X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.h;h=9793646c9bac3493651e39643a610c7df75c5d0f;hb=6ffaabac0a523617b686f40c154a25cb548cc561;hp=821171037f6faf70b11e7d8518484380ecbe587f;hpb=7c1bb8357495778076a47636c2c4192674034165;p=nageru diff --git a/audio_mixer.h b/audio_mixer.h index 8211710..9793646 100644 --- a/audio_mixer.h +++ b/audio_mixer.h @@ -8,19 +8,20 @@ // // All operations on AudioMixer (except destruction) are thread-safe. -#include +#include #include +#include #include +#include +#include #include #include #include #include +#include #include -#include -#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 @@ -43,7 +46,7 @@ enum EQBand { 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(); @@ -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 get_output(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy); + std::vector 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; } @@ -139,6 +142,18 @@ public: return locut_enabled[bus]; } + bool is_mono(unsigned bus_index); + + void set_stereo_width(unsigned bus_index, float width) + { + stereo_width[bus_index] = width; + } + + float get_stereo_width(unsigned bus_index) + { + return stereo_width[bus_index]; + } + void set_eq(unsigned bus_index, EQBand band, float db_gain) { assert(band >= 0 && band < NUM_EQ_BANDS); @@ -286,6 +301,7 @@ public: float fader_volume_db; bool muted; bool locut_enabled; + float stereo_width; float eq_level_db[NUM_EQ_BANDS]; float gain_staging_db; bool level_compressor_enabled; @@ -299,7 +315,6 @@ public: private: struct AudioDevice { std::unique_ptr 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). @@ -315,7 +330,7 @@ private: AudioDevice *find_audio_device(DeviceSpec device_spec); void find_sample_src_from_device(const std::map> &samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride); - void fill_audio_bus(const std::map> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float *output); + void fill_audio_bus(const std::map> &samples_card, const InputMapping::Bus &bus, unsigned num_samples, float stereo_width, float *output); void reset_resampler_mutex_held(DeviceSpec device_spec); void apply_eq(unsigned bus_index, std::vector *samples_bus); void update_meters(const std::vector &samples); @@ -325,13 +340,14 @@ private: std::vector 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 ffmpeg_inputs; // Under audio_mutex. std::atomic locut_cutoff_hz{120}; StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct. @@ -373,6 +389,7 @@ private: std::atomic fader_volume_db[MAX_BUSES] {{ 0.0f }}; std::atomic mute[MAX_BUSES] {{ false }}; float last_fader_volume_db[MAX_BUSES] { 0.0f }; // Under audio_mutex. + std::atomic stereo_width[MAX_BUSES] {{ 0.0f }}; // Default 1.0f (is set in constructor). std::atomic eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{{ 0.0f }}}; float last_eq_level_db[MAX_BUSES][NUM_EQ_BANDS] {{ 0.0f }}; @@ -383,6 +400,28 @@ private: CorrelationMeasurer correlation; // Under audio_measure_mutex. Resampler peak_resampler; // Under audio_measure_mutex. std::atomic peak{0.0f}; + + // Metrics. + std::atomic metric_audio_loudness_short_lufs{0.0 / 0.0}; + std::atomic metric_audio_loudness_integrated_lufs{0.0 / 0.0}; + std::atomic metric_audio_loudness_range_low_lufs{0.0 / 0.0}; + std::atomic metric_audio_loudness_range_high_lufs{0.0 / 0.0}; + std::atomic metric_audio_peak_dbfs{0.0 / 0.0}; + std::atomic metric_audio_final_makeup_gain_db{0.0}; + std::atomic 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> labels; + std::atomic current_level_dbfs[2]{{0.0/0.0},{0.0/0.0}}; + std::atomic peak_level_dbfs[2]{{0.0/0.0},{0.0/0.0}}; + std::atomic historic_peak_dbfs{0.0/0.0}; + std::atomic gain_staging_db{0.0/0.0}; + std::atomic compressor_attenuation_db{0.0/0.0}; + }; + std::unique_ptr bus_metrics; // One for each bus in . }; extern AudioMixer *global_audio_mixer;