[FFmpeg-devel] avfilter: add asr filter

Submitted by Paul B Mahol on May 5, 2019, 4:05 p.m.

Details

Message ID 20190505160504.5683-1-onemda@gmail.com
State New
Headers show

Commit Message

Paul B Mahol May 5, 2019, 4:05 p.m.
Signed-off-by: Paul B Mahol <onemda@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

Comments

Paul B Mahol May 11, 2019, 2:57 p.m.
On 5/5/19, Paul B Mahol <onemda@gmail.com> wrote:
> Signed-off-by: Paul B Mahol <onemda@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
>

Will apply.
Paul B Mahol May 12, 2019, 6:45 p.m.
On 5/11/19, Paul B Mahol <onemda@gmail.com> wrote:
> On 5/5/19, Paul B Mahol <onemda@gmail.com> wrote:
>> Signed-off-by: Paul B Mahol <onemda@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
>>
>
> Will apply.

Because nobody is against this will be applied in next 24h.

Patch hide | download patch | download mbox

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
 
+@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 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
 
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;