[FFmpeg-devel] [PATCH] avfilter: add asr filter
Paul B Mahol
onemda at gmail.com
Sun May 5 19:05:04 EEST 2019
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
configure | 4 +
doc/filters.texi | 32 +++++++
libavfilter/Makefile | 1 +
libavfilter/af_asr.c | 177 +++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
5 files changed, 215 insertions(+)
create mode 100644 libavfilter/af_asr.c
diff --git a/configure b/configure
index d644a5b1d4..586c293bb9 100755
--- a/configure
+++ b/configure
@@ -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
"
@@ -3401,6 +3403,7 @@ 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"
@@ -6299,6 +6302,7 @@ 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 ||
diff --git a/doc/filters.texi b/doc/filters.texi
index 3c15bb95f4..3f25d12511 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2131,6 +2131,38 @@ It accepts the following values:
Set additional parameter which controls sigmoid function.
@end table
+ at section asr
+Automatic Speech Recognition
+
+This filter uses PocketSphinX for speech recognition. To enable
+compilation of this filter, you need to configure FFmpeg with
+ at code{--enable-pocketsphinx}.
+
+It accepts the following options:
+
+ at table @option
+ at item rate
+Set sampling rate of input audio. Defaults is @code{16000}.
+This need to match speech models, otherwise one will get poor results.
+
+ at item dict
+Set pronunciation dictionary.
+
+ at item lm
+Set language model file.
+
+ at item lmctl
+Set language model set.
+
+ at item lmname
+Set which language model to use.
+
+ at item logfn
+Set output for log messages.
+ at end table
+
+The filter exports recognized speech as the frame metadata @code{lavfi.asr.text}.
+
@anchor{astats}
@section astats
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 59d12ce069..cf12365c8d 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
diff --git a/libavfilter/af_asr.c b/libavfilter/af_asr.c
new file mode 100644
index 0000000000..f14822215c
--- /dev/null
+++ b/libavfilter/af_asr.c
@@ -0,0 +1,177 @@
+/*
+ * 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 *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 },
+ { "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 init(AVFilterContext *ctx)
+{
+ ASRContext *s = ctx->priv;
+ const float frate = s->rate;
+ const char *rate = av_asprintf("%f", frate);
+ const char *argv[] = { "-logfn", s->logfn,
+ "-lm", s->lm,
+ "-lmctl", s->lmctl,
+ "-lmname",s->lmname,
+ "-dict", s->dict,
+ "-samprate", rate,
+ NULL };
+
+ s->config = cmd_ln_parse_r(NULL, ps_args(), 12, (char **)argv, TRUE);
+ 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 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 = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = asr_inputs,
+ .outputs = asr_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ae725cb0e0..fcbf50120b 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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;
--
2.17.1
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