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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-27 16:03:18 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-27 16:03:18 +0000 |
commit | 2dd5bc6a074165ddfbd57c4bd52c2d2dac8f47a1 (patch) | |
tree | 465b29cb405d3af0b0ad50c78e1dccc636594fec /src/tests/lo-test-util.c | |
parent | Initial commit. (diff) | |
download | pulseaudio-2dd5bc6a074165ddfbd57c4bd52c2d2dac8f47a1.tar.xz pulseaudio-2dd5bc6a074165ddfbd57c4bd52c2d2dac8f47a1.zip |
Adding upstream version 14.2.upstream/14.2upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'src/tests/lo-test-util.c')
-rw-r--r-- | src/tests/lo-test-util.c | 324 |
1 files changed, 324 insertions, 0 deletions
diff --git a/src/tests/lo-test-util.c b/src/tests/lo-test-util.c new file mode 100644 index 0000000..40002a2 --- /dev/null +++ b/src/tests/lo-test-util.c @@ -0,0 +1,324 @@ +/*** + 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, see <http://www.gnu.org/licenses/>. +***/ + +#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"); +} + +static void overflow_cb(struct pa_stream *s, void *userdata) { + pa_log_warn("Overlow"); +} + +/* + * 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, 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; + + pa_assert_se(pa_stream_write(s, &tmp, nbytes, nop_free_cb, 0, PA_SEEK_RELATIVE) == 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 nsamp; + float *in; + size_t l; + + pa_assert_se(pa_stream_peek(s, (const void **)&in, &l) == 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.", 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)", 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) { + pa_log_warn("Too much noise on capture (%g). Giving up.", 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", 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", 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_thread_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", 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."); + 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."); + 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); +} |