#include <set>
#include <vector>
+#include "alsa_input.h"
#include "bmusb/bmusb.h"
#include "db.h"
#include "defs.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);
+
+ // 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().
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;
unsigned num_cards;
- mutable std::mutex audio_mutex;
+ mutable std::timed_mutex audio_mutex;
+
+ AudioDevice video_cards[MAX_VIDEO_CARDS]; // Under audio_mutex.
- AudioDevice cards[MAX_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;