//
// 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 "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 class InputSourceType { SILENCE, CAPTURE_CARD, ALSA_INPUT };
-struct DeviceSpec {
- InputSourceType type;
- unsigned index;
-
- bool operator== (const DeviceSpec &other) const {
- return type == other.type && index == other.index;
- }
-
- bool operator< (const DeviceSpec &other) const {
- if (type != other.type)
- return type < other.type;
- return index < other.index;
- }
-};
-struct DeviceInfo {
- std::string name;
- unsigned num_channels;
-};
-
enum EQBand {
EQ_BAND_BASS = 0,
EQ_BAND_MID,
NUM_EQ_BANDS
};
-static inline uint64_t DeviceSpec_to_key(const DeviceSpec &device_spec)
-{
- return (uint64_t(device_spec.type) << 32) | device_spec.index;
-}
-
-static inline DeviceSpec key_to_DeviceSpec(uint64_t key)
-{
- return DeviceSpec{ InputSourceType(key >> 32), unsigned(key & 0xffffffff) };
-}
-
-struct InputMapping {
- struct Bus {
- std::string name;
- DeviceSpec device;
- int source_channel[2] { -1, -1 }; // Left and right. -1 = none.
- };
-
- std::vector<Bus> buses;
-};
-
class AudioMixer {
public:
- AudioMixer(unsigned num_cards);
- ~AudioMixer();
+ 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);
- std::vector<float> get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
+ // 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; }
- std::map<DeviceSpec, DeviceInfo> get_devices() const;
- void set_name(DeviceSpec device_spec, const std::string &name);
+ 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);
+
+ // 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;
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;
- int64_t next_local_pts = 0;
- std::string name;
+ 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;
- // Only used for ALSA cards, obviously.
- std::unique_ptr<ALSAInput> alsa_device;
+ 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 reset_alsa_mutex_held(DeviceSpec device_spec);
- std::map<DeviceSpec, DeviceInfo> get_devices_mutex_held() const;
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_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.
-
- // TODO: Figure out a better way to unify these two, as they are sharing indexing.
AudioDevice alsa_inputs[MAX_ALSA_CARDS]; // Under audio_mutex.
- std::vector<ALSAInput::Device> available_alsa_cards;
+ 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 }};
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