// 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 <stdint.h>
#include <mutex>
#include <set>
#include <vector>
+#include <zita-resampler/resampler.h>
+#include "alsa_input.h"
#include "bmusb/bmusb.h"
+#include "correlation_measurer.h"
#include "db.h"
#include "defs.h"
+#include "ebu_r128_proc.h"
#include "filter.h"
#include "resampling_queue.h"
#include "stereocompressor.h"
struct AudioFormat;
} // namespace bmusb
-enum class InputSourceType { SILENCE, CAPTURE_CARD };
+enum class InputSourceType { SILENCE, CAPTURE_CARD, ALSA_INPUT };
struct DeviceSpec {
InputSourceType type;
unsigned index;
class AudioMixer {
public:
AudioMixer(unsigned num_cards);
- void reset_device(DeviceSpec device_spec);
+ ~AudioMixer();
+ 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);
+ bool add_silence(DeviceSpec device_spec, unsigned samples_per_frame, unsigned num_frames, int64_t frame_length);
- // frame_length is in TIMEBASE units.
- void add_audio(DeviceSpec device_spec, const uint8_t *data, unsigned num_samples, bmusb::AudioFormat audio_format, int64_t frame_length);
- void 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);
- // See comments inside get_output().
- void set_current_loudness(double level_lufs) { loudness_lufs = level_lufs; }
-
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);
return final_makeup_gain_auto;
}
+ typedef std::function<void(float level_lufs, float peak_db,
+ float global_level_lufs, float range_low_lufs, float range_high_lufs,
+ float gain_staging_db, 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;
+ }
+
private:
struct AudioDevice {
std::unique_ptr<ResamplingQueue> resampling_queue;
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;
};
AudioDevice *find_audio_device(DeviceSpec device_spec);
- void find_sample_src_from_device(const std::vector<float> *samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride);
- void fill_audio_bus(const std::vector<float> *samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output);
- void reset_device_mutex_held(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 update_meters(const std::vector<float> &samples);
+ void send_audio_level_callback();
unsigned num_cards;
- mutable std::mutex audio_mutex;
+ mutable std::timed_mutex audio_mutex;
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;
+
StereoFilter locut; // Default cutoff 120 Hz, 24 dB/oct.
std::atomic<float> locut_cutoff_hz;
std::atomic<bool> locut_enabled{true};
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};
InputMapping input_mapping; // Under audio_mutex.
std::atomic<float> fader_volume_db[MAX_BUSES] {{ 0.0f }};
+
+ audio_level_callback_t audio_level_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};
};
#endif // !defined(_AUDIO_MIXER_H)
projects / nageru / blobdiff