344 lines
9.9 KiB
C++
344 lines
9.9 KiB
C++
// Copyright 2003 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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Here is a nice ascii overview of audio flow affected by this file:
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(RAM)---->[AI FIFO]---->[SRC]---->[Mixer]---->[DAC]---->(Speakers)
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^
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[L/R Volume]
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\
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(DVD)---->[Drive I/F]---->[SRC]---->[Counter]
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Notes:
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Output at "48KHz" is actually 48043Hz.
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Sample counter counts streaming stereo samples after upsampling.
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[DAC] causes [AI I/F] to read from RAM at rate selected by AIDFR.
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Each [SRC] will upsample a 32KHz source, or pass through the 48KHz
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source. The [Mixer]/[DAC] only operate at 48KHz.
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AIS == disc streaming == DTK(Disk Track Player) == streaming audio, etc.
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Supposedly, the retail hardware only supports 48KHz streaming from
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[Drive I/F]. However it's more likely that the hardware supports
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32KHz streaming, and the upsampling is transparent to the user.
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TODO check if anything tries to stream at 32KHz.
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The [Drive I/F] actually supports simultaneous requests for audio and
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normal data. For this reason, we can't really get rid of the crit section.
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IMPORTANT:
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This file mainly deals with the [Drive I/F], however [AIDFR] controls
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the rate at which the audio data is DMA'd from RAM into the [AI FIFO]
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(and the speed at which the FIFO is read by its SRC). Everything else
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relating to AID happens in DSP.cpp. It's kinda just bad hardware design.
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TODO maybe the files should be merged?
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*/
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#include "Core/HW/AudioInterface.h"
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#include <algorithm>
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#include "AudioCommon/AudioCommon.h"
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#include "Common/ChunkFile.h"
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#include "Common/CommonTypes.h"
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#include "Core/CoreTiming.h"
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#include "Core/HW/MMIO.h"
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#include "Core/HW/ProcessorInterface.h"
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#include "Core/HW/SystemTimers.h"
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#include "Core/PowerPC/PowerPC.h"
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namespace AudioInterface
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{
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// Internal hardware addresses
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enum
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{
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AI_CONTROL_REGISTER = 0x6C00,
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AI_VOLUME_REGISTER = 0x6C04,
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AI_SAMPLE_COUNTER = 0x6C08,
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AI_INTERRUPT_TIMING = 0x6C0C,
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};
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enum
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{
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AIS_32KHz = 0,
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AIS_48KHz = 1,
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AID_32KHz = 1,
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AID_48KHz = 0
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};
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// AI Control Register
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union AICR
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{
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AICR() = default;
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explicit AICR(u32 hex_) : hex{hex_} {}
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struct
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{
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u32 PSTAT : 1; // sample counter/playback enable
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u32 AISFR : 1; // AIS Frequency (0=32khz 1=48khz)
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u32 AIINTMSK : 1; // 0=interrupt masked 1=interrupt enabled
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u32 AIINT : 1; // audio interrupt status
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u32 AIINTVLD : 1; // This bit controls whether AIINT is affected by the Interrupt Timing
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// register
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// matching the sample counter. Once set, AIINT will hold its last value
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u32 SCRESET : 1; // write to reset counter
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u32 AIDFR : 1; // AID Frequency (0=48khz 1=32khz)
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u32 : 25;
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};
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u32 hex = 0;
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};
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// AI Volume Register
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union AIVR
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{
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struct
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{
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u32 left : 8;
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u32 right : 8;
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u32 : 16;
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};
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u32 hex = 0;
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};
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// STATE_TO_SAVE
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// Registers
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static AICR s_control;
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static AIVR s_volume;
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static u32 s_sample_counter = 0;
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static u32 s_interrupt_timing = 0;
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static u64 s_last_cpu_time = 0;
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static u64 s_cpu_cycles_per_sample = 0xFFFFFFFFFFFULL;
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static u32 s_ais_sample_rate = 48000;
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static u32 s_aid_sample_rate = 32000;
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void DoState(PointerWrap& p)
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{
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p.DoPOD(s_control);
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p.DoPOD(s_volume);
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p.Do(s_sample_counter);
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p.Do(s_interrupt_timing);
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p.Do(s_last_cpu_time);
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p.Do(s_ais_sample_rate);
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p.Do(s_aid_sample_rate);
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p.Do(s_cpu_cycles_per_sample);
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g_sound_stream->GetMixer()->DoState(p);
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}
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static void GenerateAudioInterrupt();
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static void UpdateInterrupts();
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static void IncreaseSampleCount(u32 amount);
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static int GetAIPeriod();
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static void Update(u64 userdata, s64 cycles_late);
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static CoreTiming::EventType* event_type_ai;
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void Init()
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{
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s_control.hex = 0;
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s_control.AISFR = AIS_48KHz;
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s_volume.hex = 0;
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s_sample_counter = 0;
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s_interrupt_timing = 0;
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s_last_cpu_time = 0;
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s_cpu_cycles_per_sample = 0xFFFFFFFFFFFULL;
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s_ais_sample_rate = Get48KHzSampleRate();
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s_aid_sample_rate = Get32KHzSampleRate();
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event_type_ai = CoreTiming::RegisterEvent("AICallback", Update);
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}
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void Shutdown()
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{
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}
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void RegisterMMIO(MMIO::Mapping* mmio, u32 base)
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{
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mmio->Register(
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base | AI_CONTROL_REGISTER, MMIO::DirectRead<u32>(&s_control.hex),
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MMIO::ComplexWrite<u32>([](u32, u32 val) {
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const AICR tmp_ai_ctrl(val);
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if (s_control.AIINTMSK != tmp_ai_ctrl.AIINTMSK)
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{
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "Change AIINTMSK to {}", tmp_ai_ctrl.AIINTMSK);
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s_control.AIINTMSK = tmp_ai_ctrl.AIINTMSK;
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}
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if (s_control.AIINTVLD != tmp_ai_ctrl.AIINTVLD)
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{
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "Change AIINTVLD to {}", tmp_ai_ctrl.AIINTVLD);
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s_control.AIINTVLD = tmp_ai_ctrl.AIINTVLD;
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}
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// Set frequency of streaming audio
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if (tmp_ai_ctrl.AISFR != s_control.AISFR)
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{
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// AISFR rates below are intentionally inverted wrt yagcd
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "Change AISFR to {}",
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tmp_ai_ctrl.AISFR ? "48khz" : "32khz");
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s_control.AISFR = tmp_ai_ctrl.AISFR;
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s_ais_sample_rate = tmp_ai_ctrl.AISFR ? Get48KHzSampleRate() : Get32KHzSampleRate();
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g_sound_stream->GetMixer()->SetStreamInputSampleRate(s_ais_sample_rate);
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s_cpu_cycles_per_sample = SystemTimers::GetTicksPerSecond() / s_ais_sample_rate;
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}
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// Set frequency of DMA
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if (tmp_ai_ctrl.AIDFR != s_control.AIDFR)
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{
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "Change AIDFR to {}",
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tmp_ai_ctrl.AIDFR ? "32khz" : "48khz");
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s_control.AIDFR = tmp_ai_ctrl.AIDFR;
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s_aid_sample_rate = tmp_ai_ctrl.AIDFR ? Get32KHzSampleRate() : Get48KHzSampleRate();
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g_sound_stream->GetMixer()->SetDMAInputSampleRate(s_aid_sample_rate);
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}
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// Streaming counter
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if (tmp_ai_ctrl.PSTAT != s_control.PSTAT)
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{
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "{} streaming audio",
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tmp_ai_ctrl.PSTAT ? "start" : "stop");
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s_control.PSTAT = tmp_ai_ctrl.PSTAT;
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s_last_cpu_time = CoreTiming::GetTicks();
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CoreTiming::RemoveEvent(event_type_ai);
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CoreTiming::ScheduleEvent(GetAIPeriod(), event_type_ai);
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}
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// AI Interrupt
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if (tmp_ai_ctrl.AIINT)
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{
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "Clear AIS Interrupt");
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s_control.AIINT = 0;
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}
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// Sample Count Reset
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if (tmp_ai_ctrl.SCRESET)
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{
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "Reset AIS sample counter");
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s_sample_counter = 0;
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s_last_cpu_time = CoreTiming::GetTicks();
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}
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UpdateInterrupts();
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}));
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mmio->Register(base | AI_VOLUME_REGISTER, MMIO::DirectRead<u32>(&s_volume.hex),
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MMIO::ComplexWrite<u32>([](u32, u32 val) {
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s_volume.hex = val;
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g_sound_stream->GetMixer()->SetStreamingVolume(s_volume.left, s_volume.right);
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}));
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mmio->Register(base | AI_SAMPLE_COUNTER, MMIO::ComplexRead<u32>([](u32) {
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return s_sample_counter +
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static_cast<u32>((CoreTiming::GetTicks() - s_last_cpu_time) /
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s_cpu_cycles_per_sample);
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}),
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MMIO::ComplexWrite<u32>([](u32, u32 val) {
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s_sample_counter = val;
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s_last_cpu_time = CoreTiming::GetTicks();
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CoreTiming::RemoveEvent(event_type_ai);
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CoreTiming::ScheduleEvent(GetAIPeriod(), event_type_ai);
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}));
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mmio->Register(base | AI_INTERRUPT_TIMING, MMIO::DirectRead<u32>(&s_interrupt_timing),
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MMIO::ComplexWrite<u32>([](u32, u32 val) {
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DEBUG_LOG_FMT(AUDIO_INTERFACE, "AI_INTERRUPT_TIMING={:08x} at PC: {:08x}", val,
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PowerPC::ppcState.pc);
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s_interrupt_timing = val;
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CoreTiming::RemoveEvent(event_type_ai);
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CoreTiming::ScheduleEvent(GetAIPeriod(), event_type_ai);
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}));
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}
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static void UpdateInterrupts()
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{
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ProcessorInterface::SetInterrupt(ProcessorInterface::INT_CAUSE_AI,
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s_control.AIINT & s_control.AIINTMSK);
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}
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static void GenerateAudioInterrupt()
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{
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s_control.AIINT = 1;
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UpdateInterrupts();
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}
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void GenerateAISInterrupt()
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{
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GenerateAudioInterrupt();
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}
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static void IncreaseSampleCount(const u32 amount)
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{
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if (s_control.PSTAT)
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{
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const u32 old_sample_counter = s_sample_counter + 1;
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s_sample_counter += amount;
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if ((s_interrupt_timing - old_sample_counter) <= (s_sample_counter - old_sample_counter))
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{
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DEBUG_LOG(AUDIO_INTERFACE, "GenerateAudioInterrupt %08x:%08x @ %08x s_control.AIINTVLD=%d",
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s_sample_counter, s_interrupt_timing, PowerPC::ppcState.pc, s_control.AIINTVLD);
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GenerateAudioInterrupt();
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}
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}
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}
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bool IsPlaying()
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{
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return (s_control.PSTAT == 1);
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}
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u32 GetAIDSampleRate()
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{
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return s_aid_sample_rate;
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}
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u32 GetAISSampleRate()
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{
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return s_ais_sample_rate;
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}
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u32 Get32KHzSampleRate()
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{
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return SConfig::GetInstance().bWii ? 32000 : 32029;
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}
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u32 Get48KHzSampleRate()
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{
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return SConfig::GetInstance().bWii ? 48000 : 48043;
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}
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static void Update(u64 userdata, s64 cycles_late)
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{
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if (s_control.PSTAT)
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{
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const u64 diff = CoreTiming::GetTicks() - s_last_cpu_time;
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if (diff > s_cpu_cycles_per_sample)
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{
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const u32 samples = static_cast<u32>(diff / s_cpu_cycles_per_sample);
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s_last_cpu_time += samples * s_cpu_cycles_per_sample;
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IncreaseSampleCount(samples);
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}
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CoreTiming::ScheduleEvent(GetAIPeriod() - cycles_late, event_type_ai);
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}
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}
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int GetAIPeriod()
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{
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u64 period = s_cpu_cycles_per_sample * (s_interrupt_timing - s_sample_counter);
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u64 s_period = s_cpu_cycles_per_sample * s_ais_sample_rate;
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if (period == 0)
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return static_cast<int>(s_period);
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return static_cast<int>(std::min(period, s_period));
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}
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} // end of namespace AudioInterface
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