gstreamer/subprojects/gst-plugins-bad/gst/audiolatency/gstaudiolatency.c

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/* Audio latency measurement plugin
* Copyright (C) 2018 Centricular Ltd.
* Author: Nirbheek Chauhan <nirbheek@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-audiolatency
* @title: audiolatency
*
* Measures the audio latency between the source pad and the sink pad by
* outputting period ticks on the source pad and measuring how long they take to
* arrive on the sink pad.
*
* The ticks have a period of 1 second, so this element can only measure
* latencies smaller than that.
*
* ## Example pipeline
* |[
* gst-launch-1.0 -v autoaudiosrc ! audiolatency print-latency=true ! autoaudiosink
* ]| Continuously print the latency of the audio output and the audio capture
*
* In this case, you must ensure that the audio output is captured by the audio
* source. The simplest way is to use a standard 3.5mm male-to-male audio cable
* to connect line-out to line-in, or speaker-out to mic-in, etc.
*
* Capturing speaker output with a microphone should also work, as long as the
* ambient noise level is low enough. You may have to adjust the microphone gain
* to ensure that the volume is loud enough to be detected by the element, and
* at the same time that it's not so loud that it picks up ambient noise.
*
* For programmatic use, instead of using 'print-stats', you should read the
* 'last-latency' and 'average-latency' properties at most once a second, or
* parse the "latency" element message, which contains the "last-latency" and
* "average-latency" fields in the GstStructure.
*
* The average latency is a running average of the last 5 measurements.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstaudiolatency.h"
#define AUDIOLATENCY_CAPS "audio/x-raw, " \
"format = (string) F32LE, " \
"layout = (string) interleaved, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ]"
GST_DEBUG_CATEGORY_STATIC (gst_audiolatency_debug);
#define GST_CAT_DEFAULT gst_audiolatency_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (AUDIOLATENCY_CAPS)
);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (AUDIOLATENCY_CAPS)
);
#define gst_audiolatency_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstAudioLatency, gst_audiolatency, GST_TYPE_BIN,
GST_DEBUG_CATEGORY_INIT (gst_audiolatency_debug, "audiolatency", 0,
"audiolatency"););
GST_ELEMENT_REGISTER_DEFINE (audiolatency, "audiolatency", GST_RANK_PRIMARY,
GST_TYPE_AUDIOLATENCY);
#define DEFAULT_PRINT_LATENCY FALSE
#define DEFAULT_SAMPLES_PER_BUFFER 240
enum
{
PROP_0,
PROP_PRINT_LATENCY,
PROP_LAST_LATENCY,
PROP_AVERAGE_LATENCY,
PROP_SAMPLES_PER_BUFFER,
};
static gint64 gst_audiolatency_get_latency (GstAudioLatency * self);
static gint64 gst_audiolatency_get_average_latency (GstAudioLatency * self);
static GstFlowReturn gst_audiolatency_sink_chain (GstPad * pad,
GstObject * parent, GstBuffer * buffer);
static gboolean gst_audiolatency_sink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static GstPadProbeReturn gst_audiolatency_src_probe (GstPad * pad,
GstPadProbeInfo * info, gpointer user_data);
static void
gst_audiolatency_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstAudioLatency *self = GST_AUDIOLATENCY (object);
switch (prop_id) {
case PROP_PRINT_LATENCY:
g_value_set_boolean (value, self->print_latency);
break;
case PROP_LAST_LATENCY:
g_value_set_int64 (value, gst_audiolatency_get_latency (self));
break;
case PROP_AVERAGE_LATENCY:
g_value_set_int64 (value, gst_audiolatency_get_average_latency (self));
break;
case PROP_SAMPLES_PER_BUFFER:
g_value_set_int (value, self->samples_per_buffer);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audiolatency_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstAudioLatency *self = GST_AUDIOLATENCY (object);
switch (prop_id) {
case PROP_PRINT_LATENCY:
self->print_latency = g_value_get_boolean (value);
break;
case PROP_SAMPLES_PER_BUFFER:
self->samples_per_buffer = g_value_get_int (value);
g_object_set (self->audiosrc,
"samplesperbuffer", self->samples_per_buffer, NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audiolatency_class_init (GstAudioLatencyClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gobject_class->get_property = gst_audiolatency_get_property;
gobject_class->set_property = gst_audiolatency_set_property;
g_object_class_install_property (gobject_class, PROP_PRINT_LATENCY,
g_param_spec_boolean ("print-latency", "Print latencies",
"Print the measured latencies on stdout",
DEFAULT_PRINT_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LAST_LATENCY,
g_param_spec_int64 ("last-latency", "Last measured latency",
"The last latency that was measured, in microseconds", 0,
G_USEC_PER_SEC, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_AVERAGE_LATENCY,
g_param_spec_int64 ("average-latency", "Running average latency",
"The running average latency, in microseconds", 0,
G_USEC_PER_SEC, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstAudioLatency:samplesperbuffer:
*
* The number of audio samples in each outgoing buffer.
* See also #GstAudioTestSrc:samplesperbuffer
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class, PROP_SAMPLES_PER_BUFFER,
g_param_spec_int ("samplesperbuffer", "Samples per buffer",
"Number of samples in each outgoing buffer",
1, G_MAXINT, DEFAULT_SAMPLES_PER_BUFFER,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
gst_element_class_set_static_metadata (gstelement_class, "AudioLatency",
"Audio/Util",
"Measures the audio latency between the source and the sink",
"Nirbheek Chauhan <nirbheek@centricular.com>");
}
static void
gst_audiolatency_init (GstAudioLatency * self)
{
GstPad *srcpad;
GstPadTemplate *templ;
self->send_pts = 0;
self->recv_pts = 0;
self->print_latency = DEFAULT_PRINT_LATENCY;
self->samples_per_buffer = DEFAULT_SAMPLES_PER_BUFFER;
/* Setup sinkpad */
self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_audiolatency_sink_chain));
gst_pad_set_event_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_audiolatency_sink_event));
gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
/* Setup srcpad */
self->audiosrc = gst_element_factory_make ("audiotestsrc", NULL);
g_object_set (self->audiosrc, "wave", 8, "samplesperbuffer",
DEFAULT_SAMPLES_PER_BUFFER, "is-live", TRUE, NULL);
gst_bin_add (GST_BIN (self), self->audiosrc);
templ = gst_static_pad_template_get (&src_template);
srcpad = gst_element_get_static_pad (self->audiosrc, "src");
gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BUFFER,
(GstPadProbeCallback) gst_audiolatency_src_probe, self, NULL);
self->srcpad = gst_ghost_pad_new_from_template ("src", srcpad, templ);
gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
gst_object_unref (srcpad);
gst_object_unref (templ);
GST_LOG_OBJECT (self, "Initialized audiolatency");
}
static gint64
gst_audiolatency_get_latency (GstAudioLatency * self)
{
gint64 last_latency;
gint last_latency_idx;
GST_OBJECT_LOCK (self);
/* Decrement index, with wrap-around */
last_latency_idx = self->next_latency_idx - 1;
if (last_latency_idx < 0)
last_latency_idx = GST_AUDIOLATENCY_NUM_LATENCIES - 1;
last_latency = self->latencies[last_latency_idx];
GST_OBJECT_UNLOCK (self);
return last_latency;
}
static gint64
gst_audiolatency_get_average_latency_unlocked (GstAudioLatency * self)
{
int ii, n = 0;
gint64 average = 0;
for (ii = 0; ii < GST_AUDIOLATENCY_NUM_LATENCIES; ii++) {
if (G_LIKELY (self->latencies[ii] > 0))
n += 1;
average += self->latencies[ii];
}
return average / MAX (n, 1);
}
static gint64
gst_audiolatency_get_average_latency (GstAudioLatency * self)
{
gint64 average;
GST_OBJECT_LOCK (self);
average = gst_audiolatency_get_average_latency_unlocked (self);
GST_OBJECT_UNLOCK (self);
return average;
}
static void
gst_audiolatency_set_latency (GstAudioLatency * self, gint64 latency)
{
gint64 avg_latency;
GST_OBJECT_LOCK (self);
self->latencies[self->next_latency_idx] = latency;
/* Increment index, with wrap-around */
self->next_latency_idx += 1;
if (self->next_latency_idx > GST_AUDIOLATENCY_NUM_LATENCIES - 1)
self->next_latency_idx = 0;
avg_latency = gst_audiolatency_get_average_latency_unlocked (self);
if (self->print_latency)
g_print ("last latency: %" G_GINT64_FORMAT "ms, running average: %"
G_GINT64_FORMAT "ms\n", latency / 1000, avg_latency / 1000);
GST_OBJECT_UNLOCK (self);
/* Post an element message about it */
gst_element_post_message (GST_ELEMENT (self),
gst_message_new_element (GST_OBJECT (self),
gst_structure_new ("latency", "last-latency", G_TYPE_INT64, latency,
"average-latency", G_TYPE_INT64, avg_latency, NULL)));
}
static gint64
buffer_has_wave (GstBuffer * buffer, GstPad * pad)
{
const GstStructure *s;
GstCaps *caps;
GstMapInfo minfo;
guint64 duration;
gint64 offset;
gint ii, channels, fsize, rate;
gfloat *fdata;
gboolean ret;
GstMemory *memory = NULL;
switch (gst_buffer_n_memory (buffer)) {
case 0:
GST_WARNING_OBJECT (pad, "buffer %" GST_PTR_FORMAT "has no memory?",
buffer);
return -1;
case 1:
memory = gst_buffer_peek_memory (buffer, 0);
ret = gst_memory_map (memory, &minfo, GST_MAP_READ);
break;
default:
ret = gst_buffer_map (buffer, &minfo, GST_MAP_READ);
}
if (!ret) {
GST_WARNING_OBJECT (pad, "failed to map buffer %" GST_PTR_FORMAT, buffer);
return -1;
}
caps = gst_pad_get_current_caps (pad);
s = gst_caps_get_structure (caps, 0);
/* channels and rate are required in caps, so will always be present */
gst_structure_get_int (s, "channels", &channels);
gst_structure_get_int (s, "rate", &rate);
gst_caps_unref (caps);
fdata = (gfloat *) minfo.data;
fsize = minfo.size / sizeof (gfloat);
offset = -1;
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
duration = GST_BUFFER_DURATION (buffer);
} else {
/* Cannot do a rounding-accurate duration calculation here because in the
* case when the duration is invalid, the pts might also be invalid */
duration = gst_util_uint64_scale_int_round (GST_SECOND, fsize / channels,
rate);
GST_LOG_OBJECT (pad, "buffer duration is invalid, calculated likely "
"duration as %" G_GINT64_FORMAT "us", duration / GST_USECOND);
}
/* Read only one channel */
for (ii = 1; ii < fsize; ii += channels) {
if (ABS (fdata[ii]) > 0.7) {
/* The waveform probably starts somewhere inside the buffer,
* so get the offset in nanoseconds from the buffer pts */
offset = gst_util_uint64_scale_int_round (duration, ii, fsize);
break;
}
}
if (memory)
gst_memory_unmap (memory, &minfo);
else
gst_buffer_unmap (buffer, &minfo);
/* Return offset in microseconds */
return (offset > 0) ? offset / 1000 : -1;
}
static GstPadProbeReturn
gst_audiolatency_src_probe (GstPad * pad, GstPadProbeInfo * info,
gpointer user_data)
{
GstAudioLatency *self = user_data;
GstBuffer *buffer;
gint64 pts, offset;
if (!(info->type & GST_PAD_PROBE_TYPE_BUFFER))
goto out;
if (GST_STATE (self) != GST_STATE_PLAYING)
goto out;
GST_TRACE ("audiotestsrc pushed out a buffer");
pts = g_get_monotonic_time ();
/* Ticks are once a second, so once we send something, we can skip
* checking ~1sec of buffers till the next one. */
if (self->send_pts > 0 && pts - self->send_pts <= 950 * 1000)
goto out;
/* Check if buffer contains a waveform */
buffer = gst_pad_probe_info_get_buffer (info);
offset = buffer_has_wave (buffer, pad);
if (offset < 0)
goto out;
pts -= offset;
{
gint64 after = 0;
if (self->send_pts > 0)
after = (pts - self->send_pts) / 1000;
GST_INFO ("send pts: %" G_GINT64_FORMAT "us (after %" G_GINT64_FORMAT
"ms, offset %" G_GINT64_FORMAT "ms)", pts, after, offset / 1000);
}
self->send_pts = pts + offset;
out:
return GST_PAD_PROBE_OK;
}
static GstFlowReturn
gst_audiolatency_sink_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstAudioLatency *self = GST_AUDIOLATENCY (parent);
gint64 latency, offset, pts;
/* Ignore buffers till something gets sent out by us. Fixes a bug where we'd
* start out by printing one garbage latency value on Windows. */
if (self->send_pts == 0)
goto out;
GST_TRACE_OBJECT (pad, "Got buffer %p", buffer);
pts = g_get_monotonic_time ();
/* Ticks are once a second, so once we receive something, we can skip
* checking ~1sec of buffers till the next one. This way we also don't count
* the same tick twice for latency measurement. */
if (self->recv_pts > 0 && pts - self->recv_pts <= 950 * 1000)
goto out;
offset = buffer_has_wave (buffer, pad);
if (offset < 0)
goto out;
self->recv_pts = pts + offset;
latency = (self->recv_pts - self->send_pts);
gst_audiolatency_set_latency (self, latency);
GST_INFO ("recv pts: %" G_GINT64_FORMAT "us, latency: %" G_GINT64_FORMAT
"ms, offset: %" G_GINT64_FORMAT "ms", self->recv_pts, latency / 1000,
offset / 1000);
out:
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static gboolean
gst_audiolatency_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
switch (GST_EVENT_TYPE (event)) {
/* Drop below events. audiotestsrc will push its own event */
case GST_EVENT_STREAM_START:
case GST_EVENT_CAPS:
case GST_EVENT_SEGMENT:
gst_event_unref (event);
return TRUE;
default:
break;
}
return gst_pad_event_default (pad, parent, event);
}
/* Element registration */
static gboolean
plugin_init (GstPlugin * plugin)
{
return GST_ELEMENT_REGISTER (audiolatency, plugin);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audiolatency,
"A plugin to measure audio latency",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)