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authorSimon Ser <simon.ser@intel.com>2019-06-10 15:46:40 +0300
committerSimon Ser <simon.ser@intel.com>2019-06-11 12:46:50 +0300
commit1f2acef836ffcbf5113df4ade8b587935c7d5868 (patch)
treed69c25d1d8cada5a4d3c24b981e320830a060ede /tests/kms_chamelium.c
parent6cb3d4a9457cdfb993ebb2a086a4844b85c49ee2 (diff)
tests/kms_chamelium: add error messages to assertions
This makes it clearer what happened when an assertion fails. It's also easier to write CI bug log filters. Signed-off-by: Simon Ser <simon.ser@intel.com> Reviewed-By: Martin Peres <martin.peres@linux.intel.com>
Diffstat (limited to 'tests/kms_chamelium.c')
-rw-r--r--tests/kms_chamelium.c29
1 files changed, 17 insertions, 12 deletions
diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c
index 3beea0c6..685f7709 100644
--- a/tests/kms_chamelium.c
+++ b/tests/kms_chamelium.c
@@ -863,7 +863,8 @@ static void audio_state_init(struct audio_state *state, data_t *data,
alsa_configure_output(alsa, format, channels, rate);
state->stream = chamelium_stream_init();
- igt_assert(state->stream);
+ igt_assert_f(state->stream,
+ "Failed to initialize Chamelium stream client\n");
}
static void audio_state_fini(struct audio_state *state)
@@ -902,13 +903,13 @@ static void audio_state_start(struct audio_state *state, const char *name)
stream_mode = CHAMELIUM_STREAM_REALTIME_STOP_WHEN_OVERFLOW;
ok = chamelium_stream_dump_realtime_audio(state->stream, stream_mode);
- igt_assert(ok);
+ igt_assert_f(ok, "Failed to start streaming audio capture\n");
/* Start playing audio */
state->run = true;
ret = pthread_create(&state->thread, NULL,
run_audio_thread, state->alsa);
- igt_assert(ret == 0);
+ igt_assert_f(ret == 0, "Failed to start audio playback thread\n");
/* The Chamelium device only supports this PCM format. */
state->capture.format = SND_PCM_FORMAT_S32_LE;
@@ -936,7 +937,7 @@ static void audio_state_start(struct audio_state *state, const char *name)
break;
}
}
- igt_assert(ok);
+ igt_assert_f(ok, "Cannot capture all channels\n");
}
if (igt_frame_dump_is_enabled()) {
@@ -949,7 +950,8 @@ static void audio_state_start(struct audio_state *state, const char *name)
state->capture.rate,
state->capture.channels,
&state->dump_path);
- igt_assert(state->dump_fd >= 0);
+ igt_assert_f(state->dump_fd >= 0,
+ "Failed to create audio dump file\n");
}
}
@@ -963,7 +965,7 @@ static void audio_state_receive(struct audio_state *state,
ok = chamelium_stream_receive_realtime_audio(state->stream,
&page_count,
recv, recv_len);
- igt_assert(ok);
+ igt_assert_f(ok, "Failed to receive audio from stream server\n");
state->msec = state->recv_pages * *recv_len
/ (double) state->capture.channels
@@ -972,7 +974,8 @@ static void audio_state_receive(struct audio_state *state,
if (state->dump_fd >= 0) {
recv_size = *recv_len * sizeof(int32_t);
- igt_assert(write(state->dump_fd, *recv, recv_size) == recv_size);
+ igt_assert_f(write(state->dump_fd, *recv, recv_size) == recv_size,
+ "Failed to write to audio dump file\n");
}
}
@@ -985,10 +988,10 @@ static void audio_state_stop(struct audio_state *state, bool success)
igt_debug("Stopping audio playback\n");
state->run = false;
ret = pthread_join(state->thread, NULL);
- igt_assert(ret == 0);
+ igt_assert_f(ret == 0, "Failed to join audio playback thread\n");
ok = chamelium_stream_stop_realtime_audio(state->stream);
- igt_assert(ok);
+ igt_assert_f(ok, "Failed to stop streaming audio capture\n");
audio_file = chamelium_stop_capturing_audio(state->chamelium,
state->port);
@@ -1047,7 +1050,7 @@ static bool test_audio_frequencies(struct audio_state *state)
state->signal = audio_signal_init(state->playback.channels,
state->playback.rate);
- igt_assert(state->signal);
+ igt_assert_f(state->signal, "Failed to initialize audio signal\n");
/* We'll choose different frequencies per channel to make sure they are
* independent from each other. To do so, we'll add a different offset
@@ -1075,7 +1078,9 @@ static bool test_audio_frequencies(struct audio_state *state)
audio_state_start(state, "frequencies");
- igt_assert(state->capture.rate == state->playback.rate);
+ igt_assert_f(state->capture.rate == state->playback.rate,
+ "Capture rate (%dHz) doesn't match playback rate (%dHz)\n",
+ state->capture.rate, state->playback.rate);
/* Needs to be a multiple of 128, because that's the number of samples
* we get per channel each time we receive an audio page from the
@@ -1391,7 +1396,7 @@ test_display_audio(data_t *data, struct chamelium_port *port,
for (i = 0; i < test_sampling_rates_count; i++) {
for (j = 0; j < test_formats_count; j++) {
ret = alsa_open_output(alsa, audio_device);
- igt_assert(ret >= 0);
+ igt_assert_f(ret >= 0, "Failed to open ALSA output\n");
/* TODO: playback on all 8 available channels (this
* isn't supported by Chamelium devices yet, see