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https://github.com/hrydgard/ppsspp.git
synced 2024-11-23 21:39:52 +00:00
Just some formatting/cleanup
This commit is contained in:
Henrik Rydgard
parent
70f7443453
commit
ee37d51c4b
@ -29,8 +29,6 @@
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#include "FixedSizeQueue.h"
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#include "Common/Thread.h"
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// #define LOW_LATENCY_AUDIO
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// Should be used to lock anything related to the outAudioQueue.
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recursive_mutex section;
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@ -49,14 +47,9 @@ static int audioHostIntervalUs;
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static s32 *mixBuffer;
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// High and low watermarks, basically. For perfect emulation, the correct values are 0 and 1, respectively.
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// TODO: Tweak
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#ifdef LOW_LATENCY_AUDIO
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const int chanQueueMaxSizeFactor = 0;
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const int chanQueueMinSizeFactor = 1;
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#else
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const int chanQueueMaxSizeFactor = 2;
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const int chanQueueMinSizeFactor = 1;
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#endif
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// TODO: Tweak. Hm, there aren't actually even used currently...
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static int chanQueueMaxSizeFactor;
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static int chanQueueMinSizeFactor;
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// TODO: Need to replace this with something lockless. Mutexes in the audio pipeline
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// is bad mojo.
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@ -91,9 +84,13 @@ void __AudioInit() {
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mixFrequency = 44100;
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if (g_Config.bLowLatencyAudio) {
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chanQueueMaxSizeFactor = 1;
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chanQueueMinSizeFactor = 0;
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hwBlockSize = 16;
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hostAttemptBlockSize = 256;
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} else {
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chanQueueMaxSizeFactor = 2;
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chanQueueMinSizeFactor = 1;
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hwBlockSize = 64;
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hostAttemptBlockSize = 512;
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}
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@ -113,8 +110,7 @@ void __AudioInit() {
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memset(mixBuffer, 0, hwBlockSize * 2 * sizeof(s32));
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}
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void __AudioDoState(PointerWrap &p)
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{
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void __AudioDoState(PointerWrap &p) {
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p.Do(eventAudioUpdate);
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CoreTiming::RestoreRegisterEvent(eventAudioUpdate, "AudioUpdate", &hleAudioUpdate);
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p.Do(eventHostAudioUpdate);
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@ -181,17 +177,14 @@ u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking) {
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return ret;
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}
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if (chan.format == PSP_AUDIO_FORMAT_STEREO)
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{
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if (chan.format == PSP_AUDIO_FORMAT_STEREO) {
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const u32 totalSamples = chan.sampleCount * 2;
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if (IS_LITTLE_ENDIAN)
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{
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if (IS_LITTLE_ENDIAN) {
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s16 *sampleData = (s16 *) Memory::GetPointer(chan.sampleAddress);
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// Walking a pointer for speed. But let's make sure we wouldn't trip on an invalid ptr.
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if (Memory::IsValidAddress(chan.sampleAddress + (totalSamples - 1) * sizeof(s16)))
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{
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if (Memory::IsValidAddress(chan.sampleAddress + (totalSamples - 1) * sizeof(s16))) {
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#if 0
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for (u32 i = 0; i < totalSamples; i += 2) {
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chan.sampleQueue.push(adjustvolume(*sampleData++, chan.leftVolume));
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@ -205,24 +198,20 @@ u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking) {
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int rightVol = chan.rightVolume;
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// TODO: SSE/NEON implementations
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for (u32 i = 0; i < sz1; i += 2)
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{
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for (u32 i = 0; i < sz1; i += 2) {
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buf1[i] = adjustvolume(sampleData[i], leftVol);
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buf1[i + 1] = adjustvolume(sampleData[i + 1], rightVol);
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}
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if (buf2) {
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sampleData += sz1;
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for (u32 i = 0; i < sz2; i += 2)
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{
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for (u32 i = 0; i < sz2; i += 2) {
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buf2[i] = adjustvolume(sampleData[i], leftVol);
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buf2[i + 1] = adjustvolume(sampleData[i + 1], rightVol);
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}
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}
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#endif
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}
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}
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else
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{
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} else {
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for (u32 i = 0; i < totalSamples; i++) {
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s16 sampleL = (s16)Memory::Read_U16(chan.sampleAddress + sizeof(s16) * i);
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sampleL = adjustvolume(sampleL, chan.leftVolume);
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@ -234,10 +223,8 @@ u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking) {
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}
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}
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}
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else if (chan.format == PSP_AUDIO_FORMAT_MONO)
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{
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for (u32 i = 0; i < chan.sampleCount; i++)
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{
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else if (chan.format == PSP_AUDIO_FORMAT_MONO) {
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for (u32 i = 0; i < chan.sampleCount; i++) {
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// Expand to stereo
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s16 sample = (s16)Memory::Read_U16(chan.sampleAddress + 2 * i);
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chan.sampleQueue.push(adjustvolume(sample, chan.leftVolume));
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@ -247,18 +234,15 @@ u32 __AudioEnqueue(AudioChannel &chan, int chanNum, bool blocking) {
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return ret;
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}
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inline void __AudioWakeThreads(AudioChannel &chan, int result, int step)
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{
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inline void __AudioWakeThreads(AudioChannel &chan, int result, int step) {
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u32 error;
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for (size_t w = 0; w < chan.waitingThreads.size(); ++w)
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{
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for (size_t w = 0; w < chan.waitingThreads.size(); ++w) {
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AudioChannelWaitInfo &waitInfo = chan.waitingThreads[w];
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waitInfo.numSamples -= step;
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// If it's done (there will still be samples on queue) and actually still waiting, wake it up.
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u32 waitID = __KernelGetWaitID(waitInfo.threadID, WAITTYPE_AUDIOCHANNEL, error);
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if (waitInfo.numSamples <= 0 && waitID != 0)
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{
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if (waitInfo.numSamples <= 0 && waitID != 0) {
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// DEBUG_LOG(HLE, "Woke thread %i for some buffer filling", waitingThread);
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u32 ret = result == 0 ? __KernelGetWaitValue(waitInfo.threadID, error) : SCE_ERROR_AUDIO_CHANNEL_NOT_RESERVED;
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__KernelResumeThreadFromWait(waitInfo.threadID, ret);
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@ -271,13 +255,11 @@ inline void __AudioWakeThreads(AudioChannel &chan, int result, int step)
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}
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}
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void __AudioWakeThreads(AudioChannel &chan, int result)
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{
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void __AudioWakeThreads(AudioChannel &chan, int result) {
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__AudioWakeThreads(chan, result, 0x7FFFFFFF);
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}
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void __AudioSetOutputFrequency(int freq)
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{
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void __AudioSetOutputFrequency(int freq) {
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WARN_LOG(HLE, "Switching audio frequency to %i", freq);
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mixFrequency = freq;
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}
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@ -421,6 +421,8 @@ const HLEFunction sceAudio[] =
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{0x63f2889c, WrapU_U<sceAudioOutput2ChangeLength>, "sceAudioOutput2ChangeLength"},
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{0x647cef33, WrapU_V<sceAudioOutput2GetRestSample>, "sceAudioOutput2GetRestSample"},
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{0x43196845, WrapU_V<sceAudioOutput2Release>, "sceAudioOutput2Release"},
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// "Traditional" audio channel interface
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{0x80F1F7E0, WrapU_V<sceAudioInit>, "sceAudioInit"},
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{0x210567F7, WrapU_V<sceAudioEnd>, "sceAudioEnd"},
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{0xA2BEAA6C, WrapU_U<sceAudioSetFrequency>, "sceAudioSetFrequency"},
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@ -436,6 +438,8 @@ const HLEFunction sceAudio[] =
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{0xCB2E439E, WrapU_UU<sceAudioSetChannelDataLen>, "sceAudioSetChannelDataLen"},
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{0x95FD0C2D, WrapU_UU<sceAudioChangeChannelConfig>, "sceAudioChangeChannelConfig"},
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{0xB7E1D8E7, WrapU_UUU<sceAudioChangeChannelVolume>, "sceAudioChangeChannelVolume"},
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// Not sure about the point of these, maybe like traditional but with ability to do sample rate conversion?
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{0x38553111, WrapU_UUU<sceAudioSRCChReserve>, "sceAudioSRCChReserve"},
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{0x5C37C0AE, WrapU_V<sceAudioSRCChRelease>, "sceAudioSRCChRelease"},
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{0xE0727056, WrapU_UU<sceAudioSRCOutputBlocking>, "sceAudioSRCOutputBlocking"},
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