diff options
| -rw-r--r-- | lib/igt_audio.c | 18 | ||||
| -rw-r--r-- | lib/igt_audio.h | 2 |
2 files changed, 16 insertions, 4 deletions
diff --git a/lib/igt_audio.c b/lib/igt_audio.c index f4b16959..bc2fc103 100644 --- a/lib/igt_audio.c +++ b/lib/igt_audio.c @@ -251,11 +251,14 @@ void audio_signal_fill(struct audio_signal *signal, int16_t *buffer, * Checks that frequencies specified in signal, and only those, are included * in the input data. * - * sampling_rate is given in Hz. data_len is the number of elements in data. + * sampling_rate is given in Hz. samples_len is the number of elements in + * samples. */ bool audio_signal_detect(struct audio_signal *signal, int sampling_rate, - int channel, double *data, size_t data_len) + int channel, const double *samples, size_t samples_len) { + double *data; + size_t data_len = samples_len; size_t bin_power_len = data_len / 2 + 1; double bin_power[bin_power_len]; bool detected[FREQS_MAX]; @@ -264,12 +267,19 @@ bool audio_signal_detect(struct audio_signal *signal, int sampling_rate, size_t i, j; bool above, success; + /* gsl will mutate the array in-place, so make a copy */ + data = malloc(samples_len * sizeof(double)); + memcpy(data, samples, samples_len * sizeof(double)); + /* Allowed error in Hz due to FFT step */ freq_accuracy = sampling_rate / data_len; igt_debug("Allowed freq. error: %d Hz\n", freq_accuracy); ret = gsl_fft_real_radix2_transform(data, 1, data_len); - igt_assert(ret == 0); + if (ret != 0) { + free(data); + igt_assert(0); + } /* Compute the power received by every bin of the FFT, and record the * maximum power received as a way to normalize all the others. @@ -372,6 +382,8 @@ bool audio_signal_detect(struct audio_signal *signal, int sampling_rate, } } + free(data); + return success; } diff --git a/lib/igt_audio.h b/lib/igt_audio.h index 466e772a..d5ba1caa 100644 --- a/lib/igt_audio.h +++ b/lib/igt_audio.h @@ -43,7 +43,7 @@ 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, - int channel, double *data, size_t data_len); + int channel, const double *samples, size_t samples_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); |