[igt-dev] [PATCH i-g-t 1/2] lib/igt_audio: fix synthesized signal amplitude

[Martin Peres]

On 17/05/2019 13:11, Simon Ser wrote:
> For each channel of the audio signal, we choose a set of frequencies to
> generate. However we previously divided each sample by the total number of
> frequencies, instead of the number of frequencies assigned to the channel we're
> currently processing. This resulted in low amplitudes (0.5 instead of 1.0 for
> 2 channels).
> 
> This patch fixes this issue and sets the generated signal amplitude to 0.9.
> Indeed, we want to detect if the receiver grows the signal, we don't want it
> to get capped at 1.0 immediately.
> 
> Signed-off-by: Simon Ser <simon.ser at intel.com>

Reviewed-by: Martin Peres <martin.peres at linux.intel.com>

> ---
>  lib/igt_audio.c | 33 ++++++++++++++++++++++++++++-----
>  1 file changed, 28 insertions(+), 5 deletions(-)
> 
> diff --git a/lib/igt_audio.c b/lib/igt_audio.c
> index 90d16fe4bd11..f6c8e399e345 100644
> --- a/lib/igt_audio.c
> +++ b/lib/igt_audio.c
> @@ -36,6 +36,8 @@
>  #include "igt_core.h"
>  
>  #define FREQS_MAX 64
> +#define CHANNELS_MAX 8
> +#define SYNTHESIZE_AMPLITUDE 0.9
>  
>  /**
>   * SECTION:igt_audio
> @@ -77,6 +79,8 @@ struct audio_signal *audio_signal_init(int channels, int sampling_rate)
>  {
>  	struct audio_signal *signal;
>  
> +	igt_assert(channels <= CHANNELS_MAX);
> +
>  	signal = malloc(sizeof(struct audio_signal));
>  	memset(signal, 0, sizeof(struct audio_signal));
>  
> @@ -156,7 +160,7 @@ void audio_signal_synthesize(struct audio_signal *signal)
>  
>  		for (j = 0; j < period_len; j++) {
>  			value = 2.0 * M_PI * freq / signal->sampling_rate * j;
> -			value = sin(value) / signal->freqs_count;
> +			value = sin(value) * SYNTHESIZE_AMPLITUDE;
>  
>  			period[j] = value;
>  		}
> @@ -195,6 +199,20 @@ void audio_signal_reset(struct audio_signal *signal)
>  	signal->freqs_count = 0;
>  }
>  
> +static size_t audio_signal_count_freqs(struct audio_signal *signal, int channel)
> +{
> +	size_t n, i;
> +	struct audio_signal_freq *freq;
> +
> +	for (i = 0; i < signal->freqs_count; i++) {
> +		freq = &signal->freqs[i];
> +		if (freq->channel < 0 || freq->channel == channel)
> +			n++;
> +	}
> +
> +	return n;
> +}
> +
>  /**
>   * audio_signal_fill:
>   * @signal: The target signal structure
> @@ -208,14 +226,18 @@ void audio_signal_reset(struct audio_signal *signal)
>  void audio_signal_fill(struct audio_signal *signal, double *buffer,
>  		       size_t samples)
>  {
> -	double *destination, *source;
> +	double *dst, *src;
>  	struct audio_signal_freq *freq;
>  	int total;
>  	int count;
>  	int i, j, k;
> +	size_t freqs_per_channel[CHANNELS_MAX];
>  
>  	memset(buffer, 0, sizeof(double) * signal->channels * samples);
>  
> +	for (i = 0; i < signal->channels; i++)
> +		freqs_per_channel[i] = audio_signal_count_freqs(signal, i);
> +
>  	for (i = 0; i < signal->freqs_count; i++) {
>  		freq = &signal->freqs[i];
>  		total = 0;
> @@ -223,8 +245,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
>  		igt_assert(freq->period);
>  
>  		while (total < samples) {
> -			source = freq->period + freq->offset;
> -			destination = buffer + total * signal->channels;
> +			src = freq->period + freq->offset;
> +			dst = buffer + total * signal->channels;
>  
>  			count = freq->period_len - freq->offset;
>  			if (count > samples - total)
> @@ -238,7 +260,8 @@ void audio_signal_fill(struct audio_signal *signal, double *buffer,
>  					if (freq->channel >= 0 &&
>  					    freq->channel != k)
>  						continue;
> -					destination[j * signal->channels + k] += source[j];
> +					dst[j * signal->channels + k] +=
> +						src[j] / freqs_per_channel[k];
>  				}
>  			}
>  
> 
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