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	The main problem was that the volume of the mixer wasn't savestated. The volume is typically 0 at the beginning of a game, so loading a savestate at the beginning of a game would lead to silent DTK audio. I also added savestating to StreamADPCM.cpp.
		
			
				
	
	
		
			373 lines
		
	
	
	
		
			11 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			373 lines
		
	
	
	
		
			11 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
| // Copyright 2008 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/Mixer.h"
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| 
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| #include <cmath>
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| #include <cstring>
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| 
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| #include "AudioCommon/DPL2Decoder.h"
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| #include "Common/ChunkFile.h"
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| #include "Common/CommonTypes.h"
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| #include "Common/Logging/Log.h"
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| #include "Common/MathUtil.h"
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| #include "Common/Swap.h"
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| #include "Core/ConfigManager.h"
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| 
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| Mixer::Mixer(unsigned int BackendSampleRate)
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|     : m_sampleRate(BackendSampleRate), m_stretcher(BackendSampleRate)
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| {
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|   INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized");
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|   DPL2Reset();
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| }
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| 
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| Mixer::~Mixer()
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| {
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| }
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| 
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| void Mixer::DoState(PointerWrap& p)
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| {
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|   m_dma_mixer.DoState(p);
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|   m_streaming_mixer.DoState(p);
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|   m_wiimote_speaker_mixer.DoState(p);
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| }
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| 
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| // Executed from sound stream thread
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| unsigned int Mixer::MixerFifo::Mix(short* samples, unsigned int numSamples,
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|                                    bool consider_framelimit)
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| {
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|   unsigned int currentSample = 0;
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| 
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|   // Cache access in non-volatile variable
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|   // This is the only function changing the read value, so it's safe to
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|   // cache it locally although it's written here.
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|   // The writing pointer will be modified outside, but it will only increase,
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|   // so we will just ignore new written data while interpolating.
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|   // Without this cache, the compiler wouldn't be allowed to optimize the
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|   // interpolation loop.
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|   u32 indexR = m_indexR.load();
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|   u32 indexW = m_indexW.load();
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| 
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|   // render numleft sample pairs to samples[]
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|   // advance indexR with sample position
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|   // remember fractional offset
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| 
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|   float emulationspeed = SConfig::GetInstance().m_EmulationSpeed;
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|   float aid_sample_rate = static_cast<float>(m_input_sample_rate);
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|   if (consider_framelimit && emulationspeed > 0.0f)
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|   {
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|     float numLeft = static_cast<float>(((indexW - indexR) & INDEX_MASK) / 2);
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| 
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|     u32 low_waterwark = m_input_sample_rate * SConfig::GetInstance().iTimingVariance / 1000;
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|     low_waterwark = std::min(low_waterwark, MAX_SAMPLES / 2);
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| 
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|     m_numLeftI = (numLeft + m_numLeftI * (CONTROL_AVG - 1)) / CONTROL_AVG;
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|     float offset = (m_numLeftI - low_waterwark) * CONTROL_FACTOR;
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|     if (offset > MAX_FREQ_SHIFT)
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|       offset = MAX_FREQ_SHIFT;
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|     if (offset < -MAX_FREQ_SHIFT)
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|       offset = -MAX_FREQ_SHIFT;
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| 
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|     aid_sample_rate = (aid_sample_rate + offset) * emulationspeed;
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|   }
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| 
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|   const u32 ratio = (u32)(65536.0f * aid_sample_rate / (float)m_mixer->m_sampleRate);
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| 
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|   s32 lvolume = m_LVolume.load();
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|   s32 rvolume = m_RVolume.load();
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| 
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|   // TODO: consider a higher-quality resampling algorithm.
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|   for (; currentSample < numSamples * 2 && ((indexW - indexR) & INDEX_MASK) > 2; currentSample += 2)
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|   {
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|     u32 indexR2 = indexR + 2;  // next sample
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| 
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|     s16 l1 = Common::swap16(m_buffer[indexR & INDEX_MASK]);   // current
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|     s16 l2 = Common::swap16(m_buffer[indexR2 & INDEX_MASK]);  // next
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|     int sampleL = ((l1 << 16) + (l2 - l1) * (u16)m_frac) >> 16;
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|     sampleL = (sampleL * lvolume) >> 8;
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|     sampleL += samples[currentSample + 1];
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|     samples[currentSample + 1] = MathUtil::Clamp(sampleL, -32767, 32767);
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| 
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|     s16 r1 = Common::swap16(m_buffer[(indexR + 1) & INDEX_MASK]);   // current
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|     s16 r2 = Common::swap16(m_buffer[(indexR2 + 1) & INDEX_MASK]);  // next
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|     int sampleR = ((r1 << 16) + (r2 - r1) * (u16)m_frac) >> 16;
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|     sampleR = (sampleR * rvolume) >> 8;
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|     sampleR += samples[currentSample];
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|     samples[currentSample] = MathUtil::Clamp(sampleR, -32767, 32767);
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| 
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|     m_frac += ratio;
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|     indexR += 2 * (u16)(m_frac >> 16);
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|     m_frac &= 0xffff;
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|   }
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| 
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|   // Actual number of samples written to the buffer without padding.
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|   unsigned int actual_sample_count = currentSample / 2;
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| 
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|   // Padding
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|   short s[2];
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|   s[0] = Common::swap16(m_buffer[(indexR - 1) & INDEX_MASK]);
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|   s[1] = Common::swap16(m_buffer[(indexR - 2) & INDEX_MASK]);
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|   s[0] = (s[0] * rvolume) >> 8;
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|   s[1] = (s[1] * lvolume) >> 8;
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|   for (; currentSample < numSamples * 2; currentSample += 2)
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|   {
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|     int sampleR = MathUtil::Clamp(s[0] + samples[currentSample + 0], -32767, 32767);
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|     int sampleL = MathUtil::Clamp(s[1] + samples[currentSample + 1], -32767, 32767);
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| 
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|     samples[currentSample + 0] = sampleR;
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|     samples[currentSample + 1] = sampleL;
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|   }
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| 
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|   // Flush cached variable
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|   m_indexR.store(indexR);
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| 
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|   return actual_sample_count;
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| }
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| 
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| unsigned int Mixer::Mix(short* samples, unsigned int num_samples)
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| {
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|   if (!samples)
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|     return 0;
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| 
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|   memset(samples, 0, num_samples * 2 * sizeof(short));
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| 
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|   if (SConfig::GetInstance().m_audio_stretch)
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|   {
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|     unsigned int available_samples =
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|         std::min(m_dma_mixer.AvailableSamples(), m_streaming_mixer.AvailableSamples());
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| 
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|     m_scratch_buffer.fill(0);
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| 
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|     m_dma_mixer.Mix(m_scratch_buffer.data(), available_samples, false);
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|     m_streaming_mixer.Mix(m_scratch_buffer.data(), available_samples, false);
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|     m_wiimote_speaker_mixer.Mix(m_scratch_buffer.data(), available_samples, false);
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| 
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|     if (!m_is_stretching)
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|     {
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|       m_stretcher.Clear();
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|       m_is_stretching = true;
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|     }
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|     m_stretcher.ProcessSamples(m_scratch_buffer.data(), available_samples, num_samples);
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|     m_stretcher.GetStretchedSamples(samples, num_samples);
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|   }
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|   else
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|   {
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|     m_dma_mixer.Mix(samples, num_samples, true);
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|     m_streaming_mixer.Mix(samples, num_samples, true);
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|     m_wiimote_speaker_mixer.Mix(samples, num_samples, true);
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|     m_is_stretching = false;
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|   }
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| 
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|   return num_samples;
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| }
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| 
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| unsigned int Mixer::MixSurround(float* samples, unsigned int num_samples)
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| {
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|   if (!num_samples)
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|     return 0;
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| 
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|   memset(samples, 0, num_samples * 6 * sizeof(float));
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| 
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|   // Mix() may also use m_scratch_buffer internally, but is safe because it alternates reads and
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|   // writes.
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|   unsigned int available_samples = Mix(m_scratch_buffer.data(), num_samples);
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|   for (size_t i = 0; i < static_cast<size_t>(available_samples) * 2; ++i)
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|   {
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|     m_float_conversion_buffer[i] =
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|         m_scratch_buffer[i] / static_cast<float>(std::numeric_limits<short>::max());
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|   }
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| 
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|   DPL2Decode(m_float_conversion_buffer.data(), available_samples, samples);
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| 
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|   return available_samples;
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| }
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| 
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| void Mixer::MixerFifo::PushSamples(const short* samples, unsigned int num_samples)
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| {
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|   // Cache access in non-volatile variable
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|   // indexR isn't allowed to cache in the audio throttling loop as it
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|   // needs to get updates to not deadlock.
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|   u32 indexW = m_indexW.load();
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| 
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|   // Check if we have enough free space
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|   // indexW == m_indexR results in empty buffer, so indexR must always be smaller than indexW
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|   if (num_samples * 2 + ((indexW - m_indexR.load()) & INDEX_MASK) >= MAX_SAMPLES * 2)
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|     return;
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| 
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|   // AyuanX: Actual re-sampling work has been moved to sound thread
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|   // to alleviate the workload on main thread
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|   // and we simply store raw data here to make fast mem copy
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|   int over_bytes = num_samples * 4 - (MAX_SAMPLES * 2 - (indexW & INDEX_MASK)) * sizeof(short);
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|   if (over_bytes > 0)
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|   {
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|     memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4 - over_bytes);
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|     memcpy(&m_buffer[0], samples + (num_samples * 4 - over_bytes) / sizeof(short), over_bytes);
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|   }
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|   else
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|   {
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|     memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
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|   }
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| 
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|   m_indexW.fetch_add(num_samples * 2);
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| }
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| 
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| void Mixer::PushSamples(const short* samples, unsigned int num_samples)
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| {
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|   m_dma_mixer.PushSamples(samples, num_samples);
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|   int sample_rate = m_dma_mixer.GetInputSampleRate();
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|   if (m_log_dsp_audio)
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|     m_wave_writer_dsp.AddStereoSamplesBE(samples, num_samples, sample_rate);
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| }
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| 
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| void Mixer::PushStreamingSamples(const short* samples, unsigned int num_samples)
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| {
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|   m_streaming_mixer.PushSamples(samples, num_samples);
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|   int sample_rate = m_streaming_mixer.GetInputSampleRate();
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|   if (m_log_dtk_audio)
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|     m_wave_writer_dtk.AddStereoSamplesBE(samples, num_samples, sample_rate);
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| }
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| 
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| void Mixer::PushWiimoteSpeakerSamples(const short* samples, unsigned int num_samples,
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|                                       unsigned int sample_rate)
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| {
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|   short samples_stereo[MAX_SAMPLES * 2];
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| 
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|   if (num_samples < MAX_SAMPLES)
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|   {
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|     m_wiimote_speaker_mixer.SetInputSampleRate(sample_rate);
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| 
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|     for (unsigned int i = 0; i < num_samples; ++i)
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|     {
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|       samples_stereo[i * 2] = Common::swap16(samples[i]);
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|       samples_stereo[i * 2 + 1] = Common::swap16(samples[i]);
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|     }
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| 
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|     m_wiimote_speaker_mixer.PushSamples(samples_stereo, num_samples);
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|   }
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| }
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| 
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| void Mixer::SetDMAInputSampleRate(unsigned int rate)
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| {
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|   m_dma_mixer.SetInputSampleRate(rate);
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| }
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| 
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| void Mixer::SetStreamInputSampleRate(unsigned int rate)
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| {
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|   m_streaming_mixer.SetInputSampleRate(rate);
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| }
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| 
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| void Mixer::SetStreamingVolume(unsigned int lvolume, unsigned int rvolume)
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| {
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|   m_streaming_mixer.SetVolume(lvolume, rvolume);
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| }
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| 
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| void Mixer::SetWiimoteSpeakerVolume(unsigned int lvolume, unsigned int rvolume)
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| {
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|   m_wiimote_speaker_mixer.SetVolume(lvolume, rvolume);
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| }
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| 
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| void Mixer::StartLogDTKAudio(const std::string& filename)
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| {
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|   if (!m_log_dtk_audio)
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|   {
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|     bool success = m_wave_writer_dtk.Start(filename, m_streaming_mixer.GetInputSampleRate());
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|     if (success)
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|     {
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|       m_log_dtk_audio = true;
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|       m_wave_writer_dtk.SetSkipSilence(false);
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|       NOTICE_LOG(AUDIO, "Starting DTK Audio logging");
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|     }
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|     else
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|     {
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|       m_wave_writer_dtk.Stop();
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|       NOTICE_LOG(AUDIO, "Unable to start DTK Audio logging");
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|     }
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|   }
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|   else
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|   {
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|     WARN_LOG(AUDIO, "DTK Audio logging has already been started");
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|   }
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| }
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| 
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| void Mixer::StopLogDTKAudio()
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| {
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|   if (m_log_dtk_audio)
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|   {
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|     m_log_dtk_audio = false;
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|     m_wave_writer_dtk.Stop();
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|     NOTICE_LOG(AUDIO, "Stopping DTK Audio logging");
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|   }
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|   else
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|   {
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|     WARN_LOG(AUDIO, "DTK Audio logging has already been stopped");
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|   }
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| }
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| 
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| void Mixer::StartLogDSPAudio(const std::string& filename)
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| {
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|   if (!m_log_dsp_audio)
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|   {
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|     bool success = m_wave_writer_dsp.Start(filename, m_dma_mixer.GetInputSampleRate());
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|     if (success)
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|     {
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|       m_log_dsp_audio = true;
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|       m_wave_writer_dsp.SetSkipSilence(false);
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|       NOTICE_LOG(AUDIO, "Starting DSP Audio logging");
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|     }
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|     else
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|     {
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|       m_wave_writer_dsp.Stop();
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|       NOTICE_LOG(AUDIO, "Unable to start DSP Audio logging");
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|     }
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|   }
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|   else
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|   {
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|     WARN_LOG(AUDIO, "DSP Audio logging has already been started");
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|   }
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| }
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| 
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| void Mixer::StopLogDSPAudio()
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| {
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|   if (m_log_dsp_audio)
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|   {
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|     m_log_dsp_audio = false;
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|     m_wave_writer_dsp.Stop();
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|     NOTICE_LOG(AUDIO, "Stopping DSP Audio logging");
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|   }
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|   else
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|   {
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|     WARN_LOG(AUDIO, "DSP Audio logging has already been stopped");
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|   }
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| }
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| 
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| void Mixer::MixerFifo::DoState(PointerWrap& p)
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| {
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|   p.Do(m_input_sample_rate);
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|   p.Do(m_LVolume);
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|   p.Do(m_RVolume);
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| }
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| 
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| void Mixer::MixerFifo::SetInputSampleRate(unsigned int rate)
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| {
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|   m_input_sample_rate = rate;
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| }
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| 
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| unsigned int Mixer::MixerFifo::GetInputSampleRate() const
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| {
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|   return m_input_sample_rate;
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| }
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| 
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| void Mixer::MixerFifo::SetVolume(unsigned int lvolume, unsigned int rvolume)
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| {
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|   m_LVolume.store(lvolume + (lvolume >> 7));
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|   m_RVolume.store(rvolume + (rvolume >> 7));
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| }
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| 
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| unsigned int Mixer::MixerFifo::AvailableSamples() const
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| {
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|   unsigned int samples_in_fifo = ((m_indexW.load() - m_indexR.load()) & INDEX_MASK) / 2;
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|   if (samples_in_fifo <= 1)
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|     return 0;  // Mixer::MixerFifo::Mix always keeps one sample in the buffer.
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|   return (samples_in_fifo - 1) * m_mixer->m_sampleRate / m_input_sample_rate;
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| }
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