X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.h;h=75b468a120f7d1f39aa293009e9b76153ae4e9a3;hb=c015407a3953235df07a601baa6aa8e02ba7b561;hp=c7841fbdf32125116803af67dfffe4d0f3c66a34;hpb=19a44f0f0dacb3cd96bacf83bb55ceda7590fda9;p=nageru diff --git a/audio_mixer.h b/audio_mixer.h index c7841fb..75b468a 100644 --- a/audio_mixer.h +++ b/audio_mixer.h @@ -7,9 +7,6 @@ // 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 #include @@ -19,10 +16,14 @@ #include #include #include +#include +#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" @@ -31,12 +32,36 @@ namespace bmusb { struct AudioFormat; } // namespace bmusb -enum class InputSourceType { SILENCE, CAPTURE_CARD }; +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; }; +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; @@ -50,18 +75,22 @@ struct InputMapping { 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 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::vector get_names() const; + std::map get_devices() const; void set_name(DeviceSpec device_spec, const std::string &name); void set_input_mapping(const InputMapping &input_mapping); @@ -172,25 +201,45 @@ public: return final_makeup_gain_auto; } + typedef std::function audio_level_callback_t; + void set_audio_level_callback(audio_level_callback_t callback) + { + audio_level_callback = callback; + } + private: struct AudioDevice { std::unique_ptr resampling_queue; int64_t next_local_pts = 0; std::string name; + unsigned capture_frequency = OUTPUT_FREQUENCY; // Which channels we consider interesting (ie., are part of some input_mapping). std::set interesting_channels; + // Only used for ALSA cards, obviously. + std::unique_ptr alsa_device; }; AudioDevice *find_audio_device(DeviceSpec device_spec); - void find_sample_src_from_device(const std::vector *samples_card, DeviceSpec device_spec, int source_channel, const float **srcptr, unsigned *stride); - void fill_audio_bus(const std::vector *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> &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 reset_resampler_mutex_held(DeviceSpec device_spec); + void reset_alsa_mutex_held(DeviceSpec device_spec); + std::map get_devices_mutex_held() const; + void update_meters(const std::vector &samples); + void send_audio_level_callback(); unsigned num_cards; - mutable std::mutex audio_mutex; + mutable std::timed_mutex audio_mutex; - AudioDevice cards[MAX_CARDS]; // Under 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 available_alsa_cards; StereoFilter locut; // Default cutoff 120 Hz, 24 dB/oct. std::atomic locut_cutoff_hz; @@ -205,8 +254,6 @@ private: 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 loudness_lufs{ref_level_lufs}; - StereoCompressor limiter; std::atomic limiter_threshold_dbfs{ref_level_dbfs + 4.0f}; // 4 dB. std::atomic limiter_enabled{true}; @@ -219,6 +266,13 @@ private: InputMapping input_mapping; // Under audio_mutex. std::atomic 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 peak{0.0f}; }; #endif // !defined(_AUDIO_MIXER_H)