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			173 lines
		
	
	
	
		
			4.7 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			173 lines
		
	
	
	
		
			4.7 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
| // Copyright 2013 Dolphin Emulator Project
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| // Licensed under GPLv2
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| // Refer to the license.txt file included.
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| 
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| #include "AudioCommon/PulseAudioStream.h"
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| #include "Common/CommonTypes.h"
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| #include "Common/Thread.h"
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| 
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| namespace
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| {
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| const size_t BUFFER_SAMPLES = 512; // ~10 ms
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| const size_t CHANNEL_COUNT = 2;
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| const size_t BUFFER_SIZE = BUFFER_SAMPLES * CHANNEL_COUNT * sizeof(s16);
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| }
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| 
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| PulseAudio::PulseAudio(CMixer *mixer)
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| 	: SoundStream(mixer)
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| 	, m_thread()
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| 	, m_run_thread()
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| {
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| }
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| 
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| bool PulseAudio::Start()
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| {
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| 	m_run_thread = true;
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| 	m_thread = std::thread(&PulseAudio::SoundLoop, this);
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| 	return true;
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| }
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| 
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| void PulseAudio::Stop()
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| {
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| 	m_run_thread = false;
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| 	m_thread.join();
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| }
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| 
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| void PulseAudio::Update()
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| {
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| 	// don't need to do anything here.
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| }
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| 
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| // Called on audio thread.
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| void PulseAudio::SoundLoop()
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| {
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| 	Common::SetCurrentThreadName("Audio thread - pulse");
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| 
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| 	if (PulseInit())
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| 	{
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| 		while (m_run_thread.load() && m_pa_connected == 1 && m_pa_error >= 0)
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| 			m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
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| 
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| 		if (m_pa_error < 0)
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| 			ERROR_LOG(AUDIO, "PulseAudio error: %s", pa_strerror(m_pa_error));
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| 
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| 		PulseShutdown();
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| 	}
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| }
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| 
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| bool PulseAudio::PulseInit()
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| {
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| 	m_pa_error = 0;
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| 	m_pa_connected = 0;
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| 
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| 	// create pulseaudio main loop and context
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| 	// also register the async state callback which is called when the connection to the pa server has changed
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| 	m_pa_ml = pa_mainloop_new();
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| 	m_pa_mlapi = pa_mainloop_get_api(m_pa_ml);
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| 	m_pa_ctx = pa_context_new(m_pa_mlapi, "dolphin-emu");
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| 	m_pa_error = pa_context_connect(m_pa_ctx, nullptr, PA_CONTEXT_NOFLAGS, nullptr);
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| 	pa_context_set_state_callback(m_pa_ctx, StateCallback, this);
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| 
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| 	// wait until we're connected to the pulseaudio server
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| 	while (m_pa_connected == 0 && m_pa_error >= 0)
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| 		m_pa_error = pa_mainloop_iterate(m_pa_ml, 1, nullptr);
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| 
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| 	if (m_pa_connected == 2 || m_pa_error < 0)
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| 	{
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| 		ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
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| 		return false;
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| 	}
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| 
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| 	// create a new audio stream with our sample format
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| 	// also connect the callbacks for this stream
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| 	pa_sample_spec ss;
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| 	ss.format = PA_SAMPLE_S16LE;
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| 	ss.channels = 2;
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| 	ss.rate = m_mixer->GetSampleRate();
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| 	m_pa_s = pa_stream_new(m_pa_ctx, "Playback", &ss, nullptr);
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| 	pa_stream_set_write_callback(m_pa_s, WriteCallback, this);
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| 	pa_stream_set_underflow_callback(m_pa_s, UnderflowCallback, this);
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| 
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| 	// connect this audio stream to the default audio playback
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| 	// limit buffersize to reduce latency
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| 	m_pa_ba.fragsize = -1;
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| 	m_pa_ba.maxlength = -1;          // max buffer, so also max latency
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| 	m_pa_ba.minreq = -1;             // don't read every byte, try to group them _a bit_
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| 	m_pa_ba.prebuf = -1;             // start as early as possible
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| 	m_pa_ba.tlength = BUFFER_SIZE;   // designed latency, only change this flag for low latency output
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| 	pa_stream_flags flags = pa_stream_flags(PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_AUTO_TIMING_UPDATE);
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| 	m_pa_error = pa_stream_connect_playback(m_pa_s, nullptr, &m_pa_ba, flags, nullptr, nullptr);
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| 	if (m_pa_error < 0)
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| 	{
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| 		ERROR_LOG(AUDIO, "PulseAudio failed to initialize: %s", pa_strerror(m_pa_error));
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| 		return false;
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| 	}
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| 
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| 	INFO_LOG(AUDIO, "Pulse successfully initialized");
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| 	return true;
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| }
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| 
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| void PulseAudio::PulseShutdown()
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| {
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| 	pa_context_disconnect(m_pa_ctx);
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| 	pa_context_unref(m_pa_ctx);
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| 	pa_mainloop_free(m_pa_ml);
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| }
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| 
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| void PulseAudio::StateCallback(pa_context* c)
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| {
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| 	pa_context_state_t state = pa_context_get_state(c);
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| 	switch (state)
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| 	{
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| 	case PA_CONTEXT_FAILED:
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| 	case PA_CONTEXT_TERMINATED:
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| 		m_pa_connected = 2;
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| 		break;
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| 	case PA_CONTEXT_READY:
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| 		m_pa_connected = 1;
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| 		break;
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| 	default:
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| 		break;
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| 	}
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| }
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| // on underflow, increase pulseaudio latency in ~10ms steps
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| void PulseAudio::UnderflowCallback(pa_stream* s)
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| {
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| 	m_pa_ba.tlength += BUFFER_SIZE;
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| 	pa_stream_set_buffer_attr(s, &m_pa_ba, nullptr, nullptr);
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| 
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| 	WARN_LOG(AUDIO, "pulseaudio underflow, new latency: %d bytes", m_pa_ba.tlength);
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| }
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| 
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| void PulseAudio::WriteCallback(pa_stream* s, size_t length)
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| {
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| 	// fetch dst buffer directly from pulseaudio, so no memcpy is needed
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| 	void* buffer;
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| 	m_pa_error = pa_stream_begin_write(s, &buffer, &length);
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| 
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| 	if (!buffer || m_pa_error < 0)
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| 		return; // error will be printed from main loop
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| 
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| 	m_mixer->Mix((s16*) buffer, length / sizeof(s16) / CHANNEL_COUNT);
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| 	m_pa_error = pa_stream_write(s, buffer, length, nullptr, 0, PA_SEEK_RELATIVE);
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| }
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| 
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| // Callbacks that forward to internal methods (required because PulseAudio is a C API).
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| 
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| void PulseAudio::StateCallback(pa_context* c, void* userdata)
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| {
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| 	PulseAudio* p = (PulseAudio*) userdata;
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| 	p->StateCallback(c);
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| }
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| 
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| void PulseAudio::UnderflowCallback(pa_stream* s, void* userdata)
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| {
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| 	PulseAudio* p = (PulseAudio*) userdata;
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| 	p->UnderflowCallback(s);
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| }
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| 
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| void PulseAudio::WriteCallback(pa_stream* s, size_t length, void* userdata)
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| {
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| 	PulseAudio* p = (PulseAudio*) userdata;
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| 	p->WriteCallback(s, length);
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| }
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