mirror of
https://github.com/izzy2lost/dolphin.git
synced 2026-06-19 01:16:48 -07:00
Merge branch 'master' into android-new-control-input-overlay
Conflicts: Source/Core/DolphinWX/Src/Android/ButtonManager.h
This commit is contained in:
@@ -119,7 +119,7 @@
|
||||
# libraries. Automatically turned for NDK r5x and r6x due to GLESv2
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# files. See additional details below.
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#
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@@ -187,7 +187,7 @@
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<Filter>SoundStreams</Filter>
|
||||
</ClInclude>
|
||||
<ClInclude Include="Src\AlsaSoundStream.h">
|
||||
<Filter>SoundStreams</Filter>
|
||||
</ClInclude>
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<None Include="CMakeLists.txt" />
|
||||
</ItemGroup>
|
||||
<ItemGroup>
|
||||
<Filter Include="SoundStreams">
|
||||
<UniqueIdentifier>{efb9f5b5-ab0c-455d-b78b-26df725386af}</UniqueIdentifier>
|
||||
</Filter>
|
||||
<Text Include="CMakeLists.txt" />
|
||||
</ItemGroup>
|
||||
</Project>
|
||||
@@ -42,8 +42,8 @@ public:
|
||||
return true;
|
||||
}
|
||||
|
||||
virtual bool usesMixer() const {
|
||||
return true;
|
||||
virtual bool usesMixer() const {
|
||||
return true;
|
||||
}
|
||||
|
||||
virtual void Update();
|
||||
|
||||
@@ -57,7 +57,7 @@ void AlsaSound::SoundLoop()
|
||||
// Underrun
|
||||
snd_pcm_prepare(handle);
|
||||
}
|
||||
else if (rc < 0)
|
||||
else if (rc < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "writei fail: %s", snd_strerror(rc));
|
||||
}
|
||||
@@ -77,7 +77,7 @@ bool AlsaSound::AlsaInit()
|
||||
unsigned int periods;
|
||||
|
||||
err = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Audio open error: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
@@ -86,21 +86,21 @@ bool AlsaSound::AlsaInit()
|
||||
snd_pcm_hw_params_alloca(&hwparams);
|
||||
|
||||
err = snd_pcm_hw_params_any(handle, hwparams);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Broken configuration for this PCM: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
|
||||
err = snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Access type not available: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_hw_params_set_format(handle, hwparams, SND_PCM_FORMAT_S16_LE);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Sample format not available: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
@@ -108,14 +108,14 @@ bool AlsaSound::AlsaInit()
|
||||
|
||||
dir = 0;
|
||||
err = snd_pcm_hw_params_set_rate_near(handle, hwparams, &sample_rate, &dir);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Rate not available: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_hw_params_set_channels(handle, hwparams, 2);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Channels count not available: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
@@ -123,7 +123,7 @@ bool AlsaSound::AlsaInit()
|
||||
|
||||
periods = BUFFER_SIZE_MAX / FRAME_COUNT_MIN;
|
||||
err = snd_pcm_hw_params_set_periods_max(handle, hwparams, &periods, &dir);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Cannot set Minimum periods: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
@@ -131,34 +131,34 @@ bool AlsaSound::AlsaInit()
|
||||
|
||||
buffer_size_max = BUFFER_SIZE_MAX;
|
||||
err = snd_pcm_hw_params_set_buffer_size_max(handle, hwparams, &buffer_size_max);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Cannot set minimum buffer size: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_hw_params(handle, hwparams);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Unable to install hw params: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_hw_params_get_buffer_size(hwparams, &buffer_size);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Cannot get buffer size: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_hw_params_get_periods_max(hwparams, &periods, &dir);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Cannot get periods: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
//periods is the number of fragments alsa can wait for during one
|
||||
//periods is the number of fragments alsa can wait for during one
|
||||
//buffer_size
|
||||
frames_to_deliver = buffer_size / periods;
|
||||
//limit the minimum size. pulseaudio advertises a minimum of 32 samples.
|
||||
@@ -172,28 +172,28 @@ bool AlsaSound::AlsaInit()
|
||||
snd_pcm_sw_params_alloca(&swparams);
|
||||
|
||||
err = snd_pcm_sw_params_current(handle, swparams);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "cannot init sw params: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_sw_params_set_start_threshold(handle, swparams, 0U);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "cannot set start thresh: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_sw_params(handle, swparams);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "cannot set sw params: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
}
|
||||
|
||||
err = snd_pcm_prepare(handle);
|
||||
if (err < 0)
|
||||
if (err < 0)
|
||||
{
|
||||
ERROR_LOG(AUDIO, "Unable to prepare: %s\n", snd_strerror(err));
|
||||
return false;
|
||||
|
||||
@@ -28,8 +28,8 @@ public:
|
||||
static bool isValid() {
|
||||
return true;
|
||||
}
|
||||
virtual bool usesMixer() const {
|
||||
return true;
|
||||
virtual bool usesMixer() const {
|
||||
return true;
|
||||
}
|
||||
|
||||
virtual void Update();
|
||||
|
||||
@@ -7,6 +7,7 @@
|
||||
#include "Mixer.h"
|
||||
#include "NullSoundStream.h"
|
||||
#include "DSoundStream.h"
|
||||
#include "XAudio2_7Stream.h"
|
||||
#include "XAudio2Stream.h"
|
||||
#include "AOSoundStream.h"
|
||||
#include "AlsaSoundStream.h"
|
||||
@@ -18,34 +19,47 @@
|
||||
#include "../../Core/Src/ConfigManager.h"
|
||||
|
||||
// This shouldn't be a global, at least not here.
|
||||
SoundStream *soundStream;
|
||||
SoundStream *soundStream = nullptr;
|
||||
|
||||
namespace AudioCommon
|
||||
{
|
||||
SoundStream *InitSoundStream(CMixer *mixer, void *hWnd)
|
||||
{
|
||||
SoundStream *InitSoundStream(CMixer *mixer, void *hWnd)
|
||||
{
|
||||
// TODO: possible memleak with mixer
|
||||
|
||||
std::string backend = SConfig::GetInstance().sBackend;
|
||||
if (backend == BACKEND_OPENAL && OpenALStream::isValid())
|
||||
if (backend == BACKEND_OPENAL && OpenALStream::isValid())
|
||||
soundStream = new OpenALStream(mixer);
|
||||
else if (backend == BACKEND_NULLSOUND && NullSound::isValid())
|
||||
soundStream = new NullSound(mixer, hWnd);
|
||||
else if (backend == BACKEND_DIRECTSOUND && DSound::isValid())
|
||||
else if (backend == BACKEND_NULLSOUND && NullSound::isValid())
|
||||
soundStream = new NullSound(mixer);
|
||||
else if (backend == BACKEND_DIRECTSOUND && DSound::isValid())
|
||||
soundStream = new DSound(mixer, hWnd);
|
||||
else if (backend == BACKEND_XAUDIO2 && XAudio2::isValid())
|
||||
soundStream = new XAudio2(mixer);
|
||||
else if (backend == BACKEND_AOSOUND && AOSound::isValid())
|
||||
else if (backend == BACKEND_XAUDIO2)
|
||||
{
|
||||
if (XAudio2::isValid())
|
||||
soundStream = new XAudio2(mixer);
|
||||
else if (XAudio2_7::isValid())
|
||||
soundStream = new XAudio2_7(mixer);
|
||||
}
|
||||
else if (backend == BACKEND_AOSOUND && AOSound::isValid())
|
||||
soundStream = new AOSound(mixer);
|
||||
else if (backend == BACKEND_ALSA && AlsaSound::isValid())
|
||||
soundStream = new AlsaSound(mixer);
|
||||
else if (backend == BACKEND_COREAUDIO && CoreAudioSound::isValid())
|
||||
else if (backend == BACKEND_COREAUDIO && CoreAudioSound::isValid())
|
||||
soundStream = new CoreAudioSound(mixer);
|
||||
else if (backend == BACKEND_PULSEAUDIO && PulseAudio::isValid())
|
||||
soundStream = new PulseAudio(mixer);
|
||||
else if (backend == BACKEND_OPENSLES && OpenSLESStream::isValid())
|
||||
soundStream = new OpenSLESStream(mixer);
|
||||
if (soundStream != NULL)
|
||||
|
||||
if (!soundStream && NullSound::isValid())
|
||||
{
|
||||
WARN_LOG(DSPHLE, "Could not initialize backend %s, using %s instead.",
|
||||
backend.c_str(), BACKEND_NULLSOUND);
|
||||
soundStream = new NullSound(mixer);
|
||||
}
|
||||
|
||||
if (soundStream)
|
||||
{
|
||||
UpdateSoundStream();
|
||||
if (soundStream->Start())
|
||||
@@ -61,31 +75,32 @@ namespace AudioCommon
|
||||
}
|
||||
PanicAlertT("Could not initialize backend %s.", backend.c_str());
|
||||
}
|
||||
|
||||
PanicAlertT("Sound backend %s is not valid.", backend.c_str());
|
||||
|
||||
delete soundStream;
|
||||
soundStream = NULL;
|
||||
return NULL;
|
||||
soundStream = nullptr;
|
||||
return nullptr;
|
||||
}
|
||||
|
||||
void ShutdownSoundStream()
|
||||
void ShutdownSoundStream()
|
||||
{
|
||||
INFO_LOG(DSPHLE, "Shutting down sound stream");
|
||||
|
||||
if (soundStream)
|
||||
if (soundStream)
|
||||
{
|
||||
soundStream->Stop();
|
||||
if (SConfig::GetInstance().m_DumpAudio)
|
||||
soundStream->GetMixer()->StopLogAudio();
|
||||
//soundStream->StopLogAudio();
|
||||
delete soundStream;
|
||||
soundStream = NULL;
|
||||
soundStream = nullptr;
|
||||
}
|
||||
|
||||
INFO_LOG(DSPHLE, "Done shutting down sound stream");
|
||||
INFO_LOG(DSPHLE, "Done shutting down sound stream");
|
||||
}
|
||||
|
||||
std::vector<std::string> GetSoundBackends()
|
||||
std::vector<std::string> GetSoundBackends()
|
||||
{
|
||||
std::vector<std::string> backends;
|
||||
|
||||
@@ -93,7 +108,7 @@ namespace AudioCommon
|
||||
backends.push_back(BACKEND_NULLSOUND);
|
||||
if (DSound::isValid())
|
||||
backends.push_back(BACKEND_DIRECTSOUND);
|
||||
if (XAudio2::isValid())
|
||||
if (XAudio2_7::isValid() || XAudio2::isValid())
|
||||
backends.push_back(BACKEND_XAUDIO2);
|
||||
if (AOSound::isValid())
|
||||
backends.push_back(BACKEND_AOSOUND);
|
||||
@@ -110,7 +125,7 @@ namespace AudioCommon
|
||||
return backends;
|
||||
}
|
||||
|
||||
bool UseJIT()
|
||||
bool UseJIT()
|
||||
{
|
||||
if (!Movie::IsDSPHLE() && Movie::IsPlayingInput() && Movie::IsConfigSaved())
|
||||
{
|
||||
|
||||
@@ -38,7 +38,7 @@ union UDSPControl
|
||||
UDSPControl(u16 _Hex = 0) : Hex(_Hex) {}
|
||||
};
|
||||
|
||||
namespace AudioCommon
|
||||
namespace AudioCommon
|
||||
{
|
||||
SoundStream *InitSoundStream(CMixer *mixer, void *hWnd);
|
||||
void ShutdownSoundStream();
|
||||
|
||||
@@ -6,7 +6,6 @@
|
||||
#include <functional>
|
||||
|
||||
#include <windows.h>
|
||||
#include <dxerr.h>
|
||||
|
||||
#include "AudioCommon.h"
|
||||
#include "DSoundStream.h"
|
||||
@@ -43,7 +42,7 @@ bool DSound::CreateBuffer()
|
||||
else
|
||||
{
|
||||
// Failed.
|
||||
PanicAlertT("Sound buffer creation failed: %s", DXGetErrorString(res));
|
||||
PanicAlertT("Sound buffer creation failed: %08x", res);
|
||||
dsBuffer = NULL;
|
||||
return false;
|
||||
}
|
||||
|
||||
@@ -9,6 +9,7 @@
|
||||
#include "Thread.h"
|
||||
|
||||
#ifdef _WIN32
|
||||
#include <Windows.h>
|
||||
#include <mmsystem.h>
|
||||
#include <dsound.h>
|
||||
|
||||
@@ -47,7 +48,7 @@ class DSound : public SoundStream
|
||||
bool WriteDataToBuffer(DWORD dwOffset, char* soundData, DWORD dwSoundBytes);
|
||||
|
||||
public:
|
||||
DSound(CMixer *mixer, void *_hWnd = NULL)
|
||||
DSound(CMixer *mixer, void *_hWnd)
|
||||
: SoundStream(mixer)
|
||||
, bufferSize(0)
|
||||
, currentPos(0)
|
||||
@@ -58,7 +59,7 @@ public:
|
||||
{}
|
||||
|
||||
virtual ~DSound() {}
|
||||
|
||||
|
||||
virtual bool Start();
|
||||
virtual void SoundLoop();
|
||||
virtual void SetVolume(int volume);
|
||||
@@ -70,7 +71,7 @@ public:
|
||||
|
||||
#else
|
||||
public:
|
||||
DSound(CMixer *mixer, void *hWnd = NULL)
|
||||
DSound(CMixer *mixer, void *_hWnd)
|
||||
: SoundStream(mixer)
|
||||
{}
|
||||
#endif
|
||||
|
||||
@@ -52,7 +52,7 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
|
||||
// interpolation loop.
|
||||
u32 indexR = Common::AtomicLoad(m_indexR);
|
||||
u32 indexW = Common::AtomicLoad(m_indexW);
|
||||
|
||||
|
||||
if (m_AIplaying) {
|
||||
numLeft = (numLeft > numSamples) ? numSamples : numLeft;
|
||||
|
||||
@@ -125,8 +125,8 @@ unsigned int CMixer::Mix(short* samples, unsigned int numSamples)
|
||||
*(u32*)(samples+i) = *(u32*)(s);
|
||||
// memset(&samples[numLeft * 2], 0, (numSamples - numLeft) * 4);
|
||||
}
|
||||
|
||||
// Flush cached variable
|
||||
|
||||
// Flush cached variable
|
||||
Common::AtomicStore(m_indexR, indexR);
|
||||
|
||||
//when logging, also throttle HLE audio
|
||||
@@ -158,13 +158,13 @@ void CMixer::PushSamples(const short *samples, unsigned int num_samples)
|
||||
// indexR isn't allowed to cache in the audio throttling loop as it
|
||||
// needs to get updates to not deadlock.
|
||||
u32 indexW = Common::AtomicLoad(m_indexW);
|
||||
|
||||
|
||||
if (m_throttle)
|
||||
{
|
||||
// The auto throttle function. This loop will put a ceiling on the CPU MHz.
|
||||
while (num_samples * 2 + ((indexW - Common::AtomicLoad(m_indexR)) & INDEX_MASK) >= MAX_SAMPLES * 2)
|
||||
{
|
||||
if (*PowerPC::GetStatePtr() != PowerPC::CPU_RUNNING || soundStream->IsMuted())
|
||||
if (*PowerPC::GetStatePtr() != PowerPC::CPU_RUNNING || soundStream->IsMuted())
|
||||
break;
|
||||
// Shortcut key for Throttle Skipping
|
||||
if (Host_GetKeyState('\t'))
|
||||
@@ -192,9 +192,9 @@ void CMixer::PushSamples(const short *samples, unsigned int num_samples)
|
||||
{
|
||||
memcpy(&m_buffer[indexW & INDEX_MASK], samples, num_samples * 4);
|
||||
}
|
||||
|
||||
|
||||
Common::AtomicAdd(m_indexW, num_samples * 2);
|
||||
|
||||
|
||||
return;
|
||||
}
|
||||
|
||||
@@ -206,7 +206,7 @@ unsigned int CMixer::GetNumSamples()
|
||||
// We also can't say the current interpolation state (specially
|
||||
// the frac), so to be sure, subtract one again to be sure not
|
||||
// to underflow the fifo.
|
||||
|
||||
|
||||
u32 numSamples = ((Common::AtomicLoad(m_indexW) - Common::AtomicLoad(m_indexR)) & INDEX_MASK) / 2;
|
||||
|
||||
if (AudioInterface::GetAIDSampleRate() == m_sampleRate)
|
||||
|
||||
@@ -14,7 +14,7 @@
|
||||
#define RESERVED_SAMPLES (256)
|
||||
|
||||
class CMixer {
|
||||
|
||||
|
||||
public:
|
||||
CMixer(unsigned int AISampleRate = 48000, unsigned int DACSampleRate = 48000, unsigned int BackendSampleRate = 32000)
|
||||
: m_aiSampleRate(AISampleRate)
|
||||
@@ -89,7 +89,7 @@ protected:
|
||||
int m_channels;
|
||||
|
||||
WaveFileWriter g_wave_writer;
|
||||
|
||||
|
||||
bool m_HLEready;
|
||||
bool m_logAudio;
|
||||
|
||||
|
||||
@@ -16,7 +16,7 @@ class NullSound : public SoundStream
|
||||
short realtimeBuffer[BUF_SIZE / sizeof(short)];
|
||||
|
||||
public:
|
||||
NullSound(CMixer *mixer, void *hWnd = NULL)
|
||||
NullSound(CMixer *mixer)
|
||||
: SoundStream(mixer)
|
||||
{}
|
||||
|
||||
|
||||
@@ -48,9 +48,9 @@ public:
|
||||
OpenALStream(CMixer *mixer, void *hWnd = NULL)
|
||||
: SoundStream(mixer)
|
||||
, uiSource(0)
|
||||
{};
|
||||
{}
|
||||
|
||||
virtual ~OpenALStream() {};
|
||||
virtual ~OpenALStream() {}
|
||||
|
||||
virtual bool Start();
|
||||
virtual void SoundLoop();
|
||||
@@ -74,7 +74,9 @@ private:
|
||||
u8 numBuffers;
|
||||
#else
|
||||
public:
|
||||
OpenALStream(CMixer *mixer, void *hWnd = NULL): SoundStream(mixer) {}
|
||||
OpenALStream(CMixer *mixer)
|
||||
: SoundStream(mixer)
|
||||
{}
|
||||
#endif // HAVE_OPENAL
|
||||
};
|
||||
|
||||
|
||||
@@ -21,7 +21,7 @@ static SLPlayItf bqPlayerPlay;
|
||||
static SLAndroidSimpleBufferQueueItf bqPlayerBufferQueue;
|
||||
static SLMuteSoloItf bqPlayerMuteSolo;
|
||||
static SLVolumeItf bqPlayerVolume;
|
||||
static CMixer *g_mixer;
|
||||
static CMixer *g_mixer;
|
||||
#define BUFFER_SIZE 512
|
||||
#define BUFFER_SIZE_IN_SAMPLES (BUFFER_SIZE / 2)
|
||||
|
||||
|
||||
@@ -24,7 +24,7 @@ public:
|
||||
PulseAudio(CMixer *mixer);
|
||||
|
||||
virtual bool Start();
|
||||
virtual void Stop();
|
||||
virtual void Stop();
|
||||
|
||||
static bool isValid() {return true;}
|
||||
|
||||
|
||||
@@ -21,11 +21,11 @@ protected:
|
||||
WaveFileWriter g_wave_writer;
|
||||
bool m_muted;
|
||||
|
||||
public:
|
||||
SoundStream(CMixer *mixer) : m_mixer(mixer), threadData(0), m_logAudio(false), m_muted(false) {}
|
||||
virtual ~SoundStream() { delete m_mixer;}
|
||||
public:
|
||||
SoundStream(CMixer *mixer) : m_mixer(mixer), threadData(0), m_logAudio(false), m_muted(false) {}
|
||||
virtual ~SoundStream() { delete m_mixer; }
|
||||
|
||||
static bool isValid() { return false; }
|
||||
static bool isValid() { return false; }
|
||||
virtual CMixer *GetMixer() const { return m_mixer; }
|
||||
virtual bool Start() { return false; }
|
||||
virtual void SetVolume(int) {}
|
||||
|
||||
@@ -121,7 +121,7 @@ void WaveFileWriter::AddStereoSamplesBE(const short *sample_data, u32 count)
|
||||
if (count > BUF_SIZE * 2)
|
||||
PanicAlert("WaveFileWriter - buffer too small (count = %u).", count);
|
||||
|
||||
if (skip_silence)
|
||||
if (skip_silence)
|
||||
{
|
||||
bool all_zero = true;
|
||||
|
||||
|
||||
Some files were not shown because too many files have changed in this diff Show More
Reference in New Issue
Block a user