Add audio settings information

README.md

@@ -5,7 +5,7 @@ The main goal is to offer the ability to use the Capture Card with a TV, via ful
 
 ## Features
 
-- Performance-focused design, with low latency for both audio and video (measured to oscillate between 1 and 2 frames).
+- Performance-focused design, with low latency for both audio and video (measured to oscillate between 1 and 2 frames on 120hz displays).
 - Option to split the screens in separate windows, to address them separately.
 - Make your game run in fullscreen mode. If you own multiple displays, you can even use one per-window.
 - Many builtin crop options for the screens.
@@ -140,4 +140,10 @@ On Windows, the profiles can be found in the ".config/cc3dsfs" folder inside the
 xattr -c ./cc3dsfs.app
 ```
 - It can also be achieved by opening a terminal, typing "xattr -c ", dragging the application to the terminal, and finally pressing Enter.
-
+- Certain TVs/Monitors may add some audio delay for the purpose of video/lip syncing. If you're experiencing too much audio delay when using this software, try checking in the TV/Monitor settings whether you can reduce that added delay. One of the names used for that setting is "Lip Sync", or something along that line.
+- For the best audio latency and quality, it may be needed to change OpenAL's sound settings. Their location is OS-specific. On linux, you may use the "alsoft-config" command. In particular, the following parameters may help:
+```
+frequency = 96000
+periods = 2
+period_size = 64
+```

include/audio.hpp

@@ -5,9 +5,6 @@
 #include <queue>
 #include "audio_data.hpp"
 #include "utils.hpp"
-#include "audio.hpp"
-#include "capture_structs.hpp"
-#include "hw_defs.hpp"
 
 struct Sample {
 	Sample(sf::Int16 *bytes, std::size_t size, float time);
@@ -24,6 +21,7 @@ public:
 	ConsumerMutex samples_wait;
 
 	Audio(AudioData *audio_data);
+	~Audio();
 	void update_volume();
 	void start_audio();
 	void stop_audio();
@@ -33,7 +31,7 @@ private:
 	int final_volume = -1;
 	bool terminate;
 	int num_consecutive_fast_seek;
-	sf::Int16 buffer[MAX_SAMPLES_IN];
+	sf::Int16 *buffer;
 	std::chrono::time_point<std::chrono::high_resolution_clock> clock_time_start;
 
 	bool onGetData(sf::SoundStream::Chunk &data) override;

include/hw_defs.hpp

@@ -2,7 +2,7 @@
 #define __HW_DEFS_HPP
 
 #define AUDIO_CHANNELS 2
-#define SAMPLE_RATE 48000
+#define SAMPLE_RATE 96000
 #define PITCH_RATE (67027964 / ((float)SAMPLE_RATE * 2048))
 
 #define TOP_WIDTH_3DS 400

source/audio.cpp

@@ -17,18 +17,18 @@ Sample::Sample(sf::Int16 *bytes, std::size_t size, float time) : bytes(bytes), s
 Audio::Audio(AudioData *audio_data) {
 	this->audio_data = audio_data;
 	// Consume old events
+	this->buffer = new sf::Int16[MAX_SAMPLES_IN];
 	this->audio_data->check_audio_restart_request();
 	sf::SoundStream::initialize(AUDIO_CHANNELS, SAMPLE_RATE);
 	this->setPitch(PITCH_RATE);
-	#if (SFML_VERSION_MAJOR > 2) || ((SFML_VERSION_MAJOR == 2) && (SFML_VERSION_MINOR >= 6))
-	// Since we receive data every 16.6 ms, this is useless,
-	// and only increases CPU load... (Default is 10 ms)
-	//sf::SoundStream::setProcessingInterval(sf::Time::Zero);
-	#endif
 	start_audio();
 	setVolume(0);
 	this->final_volume = 0;
 }
+
+Audio::~Audio() {
+	delete []this->buffer;
+}
 	
 void Audio::update_volume() {
 	int new_final_volume = this->audio_data->get_final_volume();