Merge pull request #20 from degasus/pulseaudioRewrite

Pulseaudio: rewrite the pa backend with the async api

CMakeLists.txt

@@ -363,7 +363,7 @@ if(NOT ANDROID)
 		message("bluez NOT found, disabling bluetooth support")
 	endif(BLUEZ_FOUND)
 
-	check_lib(PULSEAUDIO libpulse-simple QUIET)
+	check_lib(PULSEAUDIO libpulse QUIET)
 	if(PULSEAUDIO_FOUND)
 		add_definitions(-DHAVE_PULSEAUDIO=1)
 		message("PulseAudio found, enabling PulseAudio sound backend")

Source/Core/AudioCommon/PulseAudioStream.cpp

@@ -11,30 +11,28 @@
 
 namespace
 {
-const size_t BUFFER_SAMPLES = 512;
+const size_t BUFFER_SAMPLES = 512; // ~10 ms
 const size_t CHANNEL_COUNT = 2;
-const size_t BUFFER_SIZE = BUFFER_SAMPLES * CHANNEL_COUNT;
+const size_t BUFFER_SIZE = BUFFER_SAMPLES * CHANNEL_COUNT * sizeof(s16);
 }
 
 PulseAudio::PulseAudio(CMixer *mixer)
 	: SoundStream(mixer)
-	, mix_buffer(BUFFER_SIZE)
-	, thread()
-	, run_thread()
-	, pa()
+	, m_thread()
+	, m_run_thread()
 {}
 
 bool PulseAudio::Start()
 {
-	run_thread = true;
-	thread = std::thread(std::mem_fun(&PulseAudio::SoundLoop), this);
+	m_run_thread = true;
+	m_thread = std::thread(std::mem_fun(&PulseAudio::SoundLoop), this);
 	return true;
 }
 
 void PulseAudio::Stop()
 {
-	run_thread = false;
-	thread.join();
+	m_run_thread = false;
+	m_thread.join();
 }
 
 void PulseAudio::Update()
@@ -49,51 +47,129 @@ void PulseAudio::SoundLoop()
 
 	if (PulseInit())
 	{
-		while (run_thread)
-		{
-			m_mixer->Mix(&mix_buffer[0], mix_buffer.size() / CHANNEL_COUNT);
-			Write(&mix_buffer[0], mix_buffer.size() * sizeof(s16));
-		}
+		while (m_run_thread.load() && m_pa_connected == 1 && m_pa_error >= 0)
+			m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, NULL);
+
+		if(m_pa_error < 0)
+			ERROR_LOG(AUDIO, "PulseAudio error: %s", pa_strerror(m_pa_error));
 
 		PulseShutdown();
 	}
 }
 
 bool PulseAudio::PulseInit()
 {
-	pa_sample_spec ss = {};
+	m_pa_error = 0;
+	m_pa_connected = 0;
+
+	// create pulseaudio main loop and context
+	// also register the async state callback which is called when the connection to the pa server has changed
+	m_pa_ml = pa_mainloop_new();
+	m_pa_mlapi = pa_mainloop_get_api(m_pa_ml);
+	m_pa_ctx = pa_context_new(m_pa_mlapi, "dolphin-emu");
+	m_pa_error = pa_context_connect(m_pa_ctx, NULL, PA_CONTEXT_NOFLAGS, NULL);
+	pa_context_set_state_callback(m_pa_ctx, StateCallback, this);
+
+	// wait until we're connected to the pulseaudio server
+	while (m_pa_connected == 0 && m_pa_error >= 0)
+		m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, NULL);
+
+	if (m_pa_connected == 2 || m_pa_error < 0)
+	{
+		ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
+		return false;
+	}
+
+	// create a new audio stream with our sample format
+	// also connect the callbacks for this stream
+	pa_sample_spec ss;
 	ss.format = PA_SAMPLE_S16LE;
 	ss.channels = 2;
 	ss.rate = m_mixer->GetSampleRate();
-
-	int error;
-	pa = pa_simple_new(nullptr, "dolphin-emu", PA_STREAM_PLAYBACK,
-		nullptr, "audio", &ss, nullptr, nullptr, &error);
-
-	if (!pa)
+	m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, NULL);
+	pa_stream_set_write_callback(m_pa_s, WriteCallback, this);
+	pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this);
+
+	// connect this audio stream to the default audio playback
+	// limit buffersize to reduce latency
+	m_pa_ba.fragsize = -1;
+	m_pa_ba.maxlength = -1;          // max buffer, so also max latency
+	m_pa_ba.minreq = -1;             // don't read every byte, try to group them _a bit_
+	m_pa_ba.prebuf = -1;             // start as early as possible
+	m_pa_ba.tlength = BUFFER_SIZE;   // designed latency, only change this flag for low latency output
+	pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
+	m_pa_error = pa_stream_connect_playback(m_pa_s, NULL, &m_pa_ba, flags, NULL, NULL);
+	if (m_pa_error < 0)
 	{
-		ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s",
-			pa_strerror(error));
+		ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
 		return false;
 	}
-	else
-	{
-		NOTICE_LOG(AUDIO, "Pulse successfully initialized.");
-		return true;
-	}
+
+	INFO_LOG(AUDIO, "Pulse successfully initialized");
+	return true;
 }
 
 void PulseAudio::PulseShutdown()
 {
-	pa_simple_free(pa);
+	pa_context_disconnect(m_pa_ctx);
+	pa_context_unref(m_pa_ctx);
+	pa_mainloop_free(m_pa_ml);
 }
 
-void PulseAudio::Write(const void *data, size_t length)
+void PulseAudio::StateCallback(pa_context* c)
 {
-	int error;
-	if (pa_simple_write(pa, data, length, &error) < 0)
+	pa_context_state_t state = pa_context_get_state(c);
+	switch (state)
 	{
-		ERROR_LOG(AUDIO, "PulseAudio failed to write data: %s",
-			pa_strerror(error));
+	case PA_CONTEXT_FAILED:
+	case PA_CONTEXT_TERMINATED:
+		m_pa_connected = 2;
+		break;
+	case PA_CONTEXT_READY:
+		m_pa_connected = 1;
+		break;
+	default:
+		break;
 	}
 }
+// on underflow, increase pulseaudio latency in ~10ms steps
+void PulseAudio::UnderflowCallback(pa_stream* s)
+{
+	m_pa_ba.tlength += BUFFER_SIZE;
+	pa_stream_set_buffer_attr(s, &m_pa_ba, NULL, NULL);
+
+	WARN_LOG(AUDIO, "pulseaudio underflow, new latency: %d bytes", m_pa_ba.tlength);
+}
+
+void PulseAudio::WriteCallback(pa_stream* s, size_t length)
+{
+	// fetch dst buffer directly from pulseaudio, so no memcpy is needed
+	void* buffer;
+	m_pa_error = pa_stream_begin_write(s, &buffer, &length);
+
+	if (!buffer || m_pa_error < 0)
+		return; // error will be printed from main loop
+
+	m_mixer->Mix((s16*) buffer, length / sizeof(s16) / CHANNEL_COUNT);
+	m_pa_error = pa_stream_write(s, buffer, length, NULL, 0, PA_SEEK_RELATIVE);
+}
+
+// Callbacks that forward to internal methods (required because PulseAudio is a C API).
+
+void PulseAudio::StateCallback(pa_context* c, void* userdata)
+{
+	PulseAudio* p = (PulseAudio*) userdata;
+	p->StateCallback(c);
+}
+
+void PulseAudio::UnderflowCallback(pa_stream* s, void* userdata)
+{
+	PulseAudio* p = (PulseAudio*) userdata;
+	p->UnderflowCallback(s);
+}
+
+void PulseAudio::WriteCallback(pa_stream* s, size_t length, void* userdata)
+{
+	PulseAudio* p = (PulseAudio*) userdata;
+	p->WriteCallback(s, length);
+}

Source/Core/AudioCommon/PulseAudioStream.h

@@ -6,17 +6,16 @@
 #define _PULSE_AUDIO_STREAM_H
 
 #if defined(HAVE_PULSEAUDIO) && HAVE_PULSEAUDIO
-#include <pulse/simple.h>
-#include <pulse/error.h>
+#include <pulse/pulseaudio.h>
 #endif
 
+#include <atomic>
+
 #include "Common.h"
 #include "SoundStream.h"
 
 #include "Thread.h"
 
-#include <vector>
-
 class PulseAudio : public SoundStream
 {
 #if defined(HAVE_PULSEAUDIO) && HAVE_PULSEAUDIO
@@ -32,18 +31,31 @@ class PulseAudio : public SoundStream
 
 	virtual void Update();
 
+	void StateCallback(pa_context *c);
+	void WriteCallback(pa_stream *s, size_t length);
+	void UnderflowCallback(pa_stream *s);
+
 private:
 	virtual void SoundLoop();
 
 	bool PulseInit();
 	void PulseShutdown();
-	void Write(const void *data, size_t bytes);
 
-	std::vector<s16> mix_buffer;
-	std::thread thread;
-	volatile bool run_thread;
+	// wrapper callback functions, last parameter _must_ be PulseAudio*
+	static void StateCallback(pa_context *c, void *userdata);
+	static void WriteCallback(pa_stream *s, size_t length, void *userdata);
+	static void UnderflowCallback(pa_stream *s, void *userdata);
+
+	std::thread m_thread;
+	std::atomic<bool> m_run_thread;
 
-	pa_simple* pa;
+	int m_pa_error;
+	int m_pa_connected;
+	pa_mainloop *m_pa_ml;
+	pa_mainloop_api *m_pa_mlapi;
+	pa_context *m_pa_ctx;
+	pa_stream *m_pa_s;
+	pa_buffer_attr m_pa_ba;
 #else
 public:
 	PulseAudio(CMixer *mixer) : SoundStream(mixer) {}