconvert);
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curbuf = aconvert->mix_samplesref;
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}
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if (aconvert->audioconvert_ctx) {
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if (!aconvert->mix_samplesref) {
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if (aconvert->in_conv == aconvert->packed_data) {
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int i, packed_stride = av_get_bytes_per_sample(inlink->format);
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aconvert->packed_data[0] = curbuf->data[0];
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for (i = 1; i < aconvert->out_nb_channels; i++)
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aconvert->packed_data[i] = aconvert->packed_data[i-1] + packed_stride;
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} else {
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aconvert->in_conv = curbuf->data;
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}
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}
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chan_mult = inlink->planar == outlink->planar && inlink->planar == 0 ?
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aconvert->out_nb_channels : 1;
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av_audio_convert(aconvert->audioconvert_ctx,
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(void * const *) aconvert->out_conv,
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aconvert->out_strides,
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(const void * const *) aconvert->in_conv,
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aconvert->in_strides,
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curbuf->audio->nb_samples * chan_mult);
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curbuf = aconvert->out_samplesref;
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}
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avfilter_copy_buffer_ref_props(curbuf, insamplesref);
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curbuf->audio->channel_layout = outlink->channel_layout;
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curbuf->audio->planar = outlink->planar;
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avfilter_filter_samples(inlink->dst->outputs[0],
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avfilter_ref_buffer(curbuf, ~0));
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avfilter_unref_buffer(insamplesref);
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}
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AVFilter avfilter_af_aconvert = {
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.name = "aconvert",
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.description = NULL_IF_CONFIG_SMALL("Convert the input audio to sample_fmt:channel_layout:packed_fmt."),
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.priv_size = sizeof(AConvertContext),
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.init = init,
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.uninit = uninit,
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.query_formats = query_formats,
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.inputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}},
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.outputs = (AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output, },
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{ .name = NULL}},
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};
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