X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.h;h=17f95280e28eba8a9a85756cb24c626a6f3f858c;hb=96cb6414f85e0ef4d660b7bd56267303e80fcd05;hp=821171037f6faf70b11e7d8518484380ecbe587f;hpb=7c1bb8357495778076a47636c2c4192674034165;p=nageru

diff --git a/audio_mixer.h b/audio_mixer.h
index 8211710..17f9528 100644
--- a/audio_mixer.h
+++ b/audio_mixer.h
@@ -8,19 +8,20 @@
 //
 // All operations on AudioMixer (except destruction) are thread-safe.
 
-#include <math.h>
+#include <assert.h>
 #include <stdint.h>
+#include <zita-resampler/resampler.h>
 #include <atomic>
+#include <chrono>
+#include <functional>
 #include <map>
 #include <memory>
 #include <mutex>
 #include <set>
+#include <string>
 #include <vector>
-#include <zita-resampler/resampler.h>
 
-#include "alsa_input.h"
 #include "alsa_pool.h"
-#include "bmusb/bmusb.h"
 #include "correlation_measurer.h"
 #include "db.h"
 #include "defs.h"
@@ -30,6 +31,8 @@
 #include "resampling_queue.h"
 #include "stereocompressor.h"
 
+class DeviceSpecProto;
+
 namespace bmusb {
 struct AudioFormat;
 }  // namespace bmusb
@@ -52,7 +55,7 @@ public:
 	// (This is to avoid a deadlock where a card hangs on the mutex in add_audio()
 	// while we are trying to shut it down from another thread that also holds
 	// the mutex.) frame_length is in TIMEBASE units.
-	bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
+	bool add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length, std::chrono::steady_clock::time_point frame_time);
 	bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
 
 	// If a given device is offline for whatever reason and cannot deliver audio
@@ -62,7 +65,7 @@ public:
 	// affect it. Same true/false behavior as add_audio().
 	bool silence_card(DeviceSpec device_spec, bool silence);
 
-	std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
+	std::vector<float> get_output(std::chrono::steady_clock::time_point ts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
 
 	float get_fader_volume(unsigned bus_index) const { return fader_volume_db[bus_index]; }
 	void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
@@ -299,7 +302,6 @@ public:
 private:
 	struct AudioDevice {
 		std::unique_ptr<ResamplingQueue> resampling_queue;
-		int64_t next_local_pts = 0;
 		std::string display_name;
 		unsigned capture_frequency = OUTPUT_FREQUENCY;
 		// Which channels we consider interesting (ie., are part of some input_mapping).
@@ -383,6 +385,28 @@ private:
 	CorrelationMeasurer correlation;  // Under audio_measure_mutex.
 	Resampler peak_resampler;  // Under audio_measure_mutex.
 	std::atomic<float> peak{0.0f};
+
+	// Metrics.
+	std::atomic<double> metric_audio_loudness_short_lufs{0.0 / 0.0};
+	std::atomic<double> metric_audio_loudness_integrated_lufs{0.0 / 0.0};
+	std::atomic<double> metric_audio_loudness_range_low_lufs{0.0 / 0.0};
+	std::atomic<double> metric_audio_loudness_range_high_lufs{0.0 / 0.0};
+	std::atomic<double> metric_audio_peak_dbfs{0.0 / 0.0};
+	std::atomic<double> metric_audio_final_makeup_gain_db{0.0};
+	std::atomic<double> metric_audio_correlation{0.0};
+
+	// These are all gauges corresponding to the elements of BusLevel.
+	// In a sense, they'd probably do better as histograms, but that's an
+	// awful lot of time series when you have many buses.
+	struct BusMetrics {
+		std::vector<std::pair<std::string, std::string>> labels;
+		std::atomic<double> current_level_dbfs[2]{{0.0/0.0},{0.0/0.0}};
+		std::atomic<double> peak_level_dbfs[2]{{0.0/0.0},{0.0/0.0}};
+		std::atomic<double> historic_peak_dbfs{0.0/0.0};
+		std::atomic<double> gain_staging_db{0.0/0.0};
+		std::atomic<double> compressor_attenuation_db{0.0/0.0};
+	};
+	std::unique_ptr<BusMetrics[]> bus_metrics;  // One for each bus in <input_mapping>.
 };
 
 extern AudioMixer *global_audio_mixer;