lib/igt_{alsa, audio}: improve logging

- Print matched audio devices
- Print min/max values when alsa_test_configuration fails
- Print debug log line when skipping a frequency

Signed-off-by: Simon Ser <simon.ser@intel.com>
Reviewed-by: Martin Peres <martin.peres@linux.intel.com>

2 files changed, 23 insertions, 3 deletions

diff --git a/lib/igt_alsa.c b/lib/igt_alsa.c
index fc6d336b..3b1a9e90 100644
--- a/lib/igt_alsa.c
+++ b/lib/igt_alsa.c
@@ -182,6 +182,8 @@ static char *alsa_resolve_indentifier(const char *device_name, int skip)
 					continue;
 				}
 
+				igt_debug("Matched device \"%s\"\n", pcm_name);
+
 				snprintf(name, sizeof(name), "hw:%d,%d", card,
 					 dev);
 
@@ -329,6 +331,9 @@ static bool alsa_test_configuration(snd_pcm_t *handle, int channels,
 {
 	snd_pcm_hw_params_t *params;
 	int ret;
+	unsigned int min_channels, max_channels;
+	unsigned int min_rate, max_rate;
+	int min_rate_dir, max_rate_dir;
 
 	snd_pcm_hw_params_alloca(&params);
 
@@ -337,12 +342,24 @@ static bool alsa_test_configuration(snd_pcm_t *handle, int channels,
 		return false;
 
 	ret = snd_pcm_hw_params_test_rate(handle, params, sampling_rate, 0);
-	if (ret < 0)
+	if (ret < 0) {
+		snd_pcm_hw_params_get_rate_min(params, &min_rate, &min_rate_dir);
+		snd_pcm_hw_params_get_rate_max(params, &max_rate, &max_rate_dir);
+		igt_debug("Output device supports rates between %u and %u, "
+			  "requested %d\n",
+			  min_rate, max_rate, sampling_rate);
 		return false;
+	}
 
 	ret = snd_pcm_hw_params_test_channels(handle, params, channels);
-	if (ret < 0)
+	if (ret < 0) {
+		snd_pcm_hw_params_get_channels_min(params, &min_channels);
+		snd_pcm_hw_params_get_channels_max(params, &max_channels);
+		igt_debug("Output device supports between %u and "
+			  "%u channels, requested %d\n",
+			  min_channels, max_channels, channels);
 		return false;
+	}
 
 	return true;
 }
diff --git a/lib/igt_audio.c b/lib/igt_audio.c
index 7624f565..5822fed7 100644
--- a/lib/igt_audio.c
+++ b/lib/igt_audio.c
@@ -102,8 +102,11 @@ int audio_signal_add_frequency(struct audio_signal *signal, int frequency)
 		return -1;
 
 	/* Stay within the Nyquist–Shannon sampling theorem. */
-	if (frequency > signal->sampling_rate / 2)
+	if (frequency > signal->sampling_rate / 2) {
+		igt_debug("Skipping frequency %d: too high for a %d Hz "
+			  "sampling rate\n", frequency, signal->sampling_rate);
 		return -1;
+	}
 
 	/* Clip the frequency to an integer multiple of the sampling rate.
 	 * This to be able to store a full period of it and use that for