Restore support for the Nokia N-Gage (#12148)

src/audio/SDL_audio.c

@@ -77,6 +77,9 @@ static const AudioBootStrap *const bootstrap[] = {
 #ifdef SDL_AUDIO_DRIVER_N3DS
     &N3DSAUDIO_bootstrap,
 #endif
+#ifdef SDL_AUDIO_DRIVER_NGAGE
+    &NGAGEAUDIO_bootstrap,
+#endif
 #ifdef SDL_AUDIO_DRIVER_EMSCRIPTEN
     &EMSCRIPTENAUDIO_bootstrap,
 #endif

src/audio/SDL_sysaudio.h

@@ -393,6 +393,7 @@ extern AudioBootStrap PS2AUDIO_bootstrap;
 extern AudioBootStrap PSPAUDIO_bootstrap;
 extern AudioBootStrap VITAAUD_bootstrap;
 extern AudioBootStrap N3DSAUDIO_bootstrap;
+extern AudioBootStrap NGAGEAUDIO_bootstrap;
 extern AudioBootStrap EMSCRIPTENAUDIO_bootstrap;
 extern AudioBootStrap QSAAUDIO_bootstrap;
 

src/audio/ngage/SDL_ngageaudio.c

@@ -0,0 +1,103 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#include "SDL_internal.h"
+
+#ifdef SDL_AUDIO_DRIVER_NGAGE
+
+#include "../SDL_sysaudio.h"
+#include "SDL_ngageaudio.h"
+
+static SDL_AudioDevice *devptr = NULL;
+
+SDL_AudioDevice *NGAGE_GetAudioDeviceAddr()
+{
+    return devptr;
+}
+
+static bool NGAGEAUDIO_OpenDevice(SDL_AudioDevice *device)
+{
+    SDL_PrivateAudioData *phdata = SDL_calloc(1, sizeof(SDL_PrivateAudioData));
+    if (!phdata) {
+        SDL_OutOfMemory();
+        return false;
+    }
+    device->hidden = phdata;
+
+    phdata->buffer = SDL_calloc(1, device->buffer_size);
+    if (!phdata->buffer) {
+        SDL_OutOfMemory();
+        SDL_free(phdata);
+        return false;
+    }
+    devptr = device;
+
+    // Since the phone can change the sample rate during a phone call,
+    // we set the sample rate to 8KHz to be safe.  Even though it
+    // might be possible to adjust the sample rate dynamically, it's
+    // not supported by the current implementation.
+
+    device->spec.format = SDL_AUDIO_S16LE;
+    device->spec.channels = 1;
+    device->spec.freq = 8000;
+
+    SDL_UpdatedAudioDeviceFormat(device);
+
+    return true;
+}
+
+static Uint8 *NGAGEAUDIO_GetDeviceBuf(SDL_AudioDevice *device, int *buffer_size)
+{
+    SDL_PrivateAudioData *phdata = (SDL_PrivateAudioData *)device->hidden;
+    if (!phdata) {
+        *buffer_size = 0;
+        return 0;
+    }
+
+    *buffer_size = device->buffer_size;
+    return phdata->buffer;
+}
+
+static void NGAGEAUDIO_CloseDevice(SDL_AudioDevice *device)
+{
+    if (device->hidden) {
+        SDL_free(device->hidden->buffer);
+        SDL_free(device->hidden);
+    }
+
+    return;
+}
+
+static bool NGAGEAUDIO_Init(SDL_AudioDriverImpl *impl)
+{
+    impl->OpenDevice = NGAGEAUDIO_OpenDevice;
+    impl->GetDeviceBuf = NGAGEAUDIO_GetDeviceBuf;
+    impl->CloseDevice = NGAGEAUDIO_CloseDevice;
+
+    impl->ProvidesOwnCallbackThread = true;
+    impl->OnlyHasDefaultPlaybackDevice = true;
+
+    return true;
+}
+
+AudioBootStrap NGAGEAUDIO_bootstrap = { "N-Gage", "N-Gage audio driver", NGAGEAUDIO_Init, false };
+
+#endif // SDL_AUDIO_DRIVER_NGAGE

src/audio/ngage/SDL_ngageaudio.cpp

@@ -0,0 +1,368 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "SDL_ngageaudio.h"
+#include "../SDL_sysaudio.h"
+#include "SDL_internal.h"
+
+#ifdef __cplusplus
+}
+#endif
+
+#ifdef SDL_AUDIO_DRIVER_NGAGE
+
+#include "SDL_ngageaudio.hpp"
+
+CAudio::CAudio() : CActive(EPriorityStandard), iBufDes(NULL, 0) {}
+
+CAudio *CAudio::NewL(TInt aLatency)
+{
+    CAudio *self = new (ELeave) CAudio();
+    CleanupStack::PushL(self);
+    self->ConstructL(aLatency);
+    CleanupStack::Pop(self);
+    return self;
+}
+
+void CAudio::ConstructL(TInt aLatency)
+{
+    CActiveScheduler::Add(this);
+    User::LeaveIfError(iTimer.CreateLocal());
+    iTimerCreated = ETrue;
+
+    iStream = CMdaAudioOutputStream::NewL(*this);
+    if (!iStream) {
+        SDL_Log("Error: Failed to create audio stream");
+        User::Leave(KErrNoMemory);
+    }
+
+    iLatency = aLatency;
+    iLatencySamples = aLatency * 8; // 8kHz.
+
+    // Determine minimum and maximum number of samples to write with one
+    // WriteL request.
+    iMinWrite = iLatencySamples / 8;
+    iMaxWrite = iLatencySamples / 2;
+
+    // Set defaults.
+    iState = EStateNone;
+    iTimerCreated = EFalse;
+    iTimerActive = EFalse;
+}
+
+CAudio::~CAudio()
+{
+    if (iStream) {
+        iStream->Stop();
+
+        while (iState != EStateDone) {
+            User::After(100000); // 100ms.
+        }
+
+        delete iStream;
+    }
+}
+
+void CAudio::Start()
+{
+    if (iStream) {
+        // Set to 8kHz mono audio.
+        iStreamSettings.iChannels = TMdaAudioDataSettings::EChannelsMono;
+        iStreamSettings.iSampleRate = TMdaAudioDataSettings::ESampleRate8000Hz;
+        iStream->Open(&iStreamSettings);
+        iState = EStateOpening;
+    } else {
+        SDL_Log("Error: Failed to open audio stream");
+    }
+}
+
+// Feeds more processed data to the audio stream.
+void CAudio::Feed()
+{
+    // If a WriteL is already in progress, or we aren't even playing;
+    // do nothing!
+    if ((iState != EStateWriting) && (iState != EStatePlaying)) {
+        return;
+    }
+
+    // Figure out the number of samples that really have been played
+    // through the output.
+    TTimeIntervalMicroSeconds pos = iStream->Position();
+
+    TInt played = 8 * (pos.Int64() / TInt64(1000)).GetTInt(); // 8kHz.
+
+    played += iBaseSamplesPlayed;
+
+    // Determine the difference between the number of samples written to
+    // CMdaAudioOutputStream and the number of samples it has played.
+    // The difference is the amount of data in the buffers.
+    if (played < 0) {
+        played = 0;
+    }
+
+    TInt buffered = iSamplesWritten - played;
+    if (buffered < 0) {
+        buffered = 0;
+    }
+
+    if (iState == EStateWriting) {
+        return;
+    }
+
+    // The trick for low latency: Do not let the buffers fill up beyond the
+    // latency desired! We write as many samples as the difference between
+    // the latency target (in samples) and the amount of data buffered.
+    TInt samplesToWrite = iLatencySamples - buffered;
+
+    // Do not write very small blocks. This should improve efficiency, since
+    // writes to the streaming API are likely to be expensive.
+    if (samplesToWrite < iMinWrite) {
+        // Not enough data to write, set up a timer to fire after a while.
+        // Try againwhen it expired.
+        if (iTimerActive) {
+            return;
+        }
+        iTimerActive = ETrue;
+        SetActive();
+        iTimer.After(iStatus, (1000 * iLatency) / 8);
+        return;
+    }
+
+    // Do not write more than the set number of samples at once.
+    int numSamples = samplesToWrite;
+    if (numSamples > iMaxWrite) {
+        numSamples = iMaxWrite;
+    }
+
+    SDL_AudioDevice *device = NGAGE_GetAudioDeviceAddr();
+    if (device) {
+        SDL_PrivateAudioData *phdata = (SDL_PrivateAudioData *)device->hidden;
+
+        iBufDes.Set(phdata->buffer, 2 * numSamples, 2 * numSamples);
+        iStream->WriteL(iBufDes);
+        iState = EStateWriting;
+
+        // Keep track of the number of samples written (for latency calculations).
+        iSamplesWritten += numSamples;
+    } else {
+        // Output device not ready yet. Let's go for another round.
+        if (iTimerActive) {
+            return;
+        }
+        iTimerActive = ETrue;
+        SetActive();
+        iTimer.After(iStatus, (1000 * iLatency) / 8);
+    }
+}
+
+void CAudio::RunL()
+{
+    iTimerActive = EFalse;
+    Feed();
+}
+
+void CAudio::DoCancel()
+{
+    iTimerActive = EFalse;
+    iTimer.Cancel();
+}
+
+void CAudio::StartThread()
+{
+    TInt heapMinSize = 8192;        // 8 KB initial heap size.
+    TInt heapMaxSize = 1024 * 1024; // 1 MB maximum heap size.
+
+    TInt err = iProcess.Create(_L("ProcessThread"), ProcessThreadCB, KDefaultStackSize * 2, heapMinSize, heapMaxSize, this);
+    if (err == KErrNone) {
+        iProcess.SetPriority(EPriorityLess);
+        iProcess.Resume();
+    } else {
+        SDL_Log("Error: Failed to create audio processing thread: %d", err);
+    }
+}
+
+void CAudio::StopThread()
+{
+    if (iStreamStarted) {
+        iProcess.Kill(KErrNone);
+        iProcess.Close();
+        iStreamStarted = EFalse;
+    }
+}
+
+TInt CAudio::ProcessThreadCB(TAny *aPtr)
+{
+    CAudio *self = static_cast<CAudio *>(aPtr);
+    SDL_AudioDevice *device = NGAGE_GetAudioDeviceAddr();
+
+    while (self->iStreamStarted) {
+        if (device) {
+            SDL_PlaybackAudioThreadIterate(device);
+        } else {
+            device = NGAGE_GetAudioDeviceAddr();
+        }
+        User::After(100000); // 100ms.
+    }
+    return KErrNone;
+}
+
+void CAudio::MaoscOpenComplete(TInt aError)
+{
+    if (aError == KErrNone) {
+        iStream->SetVolume(1);
+        iStreamStarted = ETrue;
+        StartThread();
+
+    } else {
+        SDL_Log("Error: Failed to open audio stream: %d", aError);
+    }
+}
+
+void CAudio::MaoscBufferCopied(TInt aError, const TDesC8 & /*aBuffer*/)
+{
+    if (aError == KErrNone) {
+        iState = EStatePlaying;
+        Feed();
+    } else if (aError == KErrAbort) {
+        // The stream has been stopped.
+        iState = EStateDone;
+    } else {
+        SDL_Log("Error: Failed to copy audio buffer: %d", aError);
+    }
+}
+
+void CAudio::MaoscPlayComplete(TInt aError)
+{
+    // If we finish due to an underflow, we'll need to restart playback.
+    // Normally KErrUnderlow is raised   at stream end, but in our case the API
+    // should never see the stream end -- we are continuously feeding it more
+    // data!  Many underflow errors mean that the latency target is too low.
+    if (aError == KErrUnderflow) {
+        // The number of samples played gets resetted to zero when we restart
+        // playback after underflow.
+        iBaseSamplesPlayed = iSamplesWritten;
+
+        iStream->Stop();
+        Cancel();
+
+        iStream->SetAudioPropertiesL(TMdaAudioDataSettings::ESampleRate8000Hz, TMdaAudioDataSettings::EChannelsMono);
+
+        iState = EStatePlaying;
+        Feed();
+        return;
+
+    } else if (aError != KErrNone) {
+        // Handle error.
+    }
+
+    // We shouldn't get here.
+    SDL_Log("%s: %d", __FUNCTION__, aError);
+}
+
+static TBool gAudioRunning;
+
+TBool AudioIsReady()
+{
+    return gAudioRunning;
+}
+
+TInt AudioThreadCB(TAny *aParams)
+{
+    CTrapCleanup *cleanup = CTrapCleanup::New();
+    if (!cleanup) {
+        return KErrNoMemory;
+    }
+
+    CActiveScheduler *scheduler = new CActiveScheduler();
+    if (!scheduler) {
+        delete cleanup;
+        return KErrNoMemory;
+    }
+
+    CActiveScheduler::Install(scheduler);
+
+    TRAPD(err,
+          {
+              TInt latency = *(TInt *)aParams;
+              CAudio *audio = CAudio::NewL(latency);
+              CleanupStack::PushL(audio);
+
+              gAudioRunning = ETrue;
+              audio->Start();
+              TBool once = EFalse;
+
+              while (gAudioRunning) {
+                  // Allow active scheduler to process any events.
+                  TInt error;
+                  CActiveScheduler::RunIfReady(error, CActive::EPriorityIdle);
+
+                  if (!once) {
+                      SDL_AudioDevice *device = NGAGE_GetAudioDeviceAddr();
+                      if (device) {
+                          // Stream ready; start feeding audio data.
+                          // After feeding it once, the callbacks will take over.
+                          audio->iState = CAudio::EStatePlaying;
+                          audio->Feed();
+                          once = ETrue;
+                      }
+                  }
+
+                  User::After(100000); // 100ms.
+              }
+
+              CleanupStack::PopAndDestroy(audio);
+          });
+
+    delete scheduler;
+    delete cleanup;
+    return err;
+}
+
+RThread audioThread;
+
+void InitAudio(TInt *aLatency)
+{
+    _LIT(KAudioThreadName, "AudioThread");
+
+    TInt err = audioThread.Create(KAudioThreadName, AudioThreadCB, KDefaultStackSize, 0, aLatency);
+    if (err != KErrNone) {
+        User::Leave(err);
+    }
+
+    audioThread.Resume();
+}
+
+void DeinitAudio()
+{
+    gAudioRunning = EFalse;
+
+    TRequestStatus status;
+    audioThread.Logon(status);
+    User::WaitForRequest(status);
+
+    audioThread.Close();
+}
+
+#endif // SDL_AUDIO_DRIVER_NGAGE

src/audio/ngage/SDL_ngageaudio.h

@@ -0,0 +1,44 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+  Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+#include "SDL_internal.h"
+
+#ifndef SDL_ngageaudio_h
+#define SDL_ngageaudio_h
+
+typedef struct SDL_PrivateAudioData
+{
+    Uint8 *buffer;
+
+} SDL_PrivateAudioData;
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "../SDL_sysaudio.h"
+
+SDL_AudioDevice *NGAGE_GetAudioDeviceAddr();
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif // SDL_ngageaudio_h

src/audio/ngage/SDL_ngageaudio.hpp

@@ -0,0 +1,98 @@
+/*
+  Simple DirectMedia Layer
+  Copyright (C) 1997-2025 Sam Lantinga <slouken@libsdl.org>
+
+  This software is provided 'as-is', without any express or implied
+  warranty.  In no event will the authors be held liable for any damages
+  arising from the use of this software.
+
+      Permission is granted to anyone to use this software for any purpose,
+  including commercial applications, and to alter it and redistribute it
+  freely, subject to the following restrictions:
+
+  1. The origin of this software must not be misrepresented; you must not
+     claim that you wrote the original software. If you use this software
+     in a product, an acknowledgment in the product documentation would be
+     appreciated but is not required.
+  2. Altered source versions must be plainly marked as such, and must not be
+     misrepresented as being the original software.
+  3. This notice may not be removed or altered from any source distribution.
+*/
+
+#ifndef SDL_ngageaudio_hpp
+#define SDL_ngageaudio_hpp
+
+#include <e32base.h>
+#include <e32std.h>
+#include <mda/common/audio.h>
+#include <mdaaudiooutputstream.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#include "../SDL_sysaudio.h"
+#include "SDL_ngageaudio.h"
+
+#ifdef __cplusplus
+}
+#endif
+
+TBool AudioIsReady();
+void InitAudio(TInt *aLatency);
+void DeinitAudio();
+
+class CAudio : public CActive, public MMdaAudioOutputStreamCallback
+{
+  public:
+    static CAudio *NewL(TInt aLatency);
+    ~CAudio();
+
+    void ConstructL(TInt aLatency);
+    void Start();
+    void Feed();
+
+    void RunL();
+    void DoCancel();
+
+    static TInt ProcessThreadCB(TAny * /*aPtr*/);
+
+    // From MMdaAudioOutputStreamCallback
+    void MaoscOpenComplete(TInt aError);
+    void MaoscBufferCopied(TInt aError, const TDesC8 &aBuffer);
+    void MaoscPlayComplete(TInt aError);
+
+    enum
+    {
+        EStateNone = 0,
+        EStateOpening,
+        EStatePlaying,
+        EStateWriting,
+        EStateDone
+    } iState;
+
+  private:
+    CAudio();
+    void StartThread();
+    void StopThread();
+
+    CMdaAudioOutputStream *iStream;
+    TMdaAudioDataSettings iStreamSettings;
+    TBool iStreamStarted;
+
+    TPtr8 iBufDes;           // Descriptor for the buffer.
+    TInt iLatency;           // Latency target in ms
+    TInt iLatencySamples;    // Latency target in samples.
+    TInt iMinWrite;          // Min number of samples to write per turn.
+    TInt iMaxWrite;          // Max number of samples to write per turn.
+    TInt iBaseSamplesPlayed; // amples played before last restart.
+    TInt iSamplesWritten;    // Number of samples written so far.
+
+    RTimer iTimer;
+    TBool iTimerCreated;
+    TBool iTimerActive;
+
+    RThread iProcess;
+};
+
+#endif // SDL_ngageaudio_hpp