//
// 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"
#include "resampling_queue.h"
#include "stereocompressor.h"
+class DeviceSpecProto;
+
namespace bmusb {
struct AudioFormat;
} // namespace bmusb
class AudioMixer {
public:
- AudioMixer(unsigned num_cards);
+ AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs);
void reset_resampler(DeviceSpec device_spec);
void reset_meters();
// (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
// 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; }
+ bool get_mute(unsigned bus_index) const { return mute[bus_index]; }
+ void set_mute(unsigned bus_index, bool muted) { mute[bus_index] = muted; }
+
// Note: This operation holds all ALSA devices (see ALSAPool::get_devices()).
// You will need to call set_input_mapping() to get the hold state correctly,
// or every card will be held forever.
// Note: The card should be held (currently this isn't enforced, though).
void serialize_device(DeviceSpec device_spec, DeviceSpecProto *device_spec_proto);
+ enum class MappingMode {
+ // A single bus, only from a video card (no ALSA devices),
+ // only channel 1 and 2, locked to +0 dB. Note that this is
+ // only an UI abstraction around exactly the same audio code
+ // as MULTICHANNEL; it's just less flexible.
+ SIMPLE,
+
+ // Full, arbitrary mappings.
+ MULTICHANNEL
+ };
+
+ // Automatically sets mapping mode to MappingMode::SIMPLE.
+ void set_simple_input(unsigned card_index);
+
+ // If mapping mode is not representable as a MappingMode::SIMPLE type
+ // mapping, returns numeric_limits<unsigned>::max().
+ unsigned get_simple_input() const;
+
+ // Implicitly sets mapping mode to MappingMode::MULTICHANNEL.
void set_input_mapping(const InputMapping &input_mapping);
+
+ MappingMode get_mapping_mode() const;
InputMapping get_input_mapping() const;
+ unsigned num_buses() const;
+
void set_locut_cutoff(float cutoff_hz)
{
locut_cutoff_hz = cutoff_hz;
// or set_* functions for that bus.
struct BusSettings {
float fader_volume_db;
+ bool muted;
bool locut_enabled;
float eq_level_db[NUM_EQ_BANDS];
float gain_staging_db;
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).
void measure_bus_levels(unsigned bus_index, const std::vector<float> &left, const std::vector<float> &right);
void send_audio_level_callback();
std::vector<DeviceSpec> get_active_devices() const;
+ void set_input_mapping_lock_held(const InputMapping &input_mapping);
- unsigned num_cards;
+ unsigned num_capture_cards, num_ffmpeg_inputs;
mutable std::timed_mutex audio_mutex;
ALSAPool alsa_pool;
AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
+ std::unique_ptr<AudioDevice[]> ffmpeg_inputs; // Under audio_mutex.
std::atomic<float> locut_cutoff_hz{120};
StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct.
mutable std::mutex compressor_mutex;
std::unique_ptr<StereoCompressor> level_compressor[MAX_BUSES]; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
float gain_staging_db[MAX_BUSES]; // Under compressor_mutex.
+ float last_gain_staging_db[MAX_BUSES]; // Under compressor_mutex.
bool level_compressor_enabled[MAX_BUSES]; // Under compressor_mutex.
static constexpr float ref_level_dbfs = -14.0f; // Chosen so that we end up around 0 LU in practice.
double final_makeup_gain = 1.0; // Under compressor_mutex. Read/write by the user. Note: Not in dB, we want the numeric precision so that we can change it slowly.
bool final_makeup_gain_auto = true; // Under compressor_mutex.
+ MappingMode current_mapping_mode; // Under audio_mutex.
InputMapping input_mapping; // Under audio_mutex.
std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
+ std::atomic<bool> mute[MAX_BUSES] {{ false }};
float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex.
std::atomic<float> eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}};
float last_eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{ 0.0f }};
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;
projects / nageru / blobdiff