Commit f49cec2b authored by Paul B Mahol's avatar Paul B Mahol

avfilter: add asr filter

parent 670251de
......@@ -29,6 +29,7 @@ version <next>:
- Support decoding of HEVC 4:4:4 content in vdpau
- colorhold filter
- xmedian filter
- asr filter
version 4.1:
......
......@@ -307,6 +307,7 @@ External library support:
--enable-opengl enable OpenGL rendering [no]
--enable-openssl enable openssl, needed for https support
if gnutls, libtls or mbedtls is not used [no]
--enable-pocketsphinx enable PocketSphinx, needed for asr filter [no]
--disable-sndio disable sndio support [autodetect]
--disable-schannel disable SChannel SSP, needed for TLS support on
Windows if openssl and gnutls are not used [autodetect]
......@@ -1799,6 +1800,7 @@ EXTERNAL_LIBRARY_LIST="
mediacodec
openal
opengl
pocketsphinx
vapoursynth
"
......@@ -3400,6 +3402,7 @@ afir_filter_deps="avcodec"
afir_filter_select="fft"
amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
asr_filter_deps="pocketsphinx"
ass_filter_deps="libass"
atempo_filter_deps="avcodec"
atempo_filter_select="rdft"
......@@ -6298,6 +6301,7 @@ enabled openssl && { check_pkg_config openssl openssl openssl/ssl.h OP
check_lib openssl openssl/ssl.h SSL_library_init -lssl32 -leay32 ||
check_lib openssl openssl/ssl.h SSL_library_init -lssl -lcrypto -lws2_32 -lgdi32 ||
die "ERROR: openssl not found"; }
enabled pocketsphinx && require_pkg_config pocketsphinx pocketsphinx pocketsphinx/pocketsphinx.h ps_init
enabled rkmpp && { require_pkg_config rkmpp rockchip_mpp rockchip/rk_mpi.h mpp_create &&
require_pkg_config rockchip_mpp "rockchip_mpp >= 1.3.7" rockchip/rk_mpi.h mpp_create &&
{ enabled libdrm ||
......
......@@ -2131,6 +2131,41 @@ It accepts the following values:
Set additional parameter which controls sigmoid function.
@end table
@section asr
Automatic Speech Recognition
This filter uses PocketSphinx for speech recognition. To enable
compilation of this filter, you need to configure FFmpeg with
@code{--enable-pocketsphinx}.
It accepts the following options:
@table @option
@item rate
Set sampling rate of input audio. Defaults is @code{16000}.
This need to match speech models, otherwise one will get poor results.
@item hmm
Set dictionary containing acoustic model files.
@item dict
Set pronunciation dictionary.
@item lm
Set language model file.
@item lmctl
Set language model set.
@item lmname
Set which language model to use.
@item logfn
Set output for log messages.
@end table
The filter exports recognized speech as the frame metadata @code{lavfi.asr.text}.
@anchor{astats}
@section astats
......
......@@ -82,6 +82,7 @@ OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASIDEDATA_FILTER) += f_sidedata.o
OBJS-$(CONFIG_ASOFTCLIP_FILTER) += af_asoftclip.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o
OBJS-$(CONFIG_ASR_FILTER) += af_asr.o
OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o
OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o
OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o
......
/*
* Copyright (c) 2019 Paul B Mahol
*
* 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
*/
#include <pocketsphinx/pocketsphinx.h>
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct ASRContext {
const AVClass *class;
int rate;
char *hmm;
char *dict;
char *lm;
char *lmctl;
char *lmname;
char *logfn;
ps_decoder_t *ps;
cmd_ln_t *config;
int utt_started;
} ASRContext;
#define OFFSET(x) offsetof(ASRContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
static const AVOption asr_options[] = {
{ "rate", "set sampling rate", OFFSET(rate), AV_OPT_TYPE_INT, {.i64=16000}, 0, INT_MAX, .flags = FLAGS },
{ "hmm", "set directory containing acoustic model files", OFFSET(hmm), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
{ "dict", "set pronunciation dictionary", OFFSET(dict), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
{ "lm", "set language model file", OFFSET(lm), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
{ "lmctl", "set language model set", OFFSET(lmctl), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
{ "lmname","set which language model to use", OFFSET(lmname), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
{ "logfn", "set output for log messages", OFFSET(logfn), AV_OPT_TYPE_STRING, {.str="/dev/null"}, .flags = FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(asr);
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVDictionary **metadata = &in->metadata;
ASRContext *s = ctx->priv;
int have_speech;
const char *speech;
ps_process_raw(s->ps, (const int16_t *)in->data[0], in->nb_samples, 0, 0);
have_speech = ps_get_in_speech(s->ps);
if (have_speech && !s->utt_started)
s->utt_started = 1;
if (!have_speech && s->utt_started) {
ps_end_utt(s->ps);
speech = ps_get_hyp(s->ps, NULL);
if (speech != NULL)
av_dict_set(metadata, "lavfi.asr.text", speech, 0);
ps_start_utt(s->ps);
s->utt_started = 0;
}
return ff_filter_frame(ctx->outputs[0], in);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
ASRContext *s = ctx->priv;
ps_start_utt(s->ps);
return 0;
}
static av_cold int asr_init(AVFilterContext *ctx)
{
ASRContext *s = ctx->priv;
const float frate = s->rate;
char *rate = av_asprintf("%f", frate);
const char *argv[] = { "-logfn", s->logfn,
"-hmm", s->hmm,
"-lm", s->lm,
"-lmctl", s->lmctl,
"-lmname", s->lmname,
"-dict", s->dict,
"-samprate", rate,
NULL };
s->config = cmd_ln_parse_r(NULL, ps_args(), 14, (char **)argv, 0);
av_free(rate);
if (!s->config)
return AVERROR(ENOMEM);
ps_default_search_args(s->config);
s->ps = ps_init(s->config);
if (!s->ps)
return AVERROR(ENOMEM);
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
ASRContext *s = ctx->priv;
int sample_rates[] = { s->rate, -1 };
int ret;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16 )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_MONO )) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
(ret = ff_set_common_samplerates (ctx , ff_make_format_list(sample_rates) )) < 0)
return ret;
return 0;
}
static av_cold void asr_uninit(AVFilterContext *ctx)
{
ASRContext *s = ctx->priv;
ps_free(s->ps);
s->ps = NULL;
cmd_ln_free_r(s->config);
s->config = NULL;
}
static const AVFilterPad asr_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
{ NULL }
};
static const AVFilterPad asr_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_asr = {
.name = "asr",
.description = NULL_IF_CONFIG_SMALL("Automatic Speech Recognition."),
.priv_size = sizeof(ASRContext),
.priv_class = &asr_class,
.init = asr_init,
.uninit = asr_uninit,
.query_formats = query_formats,
.inputs = asr_inputs,
.outputs = asr_outputs,
};
......@@ -74,6 +74,7 @@ extern AVFilter ff_af_ashowinfo;
extern AVFilter ff_af_asidedata;
extern AVFilter ff_af_asoftclip;
extern AVFilter ff_af_asplit;
extern AVFilter ff_af_asr;
extern AVFilter ff_af_astats;
extern AVFilter ff_af_astreamselect;
extern AVFilter ff_af_atempo;
......
......@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 52
#define LIBAVFILTER_VERSION_MINOR 53
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
......
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