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tests: Factor out loopback setup code

Browse source 
This moves over setup code for the loopback latency test into a private
library so that we can easily write more tests using the same framework.
This commit is contained in:
Arun Raghavan 2013-05-23 15:27:40 +05:30
parent 36bdd720aa
commit e001cc1424
4 changed files with 427 additions and 303 deletions
Download patch file Download diff file Expand all files Collapse all files

9
src/Makefile.am
View file

@ -223,6 +223,7 @@ pax11publish_LDFLAGS = $(AM_LDFLAGS) $(BINLDFLAGS)
###################################
# Test programs #
###################################
noinst_LTLIBRARIES =
TESTS_default = \
mainloop-test \
@ -575,8 +576,13 @@ echo_cancel_test_CXXFLAGS = $(module_echo_cancel_la_CXXFLAGS) -DECHO_CANCEL_TEST
endif
echo_cancel_test_LDFLAGS = $(AM_LDFLAGS) $(BINLDFLAGS)
liblo_test_util_la_SOURCES = tests/lo-test-util.h tests/lo-test-util.c
liblo_test_util_la_LIBADD = libpulsecore-@PA_MAJORMINOR@.la
liblo_test_util_la_LDFLAGS = -avoid-version
noinst_LTLIBRARIES += liblo-test-util.la
lo_latency_test_SOURCES = tests/lo-latency-test.c
lo_latency_test_LDADD = $(AM_LDADD) libpulse.la
lo_latency_test_LDADD = $(AM_LDADD) libpulse.la liblo-test-util.la
lo_latency_test_CFLAGS = $(AM_CFLAGS) $(LIBCHECK_CFLAGS)
lo_latency_test_LDFLAGS = $(AM_LDFLAGS) $(BINLDFLAGS) $(LIBCHECK_LIBS)
@ -855,7 +861,6 @@ libpulsedsp_la_LDFLAGS = $(AM_LDFLAGS) -avoid-version -disable-static
###################################
lib_LTLIBRARIES += libpulsecore-@PA_MAJORMINOR@.la
noinst_LTLIBRARIES =
# Pure core stuff
libpulsecore_@PA_MAJORMINOR@_la_SOURCES = \

336
src/tests/lo-latency-test.c
View file

@ -32,62 +32,33 @@
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <check.h>
#include <pulse/pulseaudio.h>
#include <pulse/mainloop.h>
/* for pa_make_realtime */
#include <pulsecore/core-util.h>
#include "lo-test-util.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 *context_name = NULL;
static float out[N_OUT][CHANNELS];
static int ppos = 0;
static int n_underflow = 0;
static int n_overflow = 0;
pa_lo_test_context test_ctx;
static const char *context_name = NULL;
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 nop_free_cb(void *p) {
}
static void write_cb(pa_stream *s, size_t nbytes, void *userdata) {
int r, nsamp = nbytes / fs;
pa_lo_test_context *ctx = (pa_lo_test_context *) userdata;
static int ppos = 0;
int r, nsamp = nbytes / ctx->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;
r = pa_stream_write(s, &out[ppos][0], (N_OUT - ppos) * ctx->fs, nop_free_cb, 0, PA_SEEK_RELATIVE);
nbytes -= (N_OUT - ppos) * ctx->fs;
ppos = 0;
}
@ -97,22 +68,13 @@ static void write_cb(pa_stream *s, size_t nbytes, void *userdata) {
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 sqrtf(sq / n);
ppos = (ppos + nbytes / ctx->fs) % N_OUT;
}
#define WINDOW (2 * CHANNELS)
static void read_cb(pa_stream *s, size_t nbytes, void *userdata) {
pa_lo_test_context *ctx = (pa_lo_test_context *) userdata;
static float last = 0.0f;
const float *in;
float cur;
@ -143,16 +105,16 @@ static void read_cb(pa_stream *s, size_t nbytes, void *userdata) {
#if 0
{
int j;
fprintf(stderr, "%g (", rms(in, WINDOW));
fprintf(stderr, "%g (", pa_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);
if (i + (ctx->ss * WINDOW) < l)
cur = pa_rms(in, WINDOW);
else
cur = rms(in, (l - i)/ss);
cur = pa_rms(in, (l - i) / ctx->ss);
/* We leave the definition of 0 generous since the window might
* straddle the 0->1 transition, raising the average power. We keep the
@ -165,223 +127,26 @@ static void read_cb(pa_stream *s, size_t nbytes, void *userdata) {
last = cur;
in += WINDOW;
i += ss * WINDOW;
} while (i + (ss * WINDOW) <= l);
i += ctx->ss * WINDOW;
} while (i + (ctx->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 ? sinf(count++ * TONE_HZ * 2 * M_PI / SAMPLE_HZ) : 0.0f;
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. RMS is 0.5, */
if (rms(in, nsamp) < 0.40f) {
confirm = 0;
v += 0.02f;
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) > 0.1f) {
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;
pa_make_realtime(4);
/* 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 = SAMPLE_HZ / 1000;
int i, pulse_hz = SAMPLE_HZ / 1000;
test_ctx.context_name = context_name;
test_ctx.sample_spec.format = PA_SAMPLE_FLOAT32,
test_ctx.sample_spec.rate = SAMPLE_HZ,
test_ctx.sample_spec.channels = CHANNELS,
test_ctx.play_latency = 25;
test_ctx.rec_latency = 5;
test_ctx.read_cb = read_cb;
test_ctx.write_cb = write_cb;
/* Generate a square pulse */
for (i = 0; i < N_OUT; i++)
@ -390,40 +155,9 @@ START_TEST (loopback_test) {
else
out[i][0] = out[i][1] = 0.0f;
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, context_name);
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);
fail_unless(pa_lo_test_init(&test_ctx) == 0);
fail_unless(pa_lo_test_run(&test_ctx) == 0);
pa_lo_test_deinit(&test_ctx);
}
END_TEST
@ -435,8 +169,8 @@ int main(int argc, char *argv[]) {
context_name = argv[0];
s = suite_create("Loopback");
tc = tcase_create("loopback");
s = suite_create("Loopback latency");
tc = tcase_create("loopback latency");
tcase_add_test(tc, loopback_test);
tcase_set_timeout(tc, 5 * 60);
suite_add_tcase(s, tc);

328
src/tests/lo-test-util.c Normal file
View file

@ -0,0 +1,328 @@
/***
This file is part of PulseAudio.
Copyright 2013 Collabora Ltd.
Author: Arun Raghavan <arun.raghavan@collabora.co.uk>
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 <math.h>
#include <pulsecore/log.h>
#include <pulsecore/macro.h>
#include <pulsecore/core-util.h>
#include "lo-test-util.h"
/* Keep the frequency high so RMS over ranges of a few ms remains relatively
* high as well */
#define TONE_HZ 4410
static void nop_free_cb(void *p) {
}
static void underflow_cb(struct pa_stream *s, void *userdata) {
pa_log_warn("Underflow\n");
}
static void overflow_cb(struct pa_stream *s, void *userdata) {
pa_log_warn("Overlow\n");
}
/*
* 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) {
pa_lo_test_context *ctx = (pa_lo_test_context *) userdata;
int i, r, nsamp = nbytes / ctx->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 ? sinf(count++ * TONE_HZ * 2 * M_PI / ctx->sample_spec.rate) : 0.0f;
r = pa_stream_write(s, &tmp, nbytes, nop_free_cb, 0, PA_SEEK_RELATIVE);
pa_assert(r == 0);
if (cal_state == CALIBRATION_DONE)
pa_stream_set_write_callback(s, ctx->write_cb, ctx);
}
static void calibrate_read_cb(pa_stream *s, size_t nbytes, void *userdata) {
pa_lo_test_context *ctx = (pa_lo_test_context *) 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);
pa_assert(r == 0);
nsamp = l / ctx->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. RMS is 0.5, */
if (pa_rms(in, nsamp) < 0.40f) {
confirm = 0;
v += 0.02f;
if (v > 1.0) {
pa_log_error("Capture signal too weak at 100%% volume (%g). Giving up.\n", pa_rms(in, nsamp));
pa_assert_not_reached();
}
pa_cvolume_set(&vol, ctx->sample_spec.channels, v * PA_VOLUME_NORM);
o = pa_context_set_source_output_volume(ctx->context, pa_stream_get_index(s), &vol, NULL, NULL);
pa_assert(o != NULL);
pa_operation_unref(o);
} else {
/* Make sure the signal strength is steadily above our threshold */
if (++confirm > 5) {
#if 0
pa_log_debug(stderr, "Capture volume = %g (%g)\n", v, pa_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 (pa_rms(in, nsamp) > 0.1f) {
fprintf(stderr, "Too much noise on capture (%g). Giving up.\n", pa_rms(in, nsamp));
pa_assert_not_reached();
}
cal_state = CALIBRATION_DONE;
pa_stream_set_read_callback(s, ctx->read_cb, ctx);
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) {
pa_lo_test_context *ctx = (pa_lo_test_context *) 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 (s == ctx->play_stream) {
pa_cvolume_set(&vol, ctx->sample_spec.channels, PA_VOLUME_NORM);
o = pa_context_set_sink_input_volume(ctx->context, pa_stream_get_index(s), &vol, NULL, NULL);
} else {
pa_cvolume_set(&vol, ctx->sample_spec.channels, pa_sw_volume_from_linear(0.0));
o = pa_context_set_source_output_volume(ctx->context, pa_stream_get_index(s), &vol, NULL, NULL);
}
if (!o) {
pa_log_error("Could not set stream volume: %s\n", pa_strerror(pa_context_errno(ctx->context)));
pa_assert_not_reached();
} else
pa_operation_unref(o);
break;
}
case PA_STREAM_FAILED:
default:
pa_log_error("Stream error: %s\n", pa_strerror(pa_context_errno(ctx->context)));
pa_assert_not_reached();
}
}
/* This is called whenever the context status changes */
static void context_state_callback(pa_context *c, void *userdata) {
pa_lo_test_context *ctx = (pa_lo_test_context *) userdata;
pa_mainloop_api *api;
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;
pa_make_realtime(4);
/* Create playback stream */
buffer_attr.maxlength = -1;
buffer_attr.tlength = ctx->sample_spec.rate * ctx->fs * ctx->play_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;
ctx->play_stream = pa_stream_new(c, "loopback: play", &ctx->sample_spec, NULL);
pa_assert(ctx->play_stream != NULL);
pa_stream_set_state_callback(ctx->play_stream, stream_state_callback, ctx);
pa_stream_set_write_callback(ctx->play_stream, calibrate_write_cb, ctx);
pa_stream_set_underflow_callback(ctx->play_stream, underflow_cb, userdata);
pa_stream_connect_playback(ctx->play_stream, 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 = ctx->sample_spec.rate * ctx->fs * ctx->rec_latency / 1000;
ctx->rec_stream = pa_stream_new(c, "loopback: rec", &ctx->sample_spec, NULL);
pa_assert(ctx->rec_stream != NULL);
pa_stream_set_state_callback(ctx->rec_stream, stream_state_callback, ctx);
pa_stream_set_read_callback(ctx->rec_stream, calibrate_read_cb, ctx);
pa_stream_set_overflow_callback(ctx->rec_stream, overflow_cb, userdata);
pa_stream_connect_record(ctx->rec_stream, getenv("TEST_SOURCE"), &buffer_attr,
PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
break;
}
case PA_CONTEXT_TERMINATED:
api = pa_mainloop_get_api(ctx->mainloop);
api->quit(api, 0);
break;
case PA_CONTEXT_FAILED:
default:
pa_log_error("Context error: %s\n", pa_strerror(pa_context_errno(c)));
pa_assert_not_reached();
}
}
int pa_lo_test_init(pa_lo_test_context *ctx) {
/* FIXME: need to deal with non-float samples at some point */
pa_assert(ctx->sample_spec.format == PA_SAMPLE_FLOAT32);
ctx->ss = pa_sample_size(&ctx->sample_spec);
ctx->fs = pa_frame_size(&ctx->sample_spec);
ctx->mainloop = pa_mainloop_new();
ctx->context = pa_context_new(pa_mainloop_get_api(ctx->mainloop), ctx->context_name);
pa_context_set_state_callback(ctx->context, context_state_callback, ctx);
/* Connect the context */
if (pa_context_connect(ctx->context, NULL, PA_CONTEXT_NOFLAGS, NULL) < 0) {
pa_log_error("pa_context_connect() failed.\n");
goto quit;
}
return 0;
quit:
pa_context_unref(ctx->context);
pa_mainloop_free(ctx->mainloop);
return -1;
}
int pa_lo_test_run(pa_lo_test_context *ctx) {
int ret;
if (pa_mainloop_run(ctx->mainloop, &ret) < 0) {
pa_log_error("pa_mainloop_run() failed.\n");
return -1;
}
return 0;
}
void pa_lo_test_deinit(pa_lo_test_context *ctx) {
if (ctx->play_stream) {
pa_stream_disconnect(ctx->play_stream);
pa_stream_unref(ctx->play_stream);
}
if (ctx->rec_stream) {
pa_stream_disconnect(ctx->rec_stream);
pa_stream_unref(ctx->rec_stream);
}
if (ctx->context)
pa_context_unref(ctx->context);
if (ctx->mainloop)
pa_mainloop_free(ctx->mainloop);
}
float pa_rms(const float *s, int n) {
float sq = 0;
int i;
for (i = 0; i < n; i++)
sq += s[i] * s[i];
return sqrtf(sq / n);
}

57
src/tests/lo-test-util.h Normal file
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@ -0,0 +1,57 @@
/***
This file is part of PulseAudio.
Copyright 2013 Collabora Ltd.
Author: Arun Raghavan <arun.raghavan@collabora.co.uk>
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 <pulse/pulseaudio.h>
typedef struct pa_lo_test_context {
/* Tests need to set these */
const char *context_name;
pa_sample_spec sample_spec;
int play_latency; /* ms */
int rec_latency; /* ms */
pa_stream_request_cb_t write_cb, read_cb;
/* These are set by lo_test_init() */
pa_mainloop *mainloop;
pa_context *context;
pa_stream *play_stream, *rec_stream;
int ss, fs; /* sample size, frame size for convenience */
} pa_lo_test_context;
/* Initialise the test parameters, connect */
int pa_lo_test_init(pa_lo_test_context *ctx);
/* Start running the test */
int pa_lo_test_run(pa_lo_test_context *ctx);
/* Clean up */
void pa_lo_test_deinit(pa_lo_test_context *ctx);
/* Return RMS for the given signal. Assumes the data is a single channel for
* simplicity */
float pa_rms(const float *s, int n);