gstreamer/subprojects/gst-plugins-bad/ext/webrtcdsp/gstwebrtcechoprobe.cpp
Arun Raghavan d5755744c3 webrtcdsp: Update code for webrtc-audio-processing-1
Updated API usage appropriately, and now we have a versioned package to
track breaking vs. non-breaking updates.

Deprecates a number of properties (and we have to plug in our own values
for related enums which are now gone):

  * echo-suprression-level
  * experimental-agc
  * extended-filter
  * delay-agnostic
  * voice-detection-frame-size-ms
  * voice-detection-likelihood

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
2023-06-01 09:34:37 +00:00

474 lines
14 KiB
C++

/*
* WebRTC Audio Processing Elements
*
* Copyright 2016 Collabora Ltd
* @author: Nicolas Dufresne <nicolas.dufresne@collabora.com>
*
* This library 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.
*
* This library 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 this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
/**
* SECTION:element-webrtcechoprobe
*
* This echo probe is to be used with the webrtcdsp element. See #webrtcdsp
* documentation for more details.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstwebrtcechoprobe.h"
#include <modules/audio_processing/include/audio_processing.h>
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_EXTERN (webrtc_dsp_debug);
#define GST_CAT_DEFAULT (webrtc_dsp_debug)
#define MAX_ADAPTER_SIZE (1*1024*1024)
static GstStaticPadTemplate gst_webrtc_echo_probe_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX];"
"audio/x-raw, "
"format = (string) " GST_AUDIO_NE (F32) ", "
"layout = (string) non-interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX]")
);
static GstStaticPadTemplate gst_webrtc_echo_probe_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX];"
"audio/x-raw, "
"format = (string) " GST_AUDIO_NE (F32) ", "
"layout = (string) non-interleaved, "
"rate = (int) { 48000, 32000, 16000, 8000 }, "
"channels = (int) [1, MAX]")
);
G_LOCK_DEFINE_STATIC (gst_aec_probes);
static GList *gst_aec_probes = NULL;
G_DEFINE_TYPE (GstWebrtcEchoProbe, gst_webrtc_echo_probe,
GST_TYPE_AUDIO_FILTER);
GST_ELEMENT_REGISTER_DEFINE (webrtcechoprobe, "webrtcechoprobe",
GST_RANK_NONE, GST_TYPE_WEBRTC_ECHO_PROBE);
static gboolean
gst_webrtc_echo_probe_setup (GstAudioFilter * filter, const GstAudioInfo * info)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (filter);
GST_LOG_OBJECT (self, "setting format to %s with %i Hz and %i channels",
info->finfo->description, info->rate, info->channels);
GST_WEBRTC_ECHO_PROBE_LOCK (self);
self->info = *info;
self->interleaved = (info->layout == GST_AUDIO_LAYOUT_INTERLEAVED);
if (!self->interleaved)
gst_planar_audio_adapter_configure (self->padapter, info);
/* WebRTC library works with 10ms buffers, compute once this size */
self->period_samples = info->rate / 100;
self->period_size = self->period_samples * info->bpf;
if (self->interleaved &&
(MAX_DATA_SIZE_SAMPLES * 2) < self->period_size)
goto period_too_big;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return TRUE;
period_too_big:
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
GST_WARNING_OBJECT (self, "webrtcdsp format produce too big period "
"(maximum is %d samples and we have %u samples), "
"reduce the number of channels or the rate.",
MAX_DATA_SIZE_SAMPLES, self->period_size / 2);
return FALSE;
}
static gboolean
gst_webrtc_echo_probe_stop (GstBaseTransform * btrans)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (btrans);
GST_WEBRTC_ECHO_PROBE_LOCK (self);
gst_adapter_clear (self->adapter);
gst_planar_audio_adapter_clear (self->padapter);
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return TRUE;
}
static gboolean
gst_webrtc_echo_probe_src_event (GstBaseTransform * btrans, GstEvent * event)
{
GstBaseTransformClass *klass;
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (btrans);
GstClockTime latency;
GstClockTime upstream_latency = 0;
GstQuery *query;
klass = GST_BASE_TRANSFORM_CLASS (gst_webrtc_echo_probe_parent_class);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_LATENCY:
gst_event_parse_latency (event, &latency);
query = gst_query_new_latency ();
if (gst_pad_query (btrans->srcpad, query)) {
gst_query_parse_latency (query, NULL, &upstream_latency, NULL);
if (!GST_CLOCK_TIME_IS_VALID (upstream_latency))
upstream_latency = 0;
}
GST_WEBRTC_ECHO_PROBE_LOCK (self);
self->latency = latency;
self->delay = upstream_latency / GST_MSECOND;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
GST_DEBUG_OBJECT (self, "We have a latency of %" GST_TIME_FORMAT
" and delay of %ims", GST_TIME_ARGS (latency),
(gint) (upstream_latency / GST_MSECOND));
break;
default:
break;
}
return klass->src_event (btrans, event);
}
static GstFlowReturn
gst_webrtc_echo_probe_transform_ip (GstBaseTransform * btrans,
GstBuffer * buffer)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (btrans);
GstBuffer *newbuf = NULL;
GST_WEBRTC_ECHO_PROBE_LOCK (self);
newbuf = gst_buffer_copy (buffer);
/* Moves the buffer timestamp to be in Running time */
GST_BUFFER_PTS (newbuf) = gst_segment_to_running_time (&btrans->segment,
GST_FORMAT_TIME, GST_BUFFER_PTS (buffer));
if (self->interleaved) {
gst_adapter_push (self->adapter, newbuf);
if (gst_adapter_available (self->adapter) > MAX_ADAPTER_SIZE)
gst_adapter_flush (self->adapter,
gst_adapter_available (self->adapter) - MAX_ADAPTER_SIZE);
} else {
gsize available;
gst_planar_audio_adapter_push (self->padapter, newbuf);
available =
gst_planar_audio_adapter_available (self->padapter) * self->info.bpf;
if (available > MAX_ADAPTER_SIZE)
gst_planar_audio_adapter_flush (self->padapter,
(available - MAX_ADAPTER_SIZE) / self->info.bpf);
}
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return GST_FLOW_OK;
}
static void
gst_webrtc_echo_probe_finalize (GObject * object)
{
GstWebrtcEchoProbe *self = GST_WEBRTC_ECHO_PROBE (object);
G_LOCK (gst_aec_probes);
gst_aec_probes = g_list_remove (gst_aec_probes, self);
G_UNLOCK (gst_aec_probes);
gst_object_unref (self->adapter);
gst_object_unref (self->padapter);
self->adapter = NULL;
self->padapter = NULL;
G_OBJECT_CLASS (gst_webrtc_echo_probe_parent_class)->finalize (object);
}
static void
gst_webrtc_echo_probe_init (GstWebrtcEchoProbe * self)
{
self->adapter = gst_adapter_new ();
self->padapter = gst_planar_audio_adapter_new ();
gst_audio_info_init (&self->info);
g_mutex_init (&self->lock);
self->latency = GST_CLOCK_TIME_NONE;
G_LOCK (gst_aec_probes);
gst_aec_probes = g_list_prepend (gst_aec_probes, self);
G_UNLOCK (gst_aec_probes);
}
static void
gst_webrtc_echo_probe_class_init (GstWebrtcEchoProbeClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseTransformClass *btrans_class = GST_BASE_TRANSFORM_CLASS (klass);
GstAudioFilterClass *audiofilter_class = GST_AUDIO_FILTER_CLASS (klass);
gobject_class->finalize = gst_webrtc_echo_probe_finalize;
btrans_class->passthrough_on_same_caps = TRUE;
btrans_class->src_event = GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_src_event);
btrans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_transform_ip);
btrans_class->stop = GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_stop);
audiofilter_class->setup = GST_DEBUG_FUNCPTR (gst_webrtc_echo_probe_setup);
gst_element_class_add_static_pad_template (element_class,
&gst_webrtc_echo_probe_src_template);
gst_element_class_add_static_pad_template (element_class,
&gst_webrtc_echo_probe_sink_template);
gst_element_class_set_static_metadata (element_class,
"Acoustic Echo Canceller probe",
"Generic/Audio",
"Gathers playback buffers for webrtcdsp",
"Nicolas Dufresne <nicolas.dufrsesne@collabora.com>");
}
GstWebrtcEchoProbe *
gst_webrtc_acquire_echo_probe (const gchar * name)
{
GstWebrtcEchoProbe *ret = NULL;
GList *l;
G_LOCK (gst_aec_probes);
for (l = gst_aec_probes; l; l = l->next) {
GstWebrtcEchoProbe *probe = GST_WEBRTC_ECHO_PROBE (l->data);
GST_WEBRTC_ECHO_PROBE_LOCK (probe);
if (!probe->acquired && g_strcmp0 (GST_OBJECT_NAME (probe), name) == 0) {
probe->acquired = TRUE;
ret = GST_WEBRTC_ECHO_PROBE (gst_object_ref (probe));
GST_WEBRTC_ECHO_PROBE_UNLOCK (probe);
break;
}
GST_WEBRTC_ECHO_PROBE_UNLOCK (probe);
}
G_UNLOCK (gst_aec_probes);
return ret;
}
void
gst_webrtc_release_echo_probe (GstWebrtcEchoProbe * probe)
{
GST_WEBRTC_ECHO_PROBE_LOCK (probe);
probe->acquired = FALSE;
GST_WEBRTC_ECHO_PROBE_UNLOCK (probe);
gst_object_unref (probe);
}
gint
gst_webrtc_echo_probe_read (GstWebrtcEchoProbe * self, GstClockTime rec_time,
GstBuffer ** buf)
{
GstClockTimeDiff diff;
gsize avail, skip, offset, size = 0;
gint delay = -1;
GST_WEBRTC_ECHO_PROBE_LOCK (self);
/* We always return a buffer -- if don't have data (size == 0), we generate a
* silence buffer */
if (!GST_CLOCK_TIME_IS_VALID (self->latency) ||
!GST_AUDIO_INFO_IS_VALID (&self->info))
goto copy;
if (self->interleaved)
avail = gst_adapter_available (self->adapter) / self->info.bpf;
else
avail = gst_planar_audio_adapter_available (self->padapter);
/* In delay agnostic mode, just return 10ms of data */
if (!GST_CLOCK_TIME_IS_VALID (rec_time)) {
if (avail < self->period_samples)
goto copy;
size = self->period_samples;
skip = 0;
offset = 0;
goto copy;
}
if (avail == 0) {
diff = G_MAXINT64;
} else {
GstClockTime play_time;
guint64 distance;
if (self->interleaved) {
play_time = gst_adapter_prev_pts (self->adapter, &distance);
distance /= self->info.bpf;
} else {
play_time = gst_planar_audio_adapter_prev_pts (self->padapter, &distance);
}
if (GST_CLOCK_TIME_IS_VALID (play_time)) {
play_time += gst_util_uint64_scale_int (distance, GST_SECOND,
self->info.rate);
play_time += self->latency;
diff = GST_CLOCK_DIFF (rec_time, play_time) / GST_MSECOND;
} else {
/* We have no timestamp, assume perfect delay */
diff = self->delay;
}
}
if (diff > self->delay) {
skip = (diff - self->delay) * self->info.rate / 1000;
skip = MIN (self->period_samples, skip);
offset = 0;
} else {
skip = 0;
offset = (self->delay - diff) * self->info.rate / 1000;
offset = MIN (avail, offset);
}
size = MIN (avail - offset, self->period_samples - skip);
copy:
if (!size) {
/* No data, provide a period's worth of silence */
*buf = gst_buffer_new_allocate (NULL, self->period_size, NULL);
gst_buffer_memset (*buf, 0, 0, self->period_size);
gst_buffer_add_audio_meta (*buf, &self->info, self->period_samples,
NULL);
} else {
/* We have some actual data, pop period_samples' worth if have it, else pad
* with silence and provide what we do have */
GstBuffer *ret, *taken, *tmp;
if (self->interleaved) {
skip *= self->info.bpf;
offset *= self->info.bpf;
size *= self->info.bpf;
gst_adapter_flush (self->adapter, offset);
/* we need to fill silence at the beginning and/or the end of the
* buffer in order to have period_samples in the buffer */
if (size < self->period_size) {
gsize padding = self->period_size - (skip + size);
taken = gst_adapter_take_buffer (self->adapter, size);
ret = gst_buffer_new ();
/* need some silence at the beginning */
if (skip) {
tmp = gst_buffer_new_allocate (NULL, skip, NULL);
gst_buffer_memset (tmp, 0, 0, skip);
ret = gst_buffer_append (ret, tmp);
}
ret = gst_buffer_append (ret, taken);
/* need some silence at the end */
if (padding) {
tmp = gst_buffer_new_allocate (NULL, padding, NULL);
gst_buffer_memset (tmp, 0, 0, padding);
ret = gst_buffer_append (ret, tmp);
}
} else {
ret = gst_adapter_take_buffer (self->adapter, size);
}
} else {
gst_planar_audio_adapter_flush (self->padapter, offset);
/* we need to fill silence at the beginning and/or the end of each
* channel plane in order to have exactly period_samples in the buffer */
if (size < self->period_samples) {
GstAudioMeta *meta;
gint bps = self->info.finfo->width / 8;
gsize padding = self->period_samples - (skip + size);
gint c;
taken = gst_planar_audio_adapter_take_buffer (self->padapter, size,
GST_MAP_READ);
meta = gst_buffer_get_audio_meta (taken);
ret = gst_buffer_new ();
for (c = 0; c < meta->info.channels; c++) {
/* need some silence at the beginning */
if (skip) {
tmp = gst_buffer_new_allocate (NULL, skip * bps, NULL);
gst_buffer_memset (tmp, 0, 0, skip * bps);
ret = gst_buffer_append (ret, tmp);
}
tmp = gst_buffer_copy_region (taken, GST_BUFFER_COPY_MEMORY,
meta->offsets[c], size * bps);
ret = gst_buffer_append (ret, tmp);
/* need some silence at the end */
if (padding) {
tmp = gst_buffer_new_allocate (NULL, padding * bps, NULL);
gst_buffer_memset (tmp, 0, 0, padding * bps);
ret = gst_buffer_append (ret, tmp);
}
}
gst_buffer_unref (taken);
gst_buffer_add_audio_meta (ret, &self->info, self->period_samples,
NULL);
} else {
ret = gst_planar_audio_adapter_take_buffer (self->padapter, size,
GST_MAP_READWRITE);
}
}
*buf = ret;
}
delay = self->delay;
GST_WEBRTC_ECHO_PROBE_UNLOCK (self);
return delay;
}