[pulseaudio-discuss] [RFC] [PATCH] tests: Add a latency measurement test
Arun Raghavan
arun.raghavan at collabora.co.uk
Tue May 21 06:20:13 PDT 2013
This test is intended to measure real latency by playing a sample to a
sink and capturing that over a loopback interface. The loopback can
either be physical (cable running from headphone out to line in) or
virtual (monitor source or module loopback).
Also included in this is calibration code to make sure that volumes are
sufficiently adjusted to be able to detect the played back signal (and
that there aren't false positives due to line noise).
One of the objectives of all this is to later factor out the setup code
to allow us to easily write more loopback tests for various
functionality (volumes, resampling, mixing, etc.).
(I'll probably refactor before I push this out, but I thought it would be good
to get some feedback. The reported latency from the test seems to approximately
match requested latency, though it's generally a bit lower.)
---
src/Makefile.am | 8 +-
src/tests/lo-latency-test.c | 443 ++++++++++++++++++++++++++++++++++++++++++++
2 files changed, 450 insertions(+), 1 deletion(-)
create mode 100644 src/tests/lo-latency-test.c
diff --git a/src/Makefile.am b/src/Makefile.am
index a621a30..163976e 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -261,7 +261,8 @@ TESTS_norun = \
rtstutter \
sig2str-test \
stripnul \
- echo-cancel-test
+ echo-cancel-test \
+ lo-latency-test
# These tests need a running pulseaudio daemon
TESTS_daemon = \
@@ -574,6 +575,11 @@ echo_cancel_test_CXXFLAGS = $(module_echo_cancel_la_CXXFLAGS) -DECHO_CANCEL_TEST
endif
echo_cancel_test_LDFLAGS = $(AM_LDFLAGS) $(BINLDFLAGS)
+lo_latency_test_SOURCES = tests/lo-latency-test.c
+lo_latency_test_LDADD = $(AM_LDADD) libpulse.la
+lo_latency_test_CFLAGS = $(AM_CFLAGS) $(LIBCHECK_CFLAGS)
+lo_latency_test_LDFLAGS = $(AM_LDFLAGS) $(BINLDFLAGS) $(LIBCHECK_LIBS)
+
###################################
# Common library #
###################################
diff --git a/src/tests/lo-latency-test.c b/src/tests/lo-latency-test.c
new file mode 100644
index 0000000..837639a
--- /dev/null
+++ b/src/tests/lo-latency-test.c
@@ -0,0 +1,443 @@
+/***
+ This file is part of PulseAudio.
+
+ PulseAudio 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.
+
+ PulseAudio 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
+ General Public License for more details.
+
+ You should have received a copy of the GNU Lesser General Public License
+ along with PulseAudio; if not, write to the Free Software
+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ USA.
+***/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <fcntl.h>
+
+#include <errno.h>
+#include <unistd.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+
+#include <check.h>
+
+#include <pulse/pulseaudio.h>
+#include <pulse/mainloop.h>
+
+#define SAMPLE_HZ 44100
+#define CHANNELS 2
+#define N_OUT (SAMPLE_HZ * 1)
+
+#define TONE_HZ SAMPLE_HZ / 100
+#define PLAYBACK_LATENCY 25 /* ms */
+#define CAPTURE_LATENCY 5 /* ms */
+
+static pa_context *context = NULL;
+static pa_stream *pstream, *rstream;
+static pa_mainloop_api *mainloop_api = NULL;
+static const char *bname = NULL;
+
+static float out[N_OUT][CHANNELS];
+static int ppos = 0;
+
+static int n_underflow = 0;
+static int n_overflow = 0;
+
+static struct timeval tv_out, tv_in;
+
+static const pa_sample_spec sample_spec = {
+ .format = PA_SAMPLE_FLOAT32,
+ .rate = SAMPLE_HZ,
+ .channels = CHANNELS,
+};
+static int ss, fs;
+
+static void nop_free_cb(void *p) {}
+
+static void underflow_cb(struct pa_stream *s, void *userdata) {
+ fprintf(stderr, "Underflow\n");
+ n_underflow++;
+}
+
+static void overflow_cb(struct pa_stream *s, void *userdata) {
+ fprintf(stderr, "Overlow\n");
+ n_overflow++;
+}
+
+static void write_cb(pa_stream *s, size_t nbytes, void *userdata) {
+ int r, nsamp = nbytes / fs;
+
+ if (ppos + nsamp > N_OUT) {
+ r = pa_stream_write(s, &out[ppos][0], (N_OUT - ppos) * fs, nop_free_cb, 0, PA_SEEK_RELATIVE);
+ nbytes -= (N_OUT - ppos) * fs;
+ ppos = 0;
+ }
+
+ if (ppos == 0)
+ pa_gettimeofday(&tv_out);
+
+ r = pa_stream_write(s, &out[ppos][0], nbytes, nop_free_cb, 0, PA_SEEK_RELATIVE);
+ fail_unless(r == 0);
+
+ ppos = (ppos + nbytes / fs) % N_OUT;
+}
+
+static inline float rms(const float *s, int n) {
+ float sq = 0;
+ int i;
+
+ for (i = 0; i < n; i++)
+ sq += s[i] * s[i];
+
+ return sqrt(sq / n);
+}
+
+#define WINDOW 2 * CHANNELS
+
+static void read_cb(pa_stream *s, size_t nbytes, void *userdata) {
+ static float last = 0.0;
+ const float *in;
+ float cur;
+ int r;
+ unsigned int i = 0;
+ size_t l;
+
+ r = pa_stream_peek(s, (const void **)&in, &l);
+ fail_unless(r == 0);
+
+ if (l == 0)
+ return;
+
+#if 0
+ {
+ static int fd = -1;
+
+ if (fd == -1) {
+ fd = open("loopback.raw", O_CREAT | O_TRUNC | O_SYNC | O_RDWR, S_IRUSR | S_IWUSR);
+ fail_if(fd < 0);
+ }
+
+ r = write(fd, in, l);
+ }
+#endif
+
+ do {
+#if 0
+ {
+ int j;
+ fprintf(stderr, "%g (", rms(in, WINDOW));
+ for (j = 0; j < WINDOW; j++)
+ fprintf(stderr, "%g ", in[j]);
+ fprintf(stderr, ")\n");
+ }
+#endif
+ if (i + (ss * WINDOW) < l)
+ cur = rms(in, WINDOW);
+ else
+ cur = rms(in, (l - i)/ss);
+
+ /* We leave the definition of 0 generous since the window might
+ * straddle the 0->1 transition, raising the average power. We keep the
+ * definition of 1 tight in this case and detect the transition in the
+ * next round. */
+ if (last < 0.5 && cur > 0.8) {
+ pa_gettimeofday(&tv_in);
+ fprintf(stderr, "Latency %llu\n", (unsigned long long) pa_timeval_diff(&tv_in, &tv_out));
+ }
+
+ last = cur;
+ in += WINDOW;
+ i += ss * WINDOW;
+ } while (i + (ss * WINDOW) <= l);
+
+ pa_stream_drop(s);
+}
+
+/*
+ * We run a simple volume calibration so that we know we can detect the signal
+ * being played back. We start with the playback stream at 100% volume, and
+ * capture at 0.
+ *
+ * First, we then play a sine wave and increase the capture volume till the
+ * signal is clearly received.
+ *
+ * Next, we play back silence and make sure that the level is low enough to
+ * distinguish from when playback is happening.
+ *
+ * Finally, we hand off to the real read/write callbacks to run the actual
+ * test.
+ */
+
+enum {
+ CALIBRATION_ONE,
+ CALIBRATION_ZERO,
+ CALIBRATION_DONE,
+};
+
+static int cal_state = CALIBRATION_ONE;
+
+static void calibrate_write_cb(pa_stream *s, size_t nbytes, void *userdata) {
+ int i, r, nsamp = nbytes / fs;
+ float tmp[nsamp][2];
+ static int count = 0;
+
+ /* Write out a sine tone */
+ for (i = 0; i < nsamp; i++)
+ tmp[i][0] = tmp[i][1] = cal_state == CALIBRATION_ONE ? sin(count++ * TONE_HZ * 2 * M_PI / SAMPLE_HZ) : 0.0;
+
+ r = pa_stream_write(s, &tmp, nbytes, nop_free_cb, 0, PA_SEEK_RELATIVE);
+ fail_unless(r == 0);
+
+ if (cal_state == CALIBRATION_DONE)
+ pa_stream_set_write_callback(s, write_cb, NULL);
+}
+
+static void calibrate_read_cb(pa_stream *s, size_t nbytes, void *userdata) {
+ static double v = 0;
+ static int skip = 0, confirm;
+
+ pa_cvolume vol;
+ pa_operation *o;
+ int r, nsamp;
+ float *in;
+ size_t l;
+
+ r = pa_stream_peek(s, (const void **)&in, &l);
+ fail_unless(r == 0);
+
+ nsamp = l / fs;
+
+ /* For each state or volume step change, throw out a few samples so we know
+ * we're seeing the changed samples. */
+ if (skip++ < 100)
+ goto out;
+ else
+ skip = 0;
+
+ switch (cal_state) {
+ case CALIBRATION_ONE:
+ /* Try to detect the sine wave */
+ if (rms(in, nsamp) < 0.8) {
+ confirm = 0;
+ v += 0.02;
+
+ if (v > 1.0) {
+ fprintf(stderr, "Capture signal too weak at 100%% volume (%g). Giving up.\n", rms(in, nsamp));
+ fail();
+ }
+
+ pa_cvolume_set(&vol, CHANNELS, v * PA_VOLUME_NORM);
+ o = pa_context_set_source_output_volume(context, pa_stream_get_index(s), &vol, NULL, NULL);
+ fail_if(o == NULL);
+ pa_operation_unref(o);
+ } else {
+ /* Make sure the signal strength is steadily above our threshold */
+ if (++confirm > 5) {
+#if 0
+ fprintf(stderr, "Capture volume = %g (%g)\n", v, rms(in, nsamp));
+#endif
+ cal_state = CALIBRATION_ZERO;
+ }
+ }
+
+ break;
+
+ case CALIBRATION_ZERO:
+ /* Now make sure silence doesn't trigger a false positive because
+ * of noise. */
+ if (rms(in, nsamp) > 0.1) {
+ fprintf(stderr, "Too much noise on capture (%g). Giving up.\n", rms(in, nsamp));
+ fail();
+ }
+
+ cal_state = CALIBRATION_DONE;
+ pa_stream_set_read_callback(s, read_cb, NULL);
+
+ break;
+
+ default:
+ break;
+ }
+
+out:
+ pa_stream_drop(s);
+}
+
+/* This routine is called whenever the stream state changes */
+static void stream_state_callback(pa_stream *s, void *userdata) {
+ switch (pa_stream_get_state(s)) {
+ case PA_STREAM_UNCONNECTED:
+ case PA_STREAM_CREATING:
+ case PA_STREAM_TERMINATED:
+ break;
+
+ case PA_STREAM_READY: {
+ pa_cvolume vol;
+ pa_operation *o;
+
+ /* Set volumes for calibration */
+ if (!userdata) {
+ pa_cvolume_set(&vol, CHANNELS, PA_VOLUME_NORM);
+ o = pa_context_set_sink_input_volume(context, pa_stream_get_index(s), &vol, NULL, NULL);
+ } else {
+ pa_cvolume_set(&vol, CHANNELS, pa_sw_volume_from_linear(0.0));
+ o = pa_context_set_source_output_volume(context, pa_stream_get_index(s), &vol, NULL, NULL);
+ }
+
+ if (!o) {
+ fprintf(stderr, "Could not set stream volume: %s\n", pa_strerror(pa_context_errno(context)));
+ fail();
+ } else
+ pa_operation_unref(o);
+
+ break;
+ }
+
+ case PA_STREAM_FAILED:
+ default:
+ fprintf(stderr, "Stream error: %s\n", pa_strerror(pa_context_errno(pa_stream_get_context(s))));
+ fail();
+ }
+}
+
+/* This is called whenever the context status changes */
+static void context_state_callback(pa_context *c, void *userdata) {
+ fail_unless(c != NULL);
+
+ switch (pa_context_get_state(c)) {
+ case PA_CONTEXT_CONNECTING:
+ case PA_CONTEXT_AUTHORIZING:
+ case PA_CONTEXT_SETTING_NAME:
+ break;
+
+ case PA_CONTEXT_READY: {
+ pa_buffer_attr buffer_attr;
+
+ /* Create playback stream */
+ buffer_attr.maxlength = -1;
+ buffer_attr.tlength = SAMPLE_HZ * fs * PLAYBACK_LATENCY / 1000;
+ buffer_attr.prebuf = 0; /* Setting prebuf to 0 guarantees us the stream will run synchronously, no matter what */
+ buffer_attr.minreq = -1;
+ buffer_attr.fragsize = -1;
+
+ pstream = pa_stream_new(c, "loopback: play", &sample_spec, NULL);
+ fail_unless(pstream != NULL);
+ pa_stream_set_state_callback(pstream, stream_state_callback, (void *) 0);
+ pa_stream_set_write_callback(pstream, calibrate_write_cb, NULL);
+ pa_stream_set_underflow_callback(pstream, underflow_cb, userdata);
+
+ pa_stream_connect_playback(pstream, getenv("TEST_SINK"), &buffer_attr,
+ PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE, NULL, NULL);
+
+ /* Create capture stream */
+ buffer_attr.maxlength = -1;
+ buffer_attr.tlength = (uint32_t) -1;
+ buffer_attr.prebuf = 0;
+ buffer_attr.minreq = (uint32_t) -1;
+ buffer_attr.fragsize = SAMPLE_HZ * fs * CAPTURE_LATENCY / 1000;
+
+ rstream = pa_stream_new(c, "loopback: rec", &sample_spec, NULL);
+ fail_unless(rstream != NULL);
+ pa_stream_set_state_callback(rstream, stream_state_callback, (void *) 1);
+ pa_stream_set_read_callback(rstream, calibrate_read_cb, NULL);
+ pa_stream_set_overflow_callback(rstream, overflow_cb, userdata);
+
+ pa_stream_connect_record(rstream, getenv("TEST_SOURCE"), &buffer_attr,
+ PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
+
+ break;
+ }
+
+ case PA_CONTEXT_TERMINATED:
+ mainloop_api->quit(mainloop_api, 0);
+ break;
+
+ case PA_CONTEXT_FAILED:
+ default:
+ fprintf(stderr, "Context error: %s\n", pa_strerror(pa_context_errno(c)));
+ fail();
+ }
+}
+
+START_TEST (loopback_test) {
+ pa_mainloop* m = NULL;
+ int i, ret = 0, pulse_hz = N_OUT / 1000;
+
+ /* Generate a square pulse */
+ for (i = 0; i < N_OUT; i++)
+ if (i < pulse_hz)
+ out[i][0] = out[i][1] = 1.0;
+ else
+ out[i][0] = out[i][1] = 0.0;
+
+ ss = pa_sample_size(&sample_spec);
+ fs = pa_frame_size(&sample_spec);
+
+ pstream = NULL;
+
+ /* Set up a new main loop */
+ m = pa_mainloop_new();
+ fail_unless(m != NULL);
+
+ mainloop_api = pa_mainloop_get_api(m);
+
+ context = pa_context_new(mainloop_api, bname);
+ fail_unless(context != NULL);
+
+ pa_context_set_state_callback(context, context_state_callback, NULL);
+
+ /* Connect the context */
+ if (pa_context_connect(context, NULL, 0, NULL) < 0) {
+ fprintf(stderr, "pa_context_connect() failed.\n");
+ goto quit;
+ }
+
+ if (pa_mainloop_run(m, &ret) < 0)
+ fprintf(stderr, "pa_mainloop_run() failed.\n");
+
+quit:
+ pa_context_unref(context);
+
+ if (pstream)
+ pa_stream_unref(pstream);
+
+ pa_mainloop_free(m);
+
+ fail_unless(ret == 0);
+}
+END_TEST
+
+int main(int argc, char *argv[]) {
+ int failed = 0;
+ Suite *s;
+ TCase *tc;
+ SRunner *sr;
+
+ bname = argv[0];
+
+ s = suite_create("Loopback");
+ tc = tcase_create("loopback");
+ tcase_add_test(tc, loopback_test);
+ tcase_set_timeout(tc, 5 * 60);
+ suite_add_tcase(s, tc);
+
+ sr = srunner_create(s);
+ srunner_set_fork_status(sr, CK_NOFORK);
+ srunner_run_all(sr, CK_NORMAL);
+ failed = srunner_ntests_failed(sr);
+ srunner_free(sr);
+
+ return (failed == 0) ? EXIT_SUCCESS : EXIT_FAILURE;
+}
--
1.8.2.1
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