Add a benchmark for a simple audio chain; easier than profiling all of Nageru.

[nageru] / benchmark_audio_mixer.cpp

diff --git a/benchmark_audio_mixer.cpp b/benchmark_audio_mixer.cpp
new file mode 100644 (file)
index 0000000..e61c4f4
--- /dev/null
+++ b/benchmark_audio_mixer.cpp
@@ -0,0 +1,86 @@
+// Rather simplistic benchmark of AudioMixer. Sets up a simple mapping
+// with the default settings, feeds some white noise to the inputs and
+// runs a while. Useful for e.g. profiling.
+
+#include <stdio.h>
+#include <stdint.h>
+#include <vector>
+#include <chrono>
+#include "audio_mixer.h"
+#include "timebase.h"
+
+#define NUM_BENCHMARK_CARDS 4
+#define NUM_WARMUP_FRAMES 100
+#define NUM_BENCHMARK_FRAMES 1000
+#define NUM_CHANNELS 8
+#define NUM_SAMPLES 1024
+
+using namespace std;
+using namespace std::chrono;
+
+uint8_t samples[(NUM_SAMPLES * NUM_CHANNELS + 1024) * sizeof(uint16_t)];
+
+void callback(float level_lufs, float peak_db,
+              std::vector<AudioMixer::BusLevel> bus_levels,
+             float global_level_lufs, float range_low_lufs, float range_high_lufs,
+             float final_makeup_gain_db,
+             float correlation)
+{
+       // Empty.
+}
+
+int main(void)
+{
+       for (unsigned i = 0; i < NUM_SAMPLES * NUM_CHANNELS + 1024; ++i) {
+               samples[i] = rand() & 0xff;
+       }
+       AudioMixer mixer(NUM_BENCHMARK_CARDS);
+       mixer.set_audio_level_callback(callback);
+
+       InputMapping mapping;
+
+       InputMapping::Bus bus1;
+       bus1.device = DeviceSpec{InputSourceType::CAPTURE_CARD, 0};
+       bus1.source_channel[0] = 0;
+       bus1.source_channel[1] = 1;
+       mapping.buses.push_back(bus1);
+
+       InputMapping::Bus bus2;
+       bus2.device = DeviceSpec{InputSourceType::CAPTURE_CARD, 3};
+       bus2.source_channel[0] = 6;
+       bus2.source_channel[1] = 4;
+       mapping.buses.push_back(bus2);
+
+       mixer.set_input_mapping(mapping);
+
+       size_t out_samples = 0;
+
+       steady_clock::time_point start, end;
+       for (unsigned i = 0; i < NUM_WARMUP_FRAMES + NUM_BENCHMARK_FRAMES; ++i) {
+               if (i == NUM_WARMUP_FRAMES) {
+                       start = steady_clock::now();
+               }
+               // Feed the inputs.
+               for (unsigned card_index = 0; card_index < NUM_BENCHMARK_CARDS; ++card_index) {
+                       bmusb::AudioFormat audio_format;
+                       audio_format.bits_per_sample = 16;
+                       audio_format.num_channels = NUM_CHANNELS;
+                       
+                       unsigned num_samples = NUM_SAMPLES + (rand() % 9) - 5;
+                       bool ok = mixer.add_audio(DeviceSpec{InputSourceType::CAPTURE_CARD, card_index}, samples, num_samples, audio_format, NUM_SAMPLES * TIMEBASE / OUTPUT_FREQUENCY);
+                       assert(ok);
+               }
+
+               double pts = double(i) * NUM_SAMPLES / OUTPUT_FREQUENCY;
+               vector<float> output = mixer.get_output(pts, NUM_SAMPLES, ResamplingQueue::ADJUST_RATE);
+               if (i >= NUM_WARMUP_FRAMES) {
+                       out_samples += output.size();
+               }
+       }
+       end = steady_clock::now();
+
+       double elapsed = duration<double>(end - start).count();
+       double simulated = double(out_samples) / (OUTPUT_FREQUENCY * 2);
+       printf("%ld samples produced in %.1f ms (%.1f%% CPU, %.1fx realtime).\n",
+               out_samples, elapsed * 1e3, 100.0 * elapsed / simulated, simulated / elapsed);
+}