ppsspp/Core/HW/StereoResampler.cpp

// Copyright (c) 2015- PPSSPP Project and Dolphin Project.

// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, version 2.0 or later versions.

// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
// GNU General Public License 2.0 for more details.

// A copy of the GPL 2.0 should have been included with the program.
// If not, see http://www.gnu.org/licenses/

// Official git repository and contact information can be found at
// https://github.com/hrydgard/ppsspp and http://www.ppsspp.org/.

// Adapted from Dolphin.

// 16 bit Stereo

// These must be powers of 2.
#define MAX_BUFSIZE_DEFAULT (4096) // 2*64ms - had to double it for nVidia Shield which has huge buffers
#define MAX_BUFSIZE_EXTRA   (8192)

#define TARGET_BUFSIZE_MARGIN 512

#define TARGET_BUFSIZE_DEFAULT 1680 // 40 ms
#define TARGET_BUFSIZE_EXTRA 3360 // 80 ms

#define MAX_FREQ_SHIFT  600.0f  // how far off can we be from 44100 Hz
#define CONTROL_FACTOR  0.2f // in freq_shift per fifo size offset
#define CONTROL_AVG     32.0f

#include "ppsspp_config.h"
#include <algorithm>
#include <cstring>
#include <atomic>

#include "Common/Common.h"
#include "Common/System/System.h"
#include "Common/Log.h"
#include "Common/Math/SIMDHeaders.h"
#include "Common/Math/CrossSIMD.h"
#include "Common/TimeUtil.h"
#include "Core/Config.h"
#include "Core/ConfigValues.h"
#include "Core/HW/StereoResampler.h"
#include "Core/Util/AudioFormat.h"  // for clamp_u16
#include "Core/System.h"

StereoResampler::StereoResampler() noexcept
		: maxBufsize_(MAX_BUFSIZE_DEFAULT)
	  , targetBufsize_(TARGET_BUFSIZE_DEFAULT) {
	// Need to have space for the worst case in case it changes.
	buffer_ = new int16_t[MAX_BUFSIZE_EXTRA * 2]();

	// Some Android devices are v-synced to non-60Hz framerates. We simply timestretch audio to fit.
	// TODO: should only do this if auto frameskip is off?
	float refresh = System_GetPropertyFloat(SYSPROP_DISPLAY_REFRESH_RATE);

	// If framerate is "close"...
	if (refresh != 60.0f && refresh > 50.0f && refresh < 70.0f) {
		int input_sample_rate = (int)(44100 * (refresh / 60.0f));
		INFO_LOG(Log::Audio, "StereoResampler: Adjusting target sample rate to %dHz", input_sample_rate);
		inputSampleRateHz_ = input_sample_rate;
	}

	UpdateBufferSize();
}

StereoResampler::~StereoResampler() {
	delete[] buffer_;
	buffer_ = nullptr;
}

void StereoResampler::UpdateBufferSize() {
	if (g_Config.bExtraAudioBuffering) {
		maxBufsize_ = MAX_BUFSIZE_EXTRA;
		targetBufsize_ = TARGET_BUFSIZE_EXTRA;
	} else {
		maxBufsize_ = MAX_BUFSIZE_DEFAULT;
		targetBufsize_ = TARGET_BUFSIZE_DEFAULT;

		int systemBufsize = System_GetPropertyInt(SYSPROP_AUDIO_FRAMES_PER_BUFFER);
		if (systemBufsize > 0 && targetBufsize_ < systemBufsize + TARGET_BUFSIZE_MARGIN) {
			targetBufsize_ = std::min(4096, systemBufsize + TARGET_BUFSIZE_MARGIN);
			if (targetBufsize_ * 2 > MAX_BUFSIZE_DEFAULT)
				maxBufsize_ = MAX_BUFSIZE_EXTRA;
		}
	}
}

// factor is a 0.12-bit fixed point number.
template<bool multiply>
inline void ClampBufferToS16(s16 *out, const s32 *in, size_t size, int factor) {
	if (multiply) {
		// Let's SIMD later. Unfortunately for s16 operations, SSE2 is very different and odd
		// so CrossSIMD won't be very useful.
		// LLVM autovec does an okay job with this on ARM64, it turns out.
		for (size_t i = 0; i < size; i++) {
			out[i] = clamp_s16((in[i] * factor) >> 12);
		}
	} else {
#ifdef _M_SSE
		// Size will always be 16-byte aligned as the hwBlockSize is.
		while (size >= 8) {
			__m128i in1 = _mm_loadu_si128((__m128i *)in);
			__m128i in2 = _mm_loadu_si128((__m128i *)(in + 4));
			__m128i packed = _mm_packs_epi32(in1, in2);  // pack with signed saturation, perfect.
			_mm_storeu_si128((__m128i *)out, packed);
			out += 8;
			in += 8;
			size -= 8;
		}
#elif PPSSPP_ARCH(ARM_NEON)
		// Dynamic shifts can only be left, but it's signed - negate to shift right.
		while (size >= 8) {
			int32x4_t in1 = vld1q_s32(in);
			int32x4_t in2 = vld1q_s32(in + 4);
			int16x4_t packed1 = vqmovn_s32(in1);
			int16x4_t packed2 = vqmovn_s32(in2);
			vst1_s16(out, packed1);
			vst1_s16(out + 4, packed2);
			out += 8;
			in += 8;
			size -= 8;
		}
#endif
		// This does the remainder if SIMD was used, otherwise it does it all.
		for (size_t i = 0; i < size; i++) {
			out[i] = clamp_s16(in[i]);
		}
	}
}

inline void ClampBufferToS16WithVolume(s16 *out, const s32 *in, size_t size, int volume) {
	// The last parameter to ClampBufferToS16 is no longer a shift, now it's a 12-bit multiplier.
	if (volume >= 4096) {
		ClampBufferToS16<false>(out, in, size, 0);
	} else if (volume <= 0) {
		memset(out, 0, size * sizeof(s16));
	} else {
		ClampBufferToS16<true>(out, in, size, volume);
	}
}

void StereoResampler::Clear() {
	memset(buffer_, 0, maxBufsize_ * 2 * sizeof(int16_t));
}

inline int16_t MixSingleSample(int16_t s1, int16_t s2, uint16_t frac) {
	int32_t value = s1 + (((s2 - s1) * frac) >> 16);
	if (value < -32767)
		return -32767;
	else if (value > 32767)
		return 32767;
	else
		return (int16_t)value;
}

// Executed from sound stream thread, pulling sound out of the buffer.
void StereoResampler::Mix(s16 *samples, unsigned int numSamples, bool consider_framelimit, int sample_rate) {
	if (!samples)
		return;

	unsigned int currentSample;

	// 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 (until it wraps...).
	// Without this cache, the compiler wouldn't be allowed to optimize the
	// interpolation loop.
	u32 indexR = indexR_.load();
	u32 indexW = indexW_.load();

	const int INDEX_MASK = (maxBufsize_ * 2 - 1);

	// This is only for debug visualization, not used for anything.
	lastBufSize_ = ((indexW - indexR) & INDEX_MASK) / 2;

	// Drift prevention mechanism.
	float numLeft = (float)(((indexW - indexR) & INDEX_MASK) / 2);
	// If we had to discard samples the last frame due to underrun,
	// apply an adjustment here. Otherwise we'll overestimate how many
	// samples we need.
	numLeft -= droppedSamples_;
	droppedSamples_ = 0;

	// numLeftI_ here becomes a lowpass filtered version of numLeft.
	numLeftI_ = (numLeft + numLeftI_ * (CONTROL_AVG - 1.0f)) / CONTROL_AVG;

	// Here we try to keep the buffer size around m_lowwatermark (which is
	// really now more like desired_buffer_size) by adjusting the speed.
	// Note that the speed of adjustment here does not take the buffer size into
	// account. Since this is called once per "output frame", the frame size
	// will affect how fast this algorithm reacts, which can't be a good thing.
	float offset = (numLeftI_ - (float)targetBufsize_) * CONTROL_FACTOR;
	if (offset > MAX_FREQ_SHIFT) offset = MAX_FREQ_SHIFT;
	if (offset < -MAX_FREQ_SHIFT) offset = -MAX_FREQ_SHIFT;

	outputSampleRateHz_ = (float)(inputSampleRateHz_ + offset);
	const u32 ratio = (u32)(65536.0 * outputSampleRateHz_ / (double)sample_rate);
	ratio_ = ratio;
	// TODO: consider a higher-quality resampling algorithm.
	// TODO: Add a fast path for 1:1.
	u32 frac = frac_;
	for (currentSample = 0; currentSample < numSamples * 2; currentSample += 2) {
		if (((indexW - indexR) & INDEX_MASK) <= 2) {
			// Ran out!
			// int missing = numSamples * 2 - currentSample;
			// ILOG("Resampler underrun: %d (numSamples: %d, currentSample: %d)", missing, numSamples, currentSample / 2);
			underrunCount_++;
			break;
		}
		u32 indexR2 = indexR + 2; //next sample
		s16 l1 = buffer_[indexR & INDEX_MASK]; //current
		s16 r1 = buffer_[(indexR + 1) & INDEX_MASK]; //current
		s16 l2 = buffer_[indexR2 & INDEX_MASK]; //next
		s16 r2 = buffer_[(indexR2 + 1) & INDEX_MASK]; //next
		samples[currentSample] = MixSingleSample(l1, l2, (u16)frac);
		samples[currentSample + 1] = MixSingleSample(r1, r2, (u16)frac);
		frac += ratio;
		indexR += 2 * (frac >> 16);
		frac &= 0xffff;
	}
	frac_ = frac;

	// Let's not count the underrun padding here.
	outputSampleCount_ += currentSample / 2;

	// Padding with the last value to reduce clicking
	short s[2];
	s[0] = clamp_s16(buffer_[(indexR - 1) & INDEX_MASK]);
	s[1] = clamp_s16(buffer_[(indexR - 2) & INDEX_MASK]);
	for (; currentSample < numSamples * 2; currentSample += 2) {
		samples[currentSample] = s[0];
		samples[currentSample + 1] = s[1];
	}

	// Flush cached variable
	indexR_.store(indexR);
}

// Executes on the emulator thread, pushing sound into the buffer.
void StereoResampler::PushSamples(const s32 *samples, unsigned int numSamples, float multiplier) {
	inputSampleCount_ += numSamples;

	UpdateBufferSize();
	const int INDEX_MASK = (maxBufsize_ * 2 - 1);
	// 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 = indexW_.load();

	u32 cap = maxBufsize_ * 2;
	// If fast-forwarding, no need to fill up the entire buffer, just screws up timing after releasing the fast-forward button.
	if (PSP_CoreParameter().fastForward) {
		cap = targetBufsize_ * 2;
	}

	// Check if we have enough free space
	// indexW == indexR_ results in empty buffer, so indexR must always be smaller than indexW
	if (numSamples * 2 + ((indexW - indexR_.load()) & INDEX_MASK) >= cap) {
		if (!PSP_CoreParameter().fastForward) {
			overrunCount_++;
		}
		// TODO: "Timestretch" by doing a windowed overlap with existing buffer content?
		return;
	}

	// 12-bit volume.
	int volume = (int)(multiplier * 4096.0f);

	// Check if we need to roll over to the start of the buffer during the copy.
	unsigned int indexW_left_samples = maxBufsize_ * 2 - (indexW & INDEX_MASK);
	if (numSamples * 2 > indexW_left_samples) {
		ClampBufferToS16WithVolume(&buffer_[indexW & INDEX_MASK], samples, indexW_left_samples, volume);
		ClampBufferToS16WithVolume(&buffer_[0], samples + indexW_left_samples, numSamples * 2 - indexW_left_samples, volume);
	} else {
		ClampBufferToS16WithVolume(&buffer_[indexW & INDEX_MASK], samples, numSamples * 2, volume);
	}

	indexW_ += numSamples * 2;
	lastPushSize_ = numSamples;
}

void StereoResampler::GetAudioDebugStats(char *buf, size_t bufSize) {
	double elapsed = time_now_d() - startTime_;

	double effective_input_sample_rate = (double)inputSampleCount_ / elapsed;
	double effective_output_sample_rate = (double)outputSampleCount_ / elapsed;

	double bufferLatencyMs = 1000.0 * (double)lastBufSize_ / (double)inputSampleRateHz_;
	snprintf(buf, bufSize,
		"Audio buffer: %d/%d (%0.1fms, target: %d)\n"
		"Filtered: %0.2f\n"
		"Underruns: %d\n"
		"Overruns: %d\n"
		"Sample rate: %d (input: %d)\n"
		"Effective input sample rate: %0.2f\n"
		"Effective output sample rate: %0.2f\n"
		"Push size: %d\n"
		"Ratio: %0.6f\n",
		lastBufSize_,
		maxBufsize_,
		bufferLatencyMs,
		targetBufsize_,
		numLeftI_,
		underrunCountTotal_,
		overrunCountTotal_,
		(int)outputSampleRateHz_,
		inputSampleRateHz_,
		effective_input_sample_rate,
		effective_output_sample_rate,
		lastPushSize_,
		(float)ratio_ / 65536.0f);
	underrunCountTotal_ += underrunCount_;
	overrunCountTotal_ += overrunCount_;
	underrunCount_ = 0;
	overrunCount_ = 0;

	// Use this to remove the bias from the startup.
	// if (elapsed > 3.0) {
		//ResetStatCounters();
	// }
}

void StereoResampler::ResetStatCounters() {
	underrunCount_ = 0;
	overrunCount_ = 0;
	underrunCountTotal_ = 0;
	overrunCountTotal_ = 0;
	inputSampleCount_ = 0;
	outputSampleCount_ = 0;
	startTime_ = time_now_d();
}