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-rw-r--r--src/Makefile.am9
-rw-r--r--src/tests/lo-latency-test.c330
-rw-r--r--src/tests/lo-test-util.c328
-rw-r--r--src/tests/lo-test-util.h57
4 files changed, 424 insertions, 300 deletions
diff --git a/src/Makefile.am b/src/Makefile.am
index 163976e5d..44c191d75 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -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 = \
diff --git a/src/tests/lo-latency-test.c b/src/tests/lo-latency-test.c
index 8f3b04d4d..124693dc3 100644
--- a/src/tests/lo-latency-test.c
+++ b/src/tests/lo-latency-test.c
@@ -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);
-
- 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;
- }
+ i += ctx->ss * WINDOW;
+ } while (i + (ctx->ss * WINDOW) <= l);
-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();
- }
-}
+START_TEST (loopback_test) {
+ int i, pulse_hz = SAMPLE_HZ / 1000;
-/* 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;
- }
+ test_ctx.context_name = context_name;
- case PA_CONTEXT_TERMINATED:
- mainloop_api->quit(mainloop_api, 0);
- break;
+ test_ctx.sample_spec.format = PA_SAMPLE_FLOAT32,
+ test_ctx.sample_spec.rate = SAMPLE_HZ,
+ test_ctx.sample_spec.channels = CHANNELS,
- case PA_CONTEXT_FAILED:
- default:
- fprintf(stderr, "Context error: %s\n", pa_strerror(pa_context_errno(c)));
- fail();
- }
-}
+ test_ctx.play_latency = 25;
+ test_ctx.rec_latency = 5;
-START_TEST (loopback_test) {
- pa_mainloop* m = NULL;
- int i, ret = 0, pulse_hz = SAMPLE_HZ / 1000;
+ 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);
diff --git a/src/tests/lo-test-util.c b/src/tests/lo-test-util.c
new file mode 100644
index 000000000..01eb295d7
--- /dev/null
+++ b/src/tests/lo-test-util.c
@@ -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);
+}
diff --git a/src/tests/lo-test-util.h b/src/tests/lo-test-util.h
new file mode 100644
index 000000000..d0de60948
--- /dev/null
+++ b/src/tests/lo-test-util.h
@@ -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);
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-28 06:42:26 +0000