Welcome to mirror list, hosted at ThFree Co, Russian Federation.

github.com/FFmpeg/FFmpeg.git - Unnamed repository; edit this file 'description' to name the repository.
summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
authorPaul B Mahol <onemda@gmail.com>2019-05-05 16:01:53 +0300
committerPaul B Mahol <onemda@gmail.com>2019-05-14 16:17:14 +0300
commitf49cec2ba8830dd8df0ac73a39c118f6e20b06bd (patch)
tree8e2c26aa815ac1c989b13b76cf66acf95b4fe7bd /libavfilter/af_asr.c
parent670251de56cdcda0c32d588959c8ed2da09075a2 (diff)
avfilter: add asr filter
Diffstat (limited to 'libavfilter/af_asr.c')
-rw-r--r--libavfilter/af_asr.c181
1 files changed, 181 insertions, 0 deletions
diff --git a/libavfilter/af_asr.c b/libavfilter/af_asr.c
new file mode 100644
index 0000000000..0c08df1356
--- /dev/null
+++ b/libavfilter/af_asr.c
@@ -0,0 +1,181 @@
+/*
+ * 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,
+};