From 0f39100b2c70cd251c2827aab39925f4fa298453 Mon Sep 17 00:00:00 2001 From: Simon Ser Date: Tue, 23 Apr 2019 16:04:56 +0300 Subject: tests/kms_chamelium: test we receive a signal from both audio channels This commit updates the audio test to make sure we receive a signal from both audio channels. However this commit doesn't check that left and right channels are not swapped. Such a check requires some more work (because the Chamelium device does swap left and right channels) and will be implemented in a future commit. This commit adds a new channel argument to audio_signal_add_frequency, to add a frequency to a single channel only. Some light refactoring has been performed: a proper audio_signal_fini function has been introduced and size_t in now used when it should be. Signed-off-by: Simon Ser Reviewed-by: Martin Peres --- lib/igt_audio.c | 114 ++++++++++++++++++++++++++++++-------------------- lib/igt_audio.h | 11 +++-- tests/kms_chamelium.c | 58 +++++++++++++++---------- 3 files changed, 111 insertions(+), 72 deletions(-) diff --git a/lib/igt_audio.c b/lib/igt_audio.c index 5822fed7..5b0860e6 100644 --- a/lib/igt_audio.c +++ b/lib/igt_audio.c @@ -35,7 +35,7 @@ #include "igt_audio.h" #include "igt_core.h" -#define FREQS_MAX 8 +#define FREQS_MAX 64 /** * SECTION:igt_audio @@ -49,9 +49,10 @@ struct audio_signal_freq { int freq; + int channel; - short *period; - int frames; + int16_t *period; + size_t period_len; int offset; }; @@ -60,7 +61,7 @@ struct audio_signal { int sampling_rate; struct audio_signal_freq freqs[FREQS_MAX]; - int freqs_count; + size_t freqs_count; }; /** @@ -89,17 +90,21 @@ struct audio_signal *audio_signal_init(int channels, int sampling_rate) * audio_signal_add_frequency: * @signal: The target signal structure * @frequency: The frequency to add to the signal + * @channel: The channel to add this frequency to, or -1 to add it to all + * channels * * Add a frequency to the signal. * * Returns: An integer equal to zero for success and negative for failure */ -int audio_signal_add_frequency(struct audio_signal *signal, int frequency) +int audio_signal_add_frequency(struct audio_signal *signal, int frequency, + int channel) { - int index = signal->freqs_count; + size_t index = signal->freqs_count; + struct audio_signal_freq *freq; - if (index == FREQS_MAX) - return -1; + igt_assert(index < FREQS_MAX); + igt_assert(channel < signal->channels); /* Stay within the Nyquist–Shannon sampling theorem. */ if (frequency > signal->sampling_rate / 2) { @@ -114,11 +119,14 @@ int audio_signal_add_frequency(struct audio_signal *signal, int frequency) */ frequency = signal->sampling_rate / (signal->sampling_rate / frequency); - igt_debug("Adding test frequency %d\n", frequency); + igt_debug("Adding test frequency %d to channel %d\n", + frequency, channel); + + freq = &signal->freqs[index]; + memset(freq, 0, sizeof(*freq)); + freq->freq = frequency; + freq->channel = channel; - signal->freqs[index].freq = frequency; - signal->freqs[index].frames = 0; - signal->freqs[index].offset = 0; signal->freqs_count++; return 0; @@ -136,20 +144,17 @@ void audio_signal_synthesize(struct audio_signal *signal) { int16_t *period; double value; - int frames; + size_t period_len; int freq; int i, j; - if (signal->freqs_count == 0) - return; - for (i = 0; i < signal->freqs_count; i++) { freq = signal->freqs[i].freq; - frames = signal->sampling_rate / freq; + period_len = signal->sampling_rate / freq; - period = calloc(1, frames * sizeof(short)); + period = calloc(1, period_len * sizeof(int16_t)); - for (j = 0; j < frames; j++) { + for (j = 0; j < period_len; j++) { value = 2.0 * M_PI * freq / signal->sampling_rate * j; value = sin(value) * INT16_MAX / signal->freqs_count; @@ -157,26 +162,34 @@ void audio_signal_synthesize(struct audio_signal *signal) } signal->freqs[i].period = period; - signal->freqs[i].frames = frames; + signal->freqs[i].period_len = period_len; } } /** - * audio_signal_synthesize: + * audio_signal_fini: + * + * Release the signal. + */ +void audio_signal_fini(struct audio_signal *signal) +{ + audio_signal_reset(signal); + free(signal); +} + +/** + * audio_signal_reset: * @signal: The target signal structure * * Free the resources allocated by audio_signal_synthesize and remove * the previously-added frequencies. */ -void audio_signal_clean(struct audio_signal *signal) +void audio_signal_reset(struct audio_signal *signal) { - int i; + size_t i; for (i = 0; i < signal->freqs_count; i++) { - if (signal->freqs[i].period) - free(signal->freqs[i].period); - - memset(&signal->freqs[i], 0, sizeof(struct audio_signal_freq)); + free(signal->freqs[i].period); } signal->freqs_count = 0; @@ -186,44 +199,45 @@ void audio_signal_clean(struct audio_signal *signal) * audio_signal_fill: * @signal: The target signal structure * @buffer: The target buffer to fill - * @frames: The number of frames to fill + * @samples: The number of samples to fill * - * Fill the requested number of frames to the target buffer with the audio + * Fill the requested number of samples to the target buffer with the audio * signal data (in interleaved S16_LE format), at the requested sampling rate * and number of channels. */ -void audio_signal_fill(struct audio_signal *signal, int16_t *buffer, int frames) +void audio_signal_fill(struct audio_signal *signal, int16_t *buffer, + size_t buffer_len) { int16_t *destination, *source; + struct audio_signal_freq *freq; int total; - int freq_frames; - int freq_offset; int count; int i, j, k; - memset(buffer, 0, sizeof(int16_t) * signal->channels * frames); + memset(buffer, 0, sizeof(int16_t) * signal->channels * buffer_len); for (i = 0; i < signal->freqs_count; i++) { + freq = &signal->freqs[i]; total = 0; - while (total < frames) { - freq_frames = signal->freqs[i].frames; - freq_offset = signal->freqs[i].offset; + igt_assert(freq->period); - source = signal->freqs[i].period + freq_offset; + while (total < buffer_len) { + source = freq->period + freq->offset; destination = buffer + total * signal->channels; - count = freq_frames - freq_offset; - if (count > (frames - total)) - count = frames - total; + count = freq->period_len - freq->offset; + if (count > buffer_len - total) + count = buffer_len - total; - freq_offset += count; - freq_offset %= freq_frames; - - signal->freqs[i].offset = freq_offset; + freq->offset += count; + freq->offset %= freq->period_len; for (j = 0; j < count; j++) { for (k = 0; k < signal->channels; k++) { + if (freq->channel >= 0 && + freq->channel != k) + continue; destination[j * signal->channels + k] += source[j]; } } @@ -240,11 +254,11 @@ void audio_signal_fill(struct audio_signal *signal, int16_t *buffer, int frames) * sampling_rate is given in Hz. data_len is the number of elements in data. */ bool audio_signal_detect(struct audio_signal *signal, int sampling_rate, - double *data, size_t data_len) + int channel, double *data, size_t data_len) { size_t bin_power_len = data_len / 2 + 1; double bin_power[bin_power_len]; - bool detected[signal->freqs_count]; + bool detected[FREQS_MAX]; int ret, freq_accuracy, freq, local_max_freq; double max, local_max, threshold; size_t i, j; @@ -311,6 +325,10 @@ bool audio_signal_detect(struct audio_signal *signal, int sampling_rate, * invalid. */ if (bin_power[i] < threshold) { for (j = 0; j < signal->freqs_count; j++) { + if (signal->freqs[j].channel >= 0 && + signal->freqs[j].channel != channel) + continue; + if (signal->freqs[j].freq > local_max_freq - freq_accuracy && signal->freqs[j].freq < @@ -343,6 +361,10 @@ bool audio_signal_detect(struct audio_signal *signal, int sampling_rate, /* Check that all frequencies we generated have been detected. */ for (i = 0; i < signal->freqs_count; i++) { + if (signal->freqs[i].channel >= 0 && + signal->freqs[i].channel != channel) + continue; + if (!detected[i]) { igt_debug("Missing frequency: %d\n", signal->freqs[i].freq); diff --git a/lib/igt_audio.h b/lib/igt_audio.h index 4aa43e69..466e772a 100644 --- a/lib/igt_audio.h +++ b/lib/igt_audio.h @@ -35,12 +35,15 @@ struct audio_signal; struct audio_signal *audio_signal_init(int channels, int sampling_rate); -int audio_signal_add_frequency(struct audio_signal *signal, int frequency); +void audio_signal_fini(struct audio_signal *signal); +int audio_signal_add_frequency(struct audio_signal *signal, int frequency, + int channel); void audio_signal_synthesize(struct audio_signal *signal); -void audio_signal_clean(struct audio_signal *signal); -void audio_signal_fill(struct audio_signal *signal, int16_t *buffer, int frames); +void audio_signal_reset(struct audio_signal *signal); +void audio_signal_fill(struct audio_signal *signal, int16_t *buffer, + size_t buffer_len); bool audio_signal_detect(struct audio_signal *signal, int sampling_rate, - double *data, size_t data_len); + int channel, double *data, size_t data_len); size_t audio_extract_channel_s32_le(double *dst, size_t dst_cap, int32_t *src, size_t src_len, int n_channels, int channel); diff --git a/tests/kms_chamelium.c b/tests/kms_chamelium.c index 014a22b3..687f64c4 100644 --- a/tests/kms_chamelium.c +++ b/tests/kms_chamelium.c @@ -777,16 +777,16 @@ do_test_display_audio(data_t *data, struct chamelium_port *port, struct alsa *alsa, int playback_channels, int playback_rate) { - int ret, capture_rate, capture_channels, msec; + int ret, capture_rate, capture_channels, msec, freq; struct chamelium_audio_file *audio_file; struct chamelium_stream *stream; enum chamelium_stream_realtime_mode stream_mode; struct audio_signal *signal; int32_t *recv, *buf; double *channel; - size_t i, streak, page_count; + size_t i, j, streak, page_count; size_t recv_len, buf_len, buf_cap, buf_size, channel_len; - bool ok; + bool ok, success; char dump_suffix[64]; char *dump_path = NULL; int dump_fd = -1; @@ -794,10 +794,15 @@ do_test_display_audio(data_t *data, struct chamelium_port *port, struct audio_state state = {}; if (!alsa_test_output_configuration(alsa, playback_channels, - playback_rate)) + playback_rate)) { + igt_debug("Skipping test with sample rate %d and %d channels " + "because selected output devices don't support this " + "configuration\n", playback_rate, playback_channels); return false; + } - igt_debug("Testing with playback sampling rate %d\n", playback_rate); + igt_debug("Testing with playback sampling rate %d and %d channels\n", + playback_rate, playback_channels); alsa_configure_output(alsa, playback_channels, playback_rate); chamelium_start_capturing_audio(data->chamelium, port, false); @@ -825,8 +830,12 @@ do_test_display_audio(data_t *data, struct chamelium_port *port, signal = audio_signal_init(playback_channels, playback_rate); igt_assert(signal); - for (i = 0; i < test_frequencies_count; i++) - audio_signal_add_frequency(signal, test_frequencies[i]); + for (i = 0; i < test_frequencies_count; i++) { + for (j = 0; j < playback_channels; j++) { + freq = test_frequencies[i]; + audio_signal_add_frequency(signal, freq, j); + } + } audio_signal_synthesize(signal); state.signal = signal; @@ -851,10 +860,11 @@ do_test_display_audio(data_t *data, struct chamelium_port *port, recv = NULL; recv_len = 0; + success = false; streak = 0; msec = 0; i = 0; - while (streak < MIN_STREAK && msec < AUDIO_TIMEOUT) { + while (!success && msec < AUDIO_TIMEOUT) { ok = chamelium_stream_receive_realtime_audio(stream, &page_count, &recv, &recv_len); @@ -872,21 +882,27 @@ do_test_display_audio(data_t *data, struct chamelium_port *port, igt_assert(write(dump_fd, buf, buf_size) == buf_size); } - /* TODO: check other channels too, not just the first one */ - audio_extract_channel_s32_le(channel, channel_len, buf, buf_len, - capture_channels, 0); - msec = i * channel_len / (double) capture_rate * 1000; igt_debug("Detecting audio signal, t=%d msec\n", msec); - if (audio_signal_detect(signal, capture_rate, channel, - channel_len)) - streak++; - else - streak = 0; + for (j = 0; j < playback_channels; j++) { + igt_debug("Processing channel %zu\n", j); + + audio_extract_channel_s32_le(channel, channel_len, + buf, buf_len, + capture_channels, j); + + if (audio_signal_detect(signal, capture_rate, j, + channel, channel_len)) + streak++; + else + streak = 0; + } buf_len = 0; i++; + + success = streak == MIN_STREAK * playback_channels; } igt_debug("Stopping audio playback\n"); @@ -898,7 +914,7 @@ do_test_display_audio(data_t *data, struct chamelium_port *port, if (dump_fd >= 0) { close(dump_fd); - if (streak == MIN_STREAK) { + if (success) { /* Test succeeded, no need to keep the captured data */ unlink(dump_path); } else @@ -921,12 +937,10 @@ do_test_display_audio(data_t *data, struct chamelium_port *port, chamelium_destroy_audio_file(audio_file); } - audio_signal_clean(signal); - free(signal); - + audio_signal_fini(signal); chamelium_stream_deinit(stream); - igt_assert(streak == MIN_STREAK); + igt_assert(success); return true; } -- cgit v1.2.3