// all together into one final audio signal.
//
// All operations on AudioMixer (except destruction) are thread-safe.
-//
-// TODO: There might be more audio stuff that should be moved here
-// from Mixer.
-#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 "bmusb/bmusb.h"
+#include "alsa_pool.h"
+#include "correlation_measurer.h"
#include "db.h"
#include "defs.h"
+#include "ebu_r128_proc.h"
#include "filter.h"
+#include "input_mapping.h"
#include "resampling_queue.h"
#include "stereocompressor.h"
+class DeviceSpecProto;
+
namespace bmusb {
struct AudioFormat;
} // namespace bmusb
+enum EQBand {
+ EQ_BAND_BASS = 0,
+ EQ_BAND_MID,
+ EQ_BAND_TREBLE,
+ NUM_EQ_BANDS
+};
+
class AudioMixer {
public:
- AudioMixer(unsigned num_cards);
- void reset_card(unsigned card_index);
+ AudioMixer(unsigned num_capture_cards, unsigned num_ffmpeg_inputs);
+ void reset_resampler(DeviceSpec device_spec);
+ void reset_meters();
+
+ // Add audio (or silence) to the given device's queue. Can return false if
+ // the lock wasn't successfully taken; if so, you should simply try again.
+ // (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, 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
+ // (by means of add_audio() or add_silence()), you can call put it in silence mode,
+ // where it will be taken to only output silence. Note that when taking it _out_
+ // of silence mode, the resampler will be reset, so that old audio will not
+ // affect it. Same true/false behavior as add_audio().
+ bool silence_card(DeviceSpec device_spec, bool silence);
+
+ 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.
+ std::map<DeviceSpec, DeviceInfo> get_devices();
+
+ // See comments on ALSAPool::get_card_state().
+ ALSAPool::Device::State get_alsa_card_state(unsigned index)
+ {
+ return alsa_pool.get_card_state(index);
+ }
+
+ // See comments on ALSAPool::create_dead_card().
+ DeviceSpec create_dead_card(const std::string &name, const std::string &info, unsigned num_channels)
+ {
+ unsigned dead_card_index = alsa_pool.create_dead_card(name, info, num_channels);
+ return DeviceSpec{InputSourceType::ALSA_INPUT, dead_card_index};
+ }
+
+ void set_display_name(DeviceSpec device_spec, const std::string &name);
+
+ // 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);
- // frame_length is in TIMEBASE units.
- void add_audio(unsigned card_index, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
- void add_silence(unsigned card_index, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
- std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
+ // If mapping mode is not representable as a MappingMode::SIMPLE type
+ // mapping, returns numeric_limits<unsigned>::max().
+ unsigned get_simple_input() const;
- // See comments inside get_output().
- void set_current_loudness(double level_lufs) { loudness_lufs = level_lufs; }
+ // 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;
}
- void set_locut_enabled(bool enabled)
+ float get_locut_cutoff() const
+ {
+ return locut_cutoff_hz;
+ }
+
+ void set_locut_enabled(unsigned bus, bool enabled)
+ {
+ locut_enabled[bus] = enabled;
+ }
+
+ bool get_locut_enabled(unsigned bus)
+ {
+ return locut_enabled[bus];
+ }
+
+ void set_eq(unsigned bus_index, EQBand band, float db_gain)
{
- locut_enabled = enabled;
+ assert(band >= 0 && band < NUM_EQ_BANDS);
+ eq_level_db[bus_index][band] = db_gain;
}
- bool get_locut_enabled() const
+ float get_eq(unsigned bus_index, EQBand band) const
{
- return locut_enabled;
+ assert(band >= 0 && band < NUM_EQ_BANDS);
+ return eq_level_db[bus_index][band];
}
float get_limiter_threshold_dbfs() const
return limiter_threshold_dbfs;
}
- float get_compressor_threshold_dbfs() const
+ float get_compressor_threshold_dbfs(unsigned bus_index) const
{
- return compressor_threshold_dbfs;
+ return compressor_threshold_dbfs[bus_index];
}
void set_limiter_threshold_dbfs(float threshold_dbfs)
limiter_threshold_dbfs = threshold_dbfs;
}
- void set_compressor_threshold_dbfs(float threshold_dbfs)
+ void set_compressor_threshold_dbfs(unsigned bus_index, float threshold_dbfs)
{
- compressor_threshold_dbfs = threshold_dbfs;
+ compressor_threshold_dbfs[bus_index] = threshold_dbfs;
}
void set_limiter_enabled(bool enabled)
return limiter_enabled;
}
- void set_compressor_enabled(bool enabled)
+ void set_compressor_enabled(unsigned bus_index, bool enabled)
{
- compressor_enabled = enabled;
+ compressor_enabled[bus_index] = enabled;
}
- bool get_compressor_enabled() const
+ bool get_compressor_enabled(unsigned bus_index) const
{
- return compressor_enabled;
+ return compressor_enabled[bus_index];
}
- void set_gain_staging_db(float gain_db)
+ void set_gain_staging_db(unsigned bus_index, float gain_db)
{
std::unique_lock<std::mutex> lock(compressor_mutex);
- level_compressor_enabled = false;
- gain_staging_db = gain_db;
+ level_compressor_enabled[bus_index] = false;
+ gain_staging_db[bus_index] = gain_db;
}
- float get_gain_staging_db() const
+ float get_gain_staging_db(unsigned bus_index) const
{
std::unique_lock<std::mutex> lock(compressor_mutex);
- return gain_staging_db;
+ return gain_staging_db[bus_index];
}
- void set_gain_staging_auto(bool enabled)
+ void set_gain_staging_auto(unsigned bus_index, bool enabled)
{
std::unique_lock<std::mutex> lock(compressor_mutex);
- level_compressor_enabled = enabled;
+ level_compressor_enabled[bus_index] = enabled;
}
- bool get_gain_staging_auto() const
+ bool get_gain_staging_auto(unsigned bus_index) const
{
std::unique_lock<std::mutex> lock(compressor_mutex);
- return level_compressor_enabled;
+ return level_compressor_enabled[bus_index];
}
void set_final_makeup_gain_db(float gain_db)
return final_makeup_gain_auto;
}
-private:
- unsigned num_cards;
+ void reset_peak(unsigned bus_index);
- struct CaptureCard {
- std::mutex audio_mutex;
- std::unique_ptr<ResamplingQueue> resampling_queue; // Under audio_mutex.
- int64_t next_local_pts = 0; // Beginning of next frame, in TIMEBASE units. Under audio_mutex.
+ struct BusLevel {
+ float current_level_dbfs[2]; // Digital peak of last frame, left and right.
+ float peak_level_dbfs[2]; // Digital peak with hold, left and right.
+ float historic_peak_dbfs;
+ float gain_staging_db;
+ float compressor_attenuation_db; // A positive number; 0.0 for no attenuation.
};
- CaptureCard cards[MAX_CARDS];
- StereoFilter locut; // Default cutoff 120 Hz, 24 dB/oct.
- std::atomic<float> locut_cutoff_hz;
- std::atomic<bool> locut_enabled{true};
+ typedef std::function<void(float level_lufs, float peak_db,
+ std::vector<BusLevel> bus_levels,
+ float global_level_lufs, float range_low_lufs, float range_high_lufs,
+ float final_makeup_gain_db,
+ float correlation)> audio_level_callback_t;
+ void set_audio_level_callback(audio_level_callback_t callback)
+ {
+ audio_level_callback = callback;
+ }
+
+ typedef std::function<void()> state_changed_callback_t;
+ void set_state_changed_callback(state_changed_callback_t callback)
+ {
+ state_changed_callback = callback;
+ }
+
+ state_changed_callback_t get_state_changed_callback() const
+ {
+ return state_changed_callback;
+ }
+
+ void trigger_state_changed_callback()
+ {
+ if (state_changed_callback != nullptr) {
+ state_changed_callback();
+ }
+ }
+
+ // A combination of all settings for a bus. Useful if you want to get
+ // or store them as a whole without bothering to call all of the get_*
+ // 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;
+ bool level_compressor_enabled;
+ float compressor_threshold_dbfs;
+ bool compressor_enabled;
+ };
+ static BusSettings get_default_bus_settings();
+ BusSettings get_bus_settings(unsigned bus_index) const;
+ void set_bus_settings(unsigned bus_index, const BusSettings &settings);
+
+private:
+ struct AudioDevice {
+ std::unique_ptr<ResamplingQueue> resampling_queue;
+ std::string display_name;
+ unsigned capture_frequency = OUTPUT_FREQUENCY;
+ // Which channels we consider interesting (ie., are part of some input_mapping).
+ std::set<unsigned> interesting_channels;
+ bool silenced = false;
+ };
+
+ const AudioDevice *find_audio_device(DeviceSpec device_spec) const
+ {
+ return const_cast<AudioMixer *>(this)->find_audio_device(device_spec);
+ }
+
+ AudioDevice *find_audio_device(DeviceSpec device_spec);
+
+ void find_sample_src_from_device(const std::map<DeviceSpec, std::vector<float>> &samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride);
+ void fill_audio_bus(const std::map<DeviceSpec, std::vector<float>> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
+ void reset_resampler_mutex_held(DeviceSpec device_spec);
+ void apply_eq(unsigned bus_index, std::vector<float> *samples_bus);
+ void update_meters(const std::vector<float> &samples);
+ void add_bus_to_master(unsigned bus_index, const std::vector<float> &samples_bus, std::vector<float> *samples_out);
+ 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_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.
+ std::atomic<bool> locut_enabled[MAX_BUSES];
+ StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS]; // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
// First compressor; takes us up to about -12 dBFS.
mutable std::mutex compressor_mutex;
- StereoCompressor level_compressor; // Under compressor_mutex. Used to set/override gain_staging_db if <level_compressor_enabled>.
- float gain_staging_db = 0.0f; // Under compressor_mutex.
- bool level_compressor_enabled = true; // Under 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.
static constexpr float ref_level_lufs = -23.0f; // 0 LU, more or less by definition.
- std::atomic<float> loudness_lufs{ref_level_lufs};
-
StereoCompressor limiter;
std::atomic<float> limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB.
std::atomic<bool> limiter_enabled{true};
- StereoCompressor compressor;
- std::atomic<float> compressor_threshold_dbfs{ref_level_dbfs - 12.0f}; // -12 dB.
- std::atomic<bool> compressor_enabled{true};
+ std::unique_ptr<StereoCompressor> compressor[MAX_BUSES];
+ std::atomic<float> compressor_threshold_dbfs[MAX_BUSES];
+ std::atomic<bool> compressor_enabled[MAX_BUSES];
+
+ // Note: The values here are not in dB.
+ struct PeakHistory {
+ float current_level = 0.0f; // Peak of the last frame.
+ float historic_peak = 0.0f; // Highest peak since last reset; no falloff.
+ float current_peak = 0.0f; // Current peak of the peak meter.
+ float last_peak = 0.0f;
+ float age_seconds = 0.0f; // Time since "last_peak" was set.
+ };
+ PeakHistory peak_history[MAX_BUSES][2]; // Separate for each channel. Under audio_mutex.
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 }};
+
+ audio_level_callback_t audio_level_callback = nullptr;
+ state_changed_callback_t state_changed_callback = nullptr;
+ mutable std::mutex audio_measure_mutex;
+ Ebu_r128_proc r128; // Under audio_measure_mutex.
+ 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;
+
#endif // !defined(_AUDIO_MIXER_H)
projects / nageru / blobdiff