Merge branch 'master' into stable

1 files changed, 247 insertions, 0 deletions

diff --git a/Source/Core/AudioCommon/PulseAudioStream.cpp b/Source/Core/AudioCommon/PulseAudioStream.cpp
new file mode 100644
index 0000000000..1932ed031a
--- /dev/null
+++ b/Source/Core/AudioCommon/PulseAudioStream.cpp
@@ -0,0 +1,247 @@
+// Copyright 2009 Dolphin Emulator Project
+// Licensed under GPLv2+
+// Refer to the license.txt file included.
+
+#include "AudioCommon/DPL2Decoder.h"
+#include "AudioCommon/PulseAudioStream.h"
+#include "Common/CommonTypes.h"
+#include "Common/Thread.h"
+#include "Core/ConfigManager.h"
+
+namespace
+{
+const size_t BUFFER_SAMPLES = 512; // ~10 ms - needs to be at least 240 for surround
+}
+
+PulseAudio::PulseAudio()
+	: m_thread()
+	, m_run_thread()
+{
+}
+
+bool PulseAudio::Start()
+{
+	m_stereo = !SConfig::GetInstance().m_LocalCoreStartupParameter.bDPL2Decoder;
+	m_channels = m_stereo ? 2 : 5; // will tell PA we use a Stereo or 5.0 channel setup
+
+	NOTICE_LOG(AUDIO, "PulseAudio backend using %d channels", m_channels);
+
+	m_run_thread = true;
+	m_thread = std::thread(&PulseAudio::SoundLoop, this);
+
+	// Initialize DPL2 parameters
+	DPL2Reset();
+
+	return true;
+}
+
+void PulseAudio::Stop()
+{
+	m_run_thread = false;
+	m_thread.join();
+}
+
+void PulseAudio::Update()
+{
+	// don't need to do anything here.
+}
+
+// Called on audio thread.
+void PulseAudio::SoundLoop()
+{
+	Common::SetCurrentThreadName("Audio thread - pulse");
+
+	if (PulseInit())
+	{
+		while (m_run_thread.load() && m_pa_connected == 1 && m_pa_error >= 0)
+			m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
+
+		if (m_pa_error < 0)
+			ERROR_LOG(AUDIO, "PulseAudio error: %s", pa_strerror(m_pa_error));
+
+		PulseShutdown();
+	}
+}
+
+bool PulseAudio::PulseInit()
+{
+	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, nullptr, PA_CONTEXT_NOFLAGS, nullptr);
+	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, nullptr);
+
+	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;
+	pa_channel_map channel_map;
+	pa_channel_map* channel_map_p = nullptr; // auto channel map
+	if (m_stereo)
+	{
+		ss.format = PA_SAMPLE_S16LE;
+		m_bytespersample = sizeof(s16);
+	}
+	else
+	{
+		// surround is remixed in floats, use a float PA buffer to save another conversion
+		ss.format = PA_SAMPLE_FLOAT32NE;
+		m_bytespersample = sizeof(float);
+
+		channel_map_p = &channel_map; // explicit channel map:
+		channel_map.channels = 5;
+		channel_map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT;
+		channel_map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT;
+		channel_map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER;
+		channel_map.map[3] = PA_CHANNEL_POSITION_REAR_LEFT;
+		channel_map.map[4] = PA_CHANNEL_POSITION_REAR_RIGHT;
+	}
+	ss.channels = m_channels;
+	ss.rate = m_mixer->GetSampleRate();
+	assert(pa_sample_spec_valid(&ss));
+	m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, channel_map_p);
+	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_SAMPLES * m_channels * m_bytespersample; // 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, nullptr, &m_pa_ba, flags, nullptr, nullptr);
+	if (m_pa_error < 0)
+	{
+		ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
+		return false;
+	}
+
+	INFO_LOG(AUDIO, "Pulse successfully initialized");
+	return true;
+}
+
+void PulseAudio::PulseShutdown()
+{
+	pa_context_disconnect(m_pa_ctx);
+	pa_context_unref(m_pa_ctx);
+	pa_mainloop_free(m_pa_ml);
+}
+
+void PulseAudio::StateCallback(pa_context* c)
+{
+	pa_context_state_t state = pa_context_get_state(c);
+	switch (state)
+	{
+	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_SAMPLES * m_channels * m_bytespersample;
+	pa_stream_set_buffer_attr(s, &m_pa_ba, nullptr, nullptr);
+
+	WARN_LOG(AUDIO, "pulseaudio underflow, new latency: %d bytes", m_pa_ba.tlength);
+}
+
+void PulseAudio::WriteCallback(pa_stream* s, size_t length)
+{
+	int bytes_per_frame = m_channels * m_bytespersample;
+	int frames = (length / bytes_per_frame);
+	size_t trunc_length = frames * bytes_per_frame;
+
+	// fetch dst buffer directly from pulseaudio, so no memcpy is needed
+	void* buffer;
+	m_pa_error = pa_stream_begin_write(s, &buffer, &trunc_length);
+
+	if (!buffer || m_pa_error < 0)
+		return; // error will be printed from main loop
+
+	if (m_stereo)
+	{
+		// use the raw s16 stereo mix
+		m_mixer->Mix((s16*) buffer, frames);
+	}
+	else
+	{
+		// get a floating point mix
+		s16 s16buffer_stereo[frames * 2];
+		m_mixer->Mix(s16buffer_stereo, frames); // implicitly mixes to 16-bit stereo
+
+		float floatbuffer_stereo[frames * 2];
+		// s16 to float
+		for (int i=0; i < frames * 2; ++i)
+		{
+			floatbuffer_stereo[i] = s16buffer_stereo[i] / float(1 << 15);
+		}
+
+		if (m_channels == 5) // Extract dpl2/5.0 Surround
+		{
+			float floatbuffer_6chan[frames * 6];
+			// DPL2Decode output: LEFTFRONT, RIGHTFRONT, CENTREFRONT, (sub), LEFTREAR, RIGHTREAR
+			DPL2Decode(floatbuffer_stereo, frames, floatbuffer_6chan);
+
+			// Discard the subwoofer channel - DPL2Decode generates a pretty
+			// good 5.0 but not a good 5.1 output.
+			const int dpl2_to_5chan[] = {0,1,2,4,5};
+			for (int i=0; i < frames; ++i)
+			{
+				for (int j=0; j < m_channels; ++j)
+				{
+					((float*)buffer)[m_channels * i + j] = floatbuffer_6chan[6 * i + dpl2_to_5chan[j]];
+				}
+			}
+		}
+		else
+		{
+			ERROR_LOG(AUDIO, "Unsupported number of PA channels requested: %d", (int)m_channels);
+			return;
+		}
+	}
+
+	m_pa_error = pa_stream_write(s, buffer, trunc_length, nullptr, 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);
+}