class AudioMixer {
public:
AudioMixer(unsigned num_cards);
- ~AudioMixer();
void reset_resampler(DeviceSpec device_spec);
void reset_meters();
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);
+ // 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(double pts, unsigned num_samples, ResamplingQueue::RateAdjustmentPolicy rate_adjustment_policy);
void set_fader_volume(unsigned bus_index, float level_db) { fader_volume_db[bus_index] = level_db; }
- std::map<DeviceSpec, DeviceInfo> get_devices() const;
+
+ // 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);
+ }
+
void set_name(DeviceSpec device_spec, const std::string &name);
void set_input_mapping(const InputMapping &input_mapping);
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;
unsigned num_cards;
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::atomic<float> locut_cutoff_hz{120};
StereoFilter locut[MAX_BUSES]; // Default cutoff 120 Hz, 24 dB/oct.
std::atomic<float> peak{0.0f};
};
+extern AudioMixer *global_audio_mixer;
+
#endif // !defined(_AUDIO_MIXER_H)