X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=audio_mixer.h;h=855295dcd60f0bf207f435c75484a763eba77b89;hb=e4c99b681aceb0d960f1d56acc9dafc83643854b;hp=34e7dc075a1d614073b1a5c430f1f9bccf3e49ee;hpb=f3e2ecc05ad9df6b428b80bd32ad2e8d3f7192a1;p=nageru

diff --git a/audio_mixer.h b/audio_mixer.h
index 34e7dc0..855295d 100644
--- a/audio_mixer.h
+++ b/audio_mixer.h
@@ -82,7 +82,6 @@ struct InputMapping {
 class AudioMixer {
 public:
 	AudioMixer(unsigned num_cards);
-	~AudioMixer();
 	void reset_resampler(DeviceSpec device_spec);
 	void reset_meters();
 
@@ -94,10 +93,28 @@ public:
 	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);
@@ -253,33 +270,35 @@ private:
 		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;
+	std::atomic<float> locut_cutoff_hz{120};
 	StereoFilter locut[MAX_BUSES];  // Default cutoff 120 Hz, 24 dB/oct.
 	std::atomic<bool> locut_enabled[MAX_BUSES];
 	StereoFilter eq[MAX_BUSES][NUM_EQ_BANDS];  // The one for EQBand::MID isn't actually used (see comments in apply_eq()).
@@ -326,4 +345,6 @@ private:
 	std::atomic<float> peak{0.0f};
 };
 
+extern AudioMixer *global_audio_mixer;
+
 #endif  // !defined(_AUDIO_MIXER_H)