further small reframer fixes, workaround distortion bug by wrapping sdl

input with reframer (magic fix, someone pls tell my why)

src/frame_streams/audio_stream_pop_reframer.hpp

@@ -30,41 +30,49 @@ struct AudioStreamPopReFramer : public FrameStream2I<AudioFrame2> {
 
 	~AudioStreamPopReFramer(void) {}
 
+	size_t getDesiredSize(void) const {
+		return _frame_length_ms * _sample_rate * _channels / 1000;
+	}
+
 	int32_t size(void) override { return -1; }
 
 	std::optional<AudioFrame2> pop(void) override {
-		auto new_in = _stream.pop();
+		do {
-		if (new_in.has_value()) {
+			auto new_in = _stream.pop();
-			auto& new_value = new_in.value();
+			if (new_in.has_value()) {
+				auto& new_value = new_in.value();
 
-			// changed
+				// changed
-			if (_sample_rate != new_value.sample_rate || _channels != new_value.channels) {
+				if (_sample_rate != new_value.sample_rate || _channels != new_value.channels) {
-				if (_channels != 0) {
+					//if (_channels != 0) {
-					//std::cerr << "ReFramer warning: reconfiguring, dropping buffer\n";
+					//    std::cerr << "ReFramer warning: reconfiguring, dropping buffer\n";
+					//}
+					_sample_rate = new_value.sample_rate;
+					_channels = new_value.channels;
+
+					// buffer does not exist or config changed and we discard
+					_buffer = {};
 				}
-				_sample_rate = new_value.sample_rate;
-				_channels = new_value.channels;
 
-				// buffer does not exist or config changed and we discard
+				//std::cout << "new incoming frame is " << new_value.getSpan().size/new_value.channels*1000/new_value.sample_rate << "ms\n";
-				_buffer = {};
-			}
 
+				auto new_span = new_value.getSpan();
 
-			//std::cout << "new incoming frame is " << new_value.getSpan().size/new_value.channels*1000/new_value.sample_rate << "ms\n";
+				if (_buffer.empty()) {
-
+					_buffer = {new_span.cbegin(), new_span.cend()};
-			auto new_span = new_value.getSpan();
+				} else {
-
+					_buffer.insert(_buffer.cend(), new_span.cbegin(), new_span.cend());
-			if (_buffer.empty()) {
+				}
-				_buffer = {new_span.cbegin(), new_span.cend()};
+			} else if (_buffer.empty()) {
+				// first pop might result in invalid state
+				return std::nullopt;
 			} else {
-				_buffer.insert(_buffer.cend(), new_span.cbegin(), new_span.cend());
+				// inner stream pop did not give a new value
+				break; // out of loop
 			}
-		} else if (_buffer.empty()) {
+		} while (_buffer.size() < getDesiredSize());
-			// first pop might result in invalid state
-			return std::nullopt;
-		}
 
-		const size_t desired_size {_frame_length_ms * _sample_rate * _channels / 1000};
+		const auto desired_size = getDesiredSize();
 
 		// > threshold?
 		if (_buffer.size() < desired_size) {

src/frame_streams/sdl/sdl_audio2_frame_stream2.cpp

@@ -4,6 +4,8 @@
 #include <iostream>
 #include <optional>
 
+#include "../audio_stream_pop_reframer.hpp"
+
 // "thin" wrapper around sdl audio streams
 // we dont needs to get fance, as they already provide everything we need
 struct SDLAudio2StreamReader : public AudioFrame2Stream2I {
@@ -58,7 +60,7 @@ struct SDLAudio2StreamReader : public AudioFrame2Stream2I {
 			return std::nullopt;
 		}
 
-		return AudioFrame2 {
+		return AudioFrame2{
 			_sample_rate, _channels,
 			Span<int16_t>(_buffer.data(), read_bytes/sizeof(int16_t)),
 		};
@@ -127,11 +129,17 @@ std::shared_ptr<FrameStream2I<AudioFrame2>> SDLAudio2InputDevice::subscribe(void
 	// error check
 	SDL_BindAudioStream(_virtual_device_id, sdl_stream);
 
-	auto new_stream = std::make_shared<SDLAudio2StreamReader>();
+	//auto new_stream = std::make_shared<SDLAudio2StreamReader>();
-	// TODO: move to ctr
+	//// TODO: move to ctr
-	new_stream->_stream = {sdl_stream, &SDL_DestroyAudioStream};
+	//new_stream->_stream = {sdl_stream, &SDL_DestroyAudioStream};
-	new_stream->_sample_rate = spec.freq;
+	//new_stream->_sample_rate = spec.freq;
-	new_stream->_channels = spec.channels;
+	//new_stream->_channels = spec.channels;
+
+	auto new_stream = std::make_shared<AudioStreamPopReFramer<SDLAudio2StreamReader>>();
+	new_stream->_stream._stream = {sdl_stream, &SDL_DestroyAudioStream};
+	new_stream->_stream._sample_rate = spec.freq;
+	new_stream->_stream._channels = spec.channels;
+	new_stream->_frame_length_ms = 5; // WHY DOES THIS FIX MY ISSUE !!!
 
 	_streams.emplace_back(new_stream);
 

src/frame_streams/test_pop_reframer.cpp

@@ -81,15 +81,9 @@ int main(void) {
 		}
 
 		stream.push(f1);
-
-		{
-			auto ret_opt = stream.pop();
-			assert(!ret_opt);
-		}
-
-		// push the other half
 		stream.push(f2);
 
+		// supposed to combine both
 		auto ret_opt = stream.pop();
 		assert(ret_opt);
 

src/tox_av_voip_model.cpp

@@ -58,7 +58,8 @@ struct ToxAVCallAudioSink : public FrameStream2SinkI<AudioFrame2> {
 			return nullptr;
 		}
 
-		_writer = std::make_shared<AudioStreamPopReFramer<LockedFrameStream2<AudioFrame2>>>();
+		// 20ms for now, 10ms would work too, further investigate stutters at 5ms (probably too slow interval rate)
+		_writer = std::make_shared<AudioStreamPopReFramer<LockedFrameStream2<AudioFrame2>>>(20);
 
 		return _writer;
 	}