#include <stdio.h>
#include <endian.h>
#include <cmath>
+#ifdef __SSE__
+#include <immintrin.h>
+#endif
#include "db.h"
#include "flags.h"
}
}
-float find_peak(const float *samples, size_t num_samples)
+float find_peak_plain(const float *samples, size_t num_samples) __attribute__((unused));
+
+float find_peak_plain(const float *samples, size_t num_samples)
{
float m = fabs(samples[0]);
for (size_t i = 1; i < num_samples; ++i) {
return m;
}
+#ifdef __SSE__
+static inline float horizontal_max(__m128 m)
+{
+ __m128 tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 0, 3, 2));
+ m = _mm_max_ps(m, tmp);
+ tmp = _mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 3, 0, 1));
+ m = _mm_max_ps(m, tmp);
+ return _mm_cvtss_f32(m);
+}
+
+float find_peak(const float *samples, size_t num_samples)
+{
+ const __m128 abs_mask = _mm_castsi128_ps(_mm_set1_epi32(0x7fffffffu));
+ __m128 m = _mm_setzero_ps();
+ for (size_t i = 0; i < (num_samples & ~3); i += 4) {
+ __m128 x = _mm_loadu_ps(samples + i);
+ x = _mm_and_ps(x, abs_mask);
+ m = _mm_max_ps(m, x);
+ }
+ float result = horizontal_max(m);
+
+ for (size_t i = (num_samples & ~3); i < num_samples; ++i) {
+ result = max(result, fabs(samples[i]));
+ }
+
+#if 0
+ // Self-test. We should be bit-exact the same.
+ float reference_result = find_peak_plain(samples, num_samples);
+ if (result != reference_result) {
+ fprintf(stderr, "Error: Peak is %f [%f %f %f %f]; should be %f.\n",
+ result,
+ _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(0, 0, 0, 0))),
+ _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(1, 1, 1, 1))),
+ _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(2, 2, 2, 2))),
+ _mm_cvtss_f32(_mm_shuffle_ps(m, m, _MM_SHUFFLE(3, 3, 3, 3))),
+ reference_result);
+ abort();
+ }
+#endif
+ return result;
+}
+#else
+float find_peak(const float *samples, size_t num_samples)
+{
+ return find_peak_plain(samples, num_samples);
+}
+#endif
+
void deinterleave_samples(const vector<float> &in, vector<float> *out_l, vector<float> *out_r)
{
size_t num_samples = in.size() / 2;
assert(num_channels > 0);
// Convert the audio to fp32.
- vector<float> audio;
- audio.resize(num_samples * num_channels);
+ unique_ptr<float[]> audio(new float[num_samples * num_channels]);
unsigned channel_index = 0;
for (auto channel_it = device->interesting_channels.cbegin(); channel_it != device->interesting_channels.end(); ++channel_it, ++channel_index) {
switch (audio_format.bits_per_sample) {
assert(num_samples == 0);
break;
case 16:
- convert_fixed16_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ convert_fixed16_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
break;
case 24:
- convert_fixed24_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ convert_fixed24_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
break;
case 32:
- convert_fixed32_to_fp32(&audio[0], channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
+ convert_fixed32_to_fp32(audio.get(), channel_index, num_channels, data, *channel_it, audio_format.num_channels, num_samples);
break;
default:
fprintf(stderr, "Cannot handle audio with %u bits per sample\n", audio_format.bits_per_sample);
// Now add it.
int64_t local_pts = device->next_local_pts;
- device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.data(), num_samples);
+ device->resampling_queue->add_input_samples(local_pts / double(TIMEBASE), audio.get(), num_samples);
device->next_local_pts = local_pts + frame_length;
return true;
}