ffmpeg/libavfilter/asrc_aevalsrc.c

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/*
* Copyright (c) 2011 Stefano Sabatini
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* eval audio source
*/
#include "libavutil/audioconvert.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/eval.h"
#include "libavutil/opt.h"
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#include "libavutil/parseutils.h"
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#include "avfilter.h"
#include "audio.h"
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#include "internal.h"
static const char * const var_names[] = {
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"n", ///< number of frame
"t", ///< timestamp expressed in seconds
"s", ///< sample rate
NULL
};
enum var_name {
VAR_N,
VAR_T,
VAR_S,
VAR_VARS_NB
};
typedef struct {
const AVClass *class;
char *sample_rate_str;
int sample_rate;
int64_t chlayout;
char *chlayout_str;
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int nb_channels;
int64_t pts;
AVExpr *expr[8];
char *expr_str[8];
int nb_samples; ///< number of samples per requested frame
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char *duration_str; ///< total duration of the generated audio
double duration;
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uint64_t n;
double var_values[VAR_VARS_NB];
} EvalContext;
#define OFFSET(x) offsetof(EvalContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption aevalsrc_options[]= {
{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.dbl = 1024}, 0, INT_MAX, FLAGS },
{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.dbl = 1024}, 0, INT_MAX, FLAGS },
{ "sample_rate", "set the sample rate", OFFSET(sample_rate_str), AV_OPT_TYPE_STRING, {.str = "44100"}, CHAR_MIN, CHAR_MAX, FLAGS },
{ "s", "set the sample rate", OFFSET(sample_rate_str), AV_OPT_TYPE_STRING, {.str = "44100"}, CHAR_MIN, CHAR_MAX, FLAGS },
{ "duration", "set audio duration", OFFSET(duration_str), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, FLAGS },
{ "d", "set audio duration", OFFSET(duration_str), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, FLAGS },
{ "channel_layout", "set channel layout", OFFSET(chlayout_str), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, FLAGS },
{ "c", "set channel layout", OFFSET(chlayout_str), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, FLAGS },
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{NULL},
};
AVFILTER_DEFINE_CLASS(aevalsrc);
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static int init(AVFilterContext *ctx, const char *args)
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{
EvalContext *eval = ctx->priv;
char *args1 = av_strdup(args);
char *expr, *buf, *bufptr;
int ret, i;
eval->class = &aevalsrc_class;
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av_opt_set_defaults(eval);
/* parse expressions */
buf = args1;
i = 0;
while (expr = av_strtok(buf, ":", &bufptr)) {
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ret = av_expr_parse(&eval->expr[i], expr, var_names,
NULL, NULL, NULL, NULL, 0, ctx);
if (ret < 0)
goto end;
i++;
if (bufptr && *bufptr == ':') { /* found last expression */
bufptr++;
break;
}
buf = NULL;
}
eval->nb_channels = i;
if (bufptr && (ret = av_set_options_string(eval, bufptr, "=", ":")) < 0)
goto end;
if (eval->chlayout_str) {
int n;
ret = ff_parse_channel_layout(&eval->chlayout, eval->chlayout_str, ctx);
if (ret < 0)
goto end;
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n = av_get_channel_layout_nb_channels(eval->chlayout);
if (n != eval->nb_channels) {
av_log(ctx, AV_LOG_ERROR,
"Mismatch between the specified number of channels '%d' "
"and the number of channels '%d' in the specified channel layout '%s'\n",
eval->nb_channels, n, eval->chlayout_str);
ret = AVERROR(EINVAL);
goto end;
}
} else {
/* guess channel layout from nb expressions/channels */
eval->chlayout = av_get_default_channel_layout(eval->nb_channels);
if (!eval->chlayout) {
av_log(ctx, AV_LOG_ERROR, "Invalid number of channels '%d' provided\n",
eval->nb_channels);
ret = AVERROR(EINVAL);
goto end;
}
}
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if ((ret = ff_parse_sample_rate(&eval->sample_rate, eval->sample_rate_str, ctx)))
goto end;
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eval->duration = -1;
if (eval->duration_str) {
int64_t us = -1;
if ((ret = av_parse_time(&us, eval->duration_str, 1)) < 0) {
av_log(ctx, AV_LOG_ERROR, "Invalid duration: '%s'\n", eval->duration_str);
goto end;
}
eval->duration = (double)us / 1000000;
}
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eval->n = 0;
end:
av_free(args1);
return ret;
}
static void uninit(AVFilterContext *ctx)
{
EvalContext *eval = ctx->priv;
int i;
for (i = 0; i < 8; i++) {
av_expr_free(eval->expr[i]);
eval->expr[i] = NULL;
}
av_freep(&eval->chlayout_str);
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av_freep(&eval->duration_str);
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av_freep(&eval->sample_rate_str);
}
static int config_props(AVFilterLink *outlink)
{
EvalContext *eval = outlink->src->priv;
char buf[128];
outlink->time_base = (AVRational){1, eval->sample_rate};
outlink->sample_rate = eval->sample_rate;
eval->var_values[VAR_S] = eval->sample_rate;
av_get_channel_layout_string(buf, sizeof(buf), 0, eval->chlayout);
av_log(outlink->src, AV_LOG_VERBOSE,
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"sample_rate:%d chlayout:%s duration:%f\n",
eval->sample_rate, buf, eval->duration);
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return 0;
}
static int query_formats(AVFilterContext *ctx)
{
EvalContext *eval = ctx->priv;
enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE };
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int64_t chlayouts[] = { eval->chlayout, -1 };
int sample_rates[] = { eval->sample_rate, -1 };
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ff_set_common_formats (ctx, ff_make_format_list(sample_fmts));
Merge remote-tracking branch 'qatar/master' * qatar/master: (26 commits) fate: use diff -b in oneline comparison Add missing version bumps and APIchanges/Changelog entries. lavfi: move buffer management function to a separate file. lavfi: move formats-related functions from default.c to formats.c lavfi: move video-related functions to a separate file. fate: make smjpeg a demux test fate: separate sierra-vmd audio and video tests fate: separate smacker audio and video tests libmp3lame: set supported channel layouts. avconv: automatically insert asyncts when -async is used. avconv: add support for audio filters. lavfi: add asyncts filter. lavfi: add aformat filter lavfi: add an audio buffer sink. lavfi: add an audio buffer source. buffersrc: add av_buffersrc_write_frame(). buffersrc: fix invalid read in uninit if the fifo hasn't been allocated lavfi: rename vsrc_buffer.c to buffersrc.c avfiltergraph: reindent lavfi: add channel layout/sample rate negotiation. ... Conflicts: Changelog doc/APIchanges doc/filters.texi ffmpeg.c ffprobe.c libavcodec/libmp3lame.c libavfilter/Makefile libavfilter/af_aformat.c libavfilter/allfilters.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/avfiltergraph.c libavfilter/buffersrc.c libavfilter/defaults.c libavfilter/formats.c libavfilter/src_buffer.c libavfilter/version.h libavfilter/vf_yadif.c libavfilter/vsrc_buffer.c libavfilter/vsrc_buffer.h libavutil/avutil.h tests/fate/audio.mak tests/fate/demux.mak tests/fate/video.mak Merged-by: Michael Niedermayer <michaelni@gmx.at>
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ff_set_common_channel_layouts(ctx, avfilter_make_format64_list(chlayouts));
ff_set_common_samplerates(ctx, ff_make_format_list(sample_rates));
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return 0;
}
static int request_frame(AVFilterLink *outlink)
{
EvalContext *eval = outlink->src->priv;
AVFilterBufferRef *samplesref;
int i, j;
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double t = eval->var_values[VAR_N] * (double)1/eval->sample_rate;
if (eval->duration >= 0 && t > eval->duration)
return AVERROR_EOF;
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Merge remote-tracking branch 'qatar/master' * qatar/master: (25 commits) rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC ape: Use unsigned integer maths arm: dsputil: fix overreads in put/avg_pixels functions h264: K&R formatting cosmetics for header files (part II/II) h264: K&R formatting cosmetics for header files (part I/II) rtmp: Implement check bandwidth notification. rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player. rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin. rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream. cmdutils: Add fallback case to switch in check_stream_specifier(). sctp: be consistent with socket option level configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags. vcr1enc: drop pointless empty encode_init() wrapper function vcr1: drop pointless write-only AVCodecContext member from VCR1Context vcr1: group encoder code together to save #ifdefs vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments mov: make one comment slightly more specific lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX lavfi: move audio-related functions to a separate file. lavfi: remove some audio-related function from public API. ... Conflicts: cmdutils.c libavcodec/h264.h libavcodec/h264_mvpred.h libavcodec/vcr1.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/defaults.c libavfilter/internal.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
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samplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, eval->nb_samples);
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/* evaluate expression for each single sample and for each channel */
for (i = 0; i < eval->nb_samples; i++, eval->n++) {
eval->var_values[VAR_N] = eval->n;
eval->var_values[VAR_T] = eval->var_values[VAR_N] * (double)1/eval->sample_rate;
for (j = 0; j < eval->nb_channels; j++) {
*((double *) samplesref->extended_data[j] + i) =
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av_expr_eval(eval->expr[j], eval->var_values, NULL);
}
}
samplesref->pts = eval->pts;
samplesref->pos = -1;
samplesref->audio->sample_rate = eval->sample_rate;
eval->pts += eval->nb_samples;
Merge remote-tracking branch 'qatar/master' * qatar/master: (25 commits) rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC ape: Use unsigned integer maths arm: dsputil: fix overreads in put/avg_pixels functions h264: K&R formatting cosmetics for header files (part II/II) h264: K&R formatting cosmetics for header files (part I/II) rtmp: Implement check bandwidth notification. rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player. rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin. rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream. cmdutils: Add fallback case to switch in check_stream_specifier(). sctp: be consistent with socket option level configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags. vcr1enc: drop pointless empty encode_init() wrapper function vcr1: drop pointless write-only AVCodecContext member from VCR1Context vcr1: group encoder code together to save #ifdefs vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments mov: make one comment slightly more specific lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX lavfi: move audio-related functions to a separate file. lavfi: remove some audio-related function from public API. ... Conflicts: cmdutils.c libavcodec/h264.h libavcodec/h264_mvpred.h libavcodec/vcr1.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/defaults.c libavfilter/internal.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
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ff_filter_samples(outlink, samplesref);
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return 0;
}
AVFilter avfilter_asrc_aevalsrc = {
.name = "aevalsrc",
.description = NULL_IF_CONFIG_SMALL("Generate an audio signal generated by an expression."),
.query_formats = query_formats,
.init = init,
.uninit = uninit,
.priv_size = sizeof(EvalContext),
.inputs = (const AVFilterPad[]) {{ .name = NULL}},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
.request_frame = request_frame, },
{ .name = NULL}},
.priv_class = &aevalsrc_class,
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};