dolphin/Source/Core/AudioCommon/Src/Mixer.cpp

// Copyright 2013 Dolphin Emulator Project
// Licensed under GPLv2
// Refer to the license.txt file included.

#include "Atomic.h"
#include "Mixer.h"
#include "AudioCommon.h"
#include "CPUDetect.h"
#include "../../Core/Src/Host.h"

#include "../../Core/Src/HW/AudioInterface.h"

// UGLINESS
#include "../../Core/Src/PowerPC/PowerPC.h"

#if _M_SSE >= 0x301 && !(defined __GNUC__ && !defined __SSSE3__)
#include <tmmintrin.h>
#endif

// Executed from sound stream thread
unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
{
	if (!samples)
		return 0;

	std::lock_guard<std::mutex> lk(m_csMixing);

	if (PowerPC::GetState() != PowerPC::CPU_RUNNING)
	{
		// Silence
		memset(samples, 0, numSamples * 4);
		return numSamples;
	}

	unsigned int numLeft = GetNumSamples();
	if (m_AIplaying) {
		if (numLeft < numSamples)//cannot do much about this
			m_AIplaying = false;
		if (numLeft < MAX_SAMPLES/4)//low watermark
			m_AIplaying = false;
	} else {
		if (numLeft > MAX_SAMPLES/2)//high watermark
			m_AIplaying = true;
	}

	// Cache access in non-volatile variable
	// This is the only function changing the read value, so it's safe to
	// cache it locally although it's written here.
	// The writing pointer will be modified outside, but it will only increase,
	// so we will just ignore new written data while interpolating.
	// Without this cache, the compiler wouldn't be allowed to optimize the
	// interpolation loop.
	u32 indexR = Common::AtomicLoad(m_indexR);
	u32 indexW = Common::AtomicLoad(m_indexW);

	if (m_AIplaying) {
		numLeft = (numLeft > numSamples) ? numSamples : numLeft;

		if (AudioInterface::GetAIDSampleRate() == m_sampleRate) // (1:1)
		{
#if _M_SSE >= 0x301
			if (cpu_info.bSSSE3 && !((numLeft * 2) % 8))
			{
				static const __m128i sr_mask =
					_mm_set_epi32(0x0C0D0E0FL, 0x08090A0BL,
								  0x04050607L, 0x00010203L);

				for (unsigned int i = 0; i < numLeft * 2; i += 8)
				{
					_mm_storeu_si128((__m128i *)&samples[i], _mm_shuffle_epi8(_mm_loadu_si128((__m128i *)&m_buffer[(indexR + i) & INDEX_MASK]), sr_mask));
				}
			}
			else
#endif
			{
				for (unsigned int i = 0; i < numLeft * 2; i+=2)
				{
					samples[i] = Common::swap16(m_buffer[(indexR + i + 1) & INDEX_MASK]);
					samples[i+1] = Common::swap16(m_buffer[(indexR + i) & INDEX_MASK]);
				}
			}
			indexR += numLeft * 2;
		}
		else //linear interpolation
		{
			//render numleft sample pairs to samples[]
			//advance indexR with sample position
			//remember fractional offset

			static u32 frac = 0;
			const u32 ratio = (u32)( 65536.0f * (float)AudioInterface::GetAIDSampleRate() / (float)m_sampleRate );

			for (u32 i = 0; i < numLeft * 2; i+=2) {
				u32 indexR2 = indexR + 2; //next sample
				if ((indexR2 & INDEX_MASK) == (indexW & INDEX_MASK)) //..if it exists
					indexR2 = indexR;

				s16 l1 = Common::swap16(m_buffer[indexR & INDEX_MASK]); //current
				s16 l2 = Common::swap16(m_buffer[indexR2 & INDEX_MASK]); //next
				int sampleL = ((l1 << 16) + (l2 - l1) * (u16)frac)  >> 16;
				samples[i+1] = sampleL;

				s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]); //current
				s16 r2 = Common::swap16(m_buffer[(indexR2 + 1) & INDEX_MASK]); //next
				int sampleR = ((r1 << 16) + (r2 - r1) * (u16)frac)  >> 16;
				samples[i] = sampleR;

				frac += ratio;
				indexR += 2 * (u16)(frac >> 16);
				frac &= 0xffff;
			}
		}

	} else {
		numLeft = 0;
	}

	// Padding
	if (numSamples > numLeft)
	{
		unsigned short s[2];
		s[0] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
		s[1] = Common::swap16(m_buffer[(indexR - 2) & INDEX_MASK]);
		for (unsigned int i = numLeft*2; i < numSamples*2; i+=2)
			*(u32*)(samples+i) = *(u32*)(s);
//		memset(&samples[numLeft * 2], 0, (numSamples - numLeft) * 4);
	}

	// Flush cached variable
	Common::AtomicStore(m_indexR, indexR);

	//when logging, also throttle HLE audio
	if (m_logAudio) {
		if (m_AIplaying) {
			Premix(samples, numLeft);

			AudioInterface::Callback_GetStreaming(samples, numLeft, m_sampleRate);

			g_wave_writer.AddStereoSamples(samples, numLeft);
		}
	}
	else { 	//or mix as usual
		// Add the DSPHLE sound, re-sampling is done inside
		Premix(samples, numSamples);

		// Add the DTK Music
		// Re-sampling is done inside
		AudioInterface::Callback_GetStreaming(samples, numSamples, m_sampleRate);
	}

	return numSamples;
}


void CMixer::PushSamples(const short *samples, unsigned int num_samples)
{
	// Cache access in non-volatile variable
	// indexR isn't allowed to cache in the audio throttling loop as it
	// needs to get updates to not deadlock.
	u32 indexW = Common::AtomicLoad(m_indexW);

	if (m_throttle)
	{
		// The auto throttle function. This loop will put a ceiling on the CPU MHz.
		while (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
		{
			if (*PowerPC::GetStatePtr() != PowerPC::CPU_RUNNING || soundStream->IsMuted())
				break;
			// Shortcut key for Throttle Skipping
			if (Host_GetKeyState('\t'))
				break;
			SLEEP(1);
			soundStream->Update();
		}
	}

	// Check if we have enough free space
	// indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
	if (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
		return;

	// AyuanX: Actual re-sampling work has been moved to sound thread
	// to alleviate the workload on main thread
	// and we simply store raw data here to make fast mem copy
	int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
	if (over_bytes > 0)
	{
		memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
		memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
	}
	else
	{
		memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
	}

	Common::AtomicAdd(m_indexW, num_samples * 2);

	return;
}

unsigned int CMixer::GetNumSamples()
{
	// Guess how many samples would be available after interpolation.
	// As interpolation needs at least on sample from the future to
	// linear interpolate between them, one sample less is available.
	// We also can't say the current interpolation state (specially
	// the frac), so to be sure, subtract one again to be sure not
	// to underflow the fifo.

	u32 numSamples = ((Common::AtomicLoad(m_indexW) - Common::AtomicLoad(m_indexR)) & INDEX_MASK) / 2;

	if (AudioInterface::GetAIDSampleRate() == m_sampleRate)
		; //numSamples = numSamples; // 1:1
	else if (m_sampleRate == 48000 && AudioInterface::GetAIDSampleRate() == 32000)
		numSamples = numSamples * 3 / 2 - 2; // most common case
	else
		numSamples = numSamples * m_sampleRate / AudioInterface::GetAIDSampleRate() - 2;

	return numSamples;
}