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https://github.com/dolphin-emu/dolphin.git
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ce1057f17d
This should eliminate the crackling with alsa and pulseaudio backends and replace it with much nicer pauses. This is only interesting for audio backends that do not respect Mixer::GetNumSamples() and should not impact users able to run the DSPLLE at full speed. The cost of this is an added LLE audio latency of about 0.06 s in the continuous playback case. If that is too much, lower the low watermark. git-svn-id: https://dolphin-emu.googlecode.com/svn/trunk@6239 8ced0084-cf51-0410-be5f-012b33b47a6e
91 lines
2.5 KiB
C++
91 lines
2.5 KiB
C++
// Copyright (C) 2003 Dolphin Project.
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// This program is free software: you can redistribute it and/or modify
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// it under the terms of the GNU General Public License as published by
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// the Free Software Foundation, version 2.0.
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// This program is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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// GNU General Public License 2.0 for more details.
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// A copy of the GPL 2.0 should have been included with the program.
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// If not, see http://www.gnu.org/licenses/
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// Official SVN repository and contact information can be found at
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// http://code.google.com/p/dolphin-emu/
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#ifndef _MIXER_H_
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#define _MIXER_H_
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// 16 bit Stereo
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#define MAX_SAMPLES (1024 * 8)
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#define INDEX_MASK (MAX_SAMPLES * 2 - 1)
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#define RESERVED_SAMPLES (256)
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class CMixer {
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public:
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CMixer(unsigned int AISampleRate = 48000, unsigned int DACSampleRate = 48000)
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: m_aiSampleRate(AISampleRate)
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, m_dacSampleRate(DACSampleRate)
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, m_bits(16)
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, m_channels(2)
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, m_HLEready(false)
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, m_numSamples(0)
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, m_indexW(0)
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, m_indexR(0)
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, m_LLEplaying(true)
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{
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// AyuanX: The internal (Core & DSP) sample rate is fixed at 32KHz
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// So when AI/DAC sample rate differs than 32KHz, we have to do re-sampling
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// I prefer speed so let's do down-sampling instead of up-sampling
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// If you like better sound than speed, feel free to implement the up-sampling code
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m_sampleRate = 32000;
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INFO_LOG(AUDIO_INTERFACE, "Mixer is initialized (AISampleRate:%i, DACSampleRate:%i)", AISampleRate, DACSampleRate);
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}
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virtual ~CMixer() {}
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// Called from audio threads
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virtual unsigned int Mix(short* samples, unsigned int numSamples);
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virtual void Premix(short *samples, unsigned int numSamples) {}
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unsigned int GetNumSamples();
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// Called from main thread
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virtual void PushSamples(short* samples, unsigned int num_samples);
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unsigned int GetSampleRate() {return m_sampleRate;}
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void SetThrottle(bool use) { m_throttle = use;}
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void SetDTKMusic(bool use) { m_EnableDTKMusic = use;}
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// TODO: do we need this
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bool IsHLEReady() { return m_HLEready;}
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void SetHLEReady(bool ready) { m_HLEready = ready;}
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// ---------------------
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protected:
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unsigned int m_sampleRate;
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unsigned int m_aiSampleRate;
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unsigned int m_dacSampleRate;
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int m_bits;
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int m_channels;
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bool m_HLEready;
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bool m_EnableDTKMusic;
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bool m_throttle;
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short m_buffer[MAX_SAMPLES * 2];
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volatile u32 m_numSamples;
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u32 m_indexW;
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u32 m_indexR;
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bool m_LLEplaying;
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private:
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};
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#endif // _MIXER_H_
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