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Diffstat (limited to 'samples/quake/jni/snd_android.cpp')
| -rw-r--r-- | samples/quake/jni/snd_android.cpp | 738 |
1 files changed, 0 insertions, 738 deletions
diff --git a/samples/quake/jni/snd_android.cpp b/samples/quake/jni/snd_android.cpp deleted file mode 100644 index 1327023..0000000 --- a/samples/quake/jni/snd_android.cpp +++ /dev/null @@ -1,738 +0,0 @@ -/* - * snd_android.c - * Android-specific sound interface - * - */ - -#include "quakedef.h" - -#include <pthread.h> -#include <time.h> -#include <math.h> -#include <stdlib.h> -#include <unistd.h> - -#include <android/log.h> -#include <SLES/OpenSLES.h> - -#define LOG_TAG "Quake snd_android" -#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__) -#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__) - -const size_t SAMPLE_RATE = 11025; - - -const size_t BYTES_PER_SAMPLE = 2; -const size_t CHANNEL_COUNT = 2; -const size_t BITS_PER_SAMPLE = 8 * BYTES_PER_SAMPLE; - -const size_t TOTAL_BUFFER_SIZE = 4 * 1024; - -#define MAX_NUMBER_INTERFACES 3 - -/* Local storage for Audio data in 16 bit words */ -#define AUDIO_DATA_STORAGE_SIZE (TOTAL_BUFFER_SIZE / 2) -/* Audio data buffer size in 16 bit words. 8 data segments are used in -this simple example */ -#define AUDIO_DATA_BUFFER_SIZE (4096/8) - -const size_t NUMBER_OF_BUFFERS = AUDIO_DATA_STORAGE_SIZE / AUDIO_DATA_BUFFER_SIZE; - -/* Checks for error. If any errors exit the application! */ -void CheckErr( SLresult res ) -{ - if ( res != SL_RESULT_SUCCESS ) - { - fprintf(stdout, "%u SL failure, exiting\n", res); - exit(EXIT_FAILURE); - } - else { - //fprintf(stdout, "%d SL success, proceeding...\n", res); - } -} - -/* Structure for passing information to callback function */ -typedef struct CallbackCntxt_ { - SLPlayItf playItf; - SLint16* pDataBase; // Base adress of local audio data storage - SLint16* pData; // Current adress of local audio data storage - SLuint32 size; -} CallbackCntxt; - -/* Local storage for Audio data */ -SLint16 pcmData[AUDIO_DATA_STORAGE_SIZE]; - -/* Callback for Buffer Queue events */ -void BufferQueueCallback( - SLBufferQueueItf queueItf, - void *pContext) -{ - //fprintf(stdout, "BufferQueueCallback called\n"); - SLresult res; - //fprintf(stdout, " pContext=%p\n", pContext); - CallbackCntxt *pCntxt = (CallbackCntxt*)pContext; - - if (pCntxt->pData >= (pCntxt->pDataBase + pCntxt->size)) { - pCntxt->pData = pCntxt->pDataBase; - } - { - //fprintf(stdout, "callback: before enqueue\n"); - res = (*queueItf)->Enqueue(queueItf, (void*) pCntxt->pData, - 2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */ - CheckErr(res); - /* Increase data pointer by buffer size */ - pCntxt->pData += AUDIO_DATA_BUFFER_SIZE; - } - //fprintf(stdout, "end of BufferQueueCallback()\n"); -} - -SLEngineItf EngineItf; - -SLint32 numOutputs = 0; -SLuint32 deviceID = 0; - - -SLDataSource audioSource; -SLDataLocator_BufferQueue bufferQueue; -SLDataFormat_PCM pcm; - -SLDataSink audioSink; -SLDataLocator_OutputMix locator_outputmix; - - -SLVolumeItf volumeItf; - - -SLboolean required[MAX_NUMBER_INTERFACES]; -SLInterfaceID iidArray[MAX_NUMBER_INTERFACES]; - -/* Callback context for the buffer queue callback function */ -CallbackCntxt cntxt; - -static SLObjectItf OutputMix; -static SLPlayItf playItf; -static SLObjectItf player; -static SLBufferQueueItf bufferQueueItf; -static SLBufferQueueState state; - -/* Play some audio from a buffer queue */ -void TestPlaySawtoothBufferQueue( SLObjectItf sl ) -{ - SLresult res; - int i; - - /* Get the SL Engine Interface which is implicit */ - res = (*sl)->GetInterface(sl, SL_IID_ENGINE, (void*)&EngineItf); - CheckErr(res); - - /* Initialize arrays required[] and iidArray[] */ - for (i=0;i<MAX_NUMBER_INTERFACES;i++) - { - required[i] = SL_BOOLEAN_FALSE; - iidArray[i] = SL_IID_NULL; - } - - // Set arrays required[] and iidArray[] for VOLUME interface - required[0] = SL_BOOLEAN_TRUE; - iidArray[0] = SL_IID_VOLUME; - // Create Output Mix object to be used by player - res = (*EngineItf)->CreateOutputMix(EngineItf, &OutputMix, 0, - iidArray, required); CheckErr(res); - - // Realizing the Output Mix object in synchronous mode. - res = (*OutputMix)->Realize(OutputMix, SL_BOOLEAN_FALSE); - CheckErr(res); - -#if 0 - res = (*OutputMix)->GetInterface(OutputMix, SL_IID_VOLUME, - (void*)&volumeItf); CheckErr(res); -#endif - - /* Setup the data source structure for the buffer queue */ - bufferQueue.locatorType = SL_DATALOCATOR_BUFFERQUEUE; - bufferQueue.numBuffers = 4; /* Four buffers in our buffer queue */ - - /* Setup the format of the content in the buffer queue */ - pcm.formatType = SL_DATAFORMAT_PCM; - pcm.numChannels = 2; - pcm.samplesPerSec = SL_SAMPLINGRATE_11_025; - pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16; - pcm.containerSize = 16; - pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT; - pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; - - audioSource.pFormat = (void *)&pcm; - audioSource.pLocator = (void *)&bufferQueue; - - /* Setup the data sink structure */ - locator_outputmix.locatorType = SL_DATALOCATOR_OUTPUTMIX; - locator_outputmix.outputMix = OutputMix; - audioSink.pLocator = (void *)&locator_outputmix; - audioSink.pFormat = NULL; - - /* Initialize the audio data to silence */ - memset(pcmData, 0, sizeof(pcmData)); - - /* Initialize the context for Buffer queue callbacks */ - cntxt.pDataBase = /*(void*)&*/pcmData; - cntxt.pData = cntxt.pDataBase; - cntxt.size = sizeof(pcmData) / 2; - - /* Set arrays required[] and iidArray[] for SEEK interface - (PlayItf is implicit) */ - required[0] = SL_BOOLEAN_TRUE; - iidArray[0] = SL_IID_BUFFERQUEUE; - - /* Create the music player */ - res = (*EngineItf)->CreateAudioPlayer(EngineItf, &player, - &audioSource, &audioSink, 1, iidArray, required); CheckErr(res); - fprintf(stdout, "bufferQueue example: after CreateAudioPlayer\n"); - - /* Realizing the player in synchronous mode. */ - res = (*player)->Realize(player, SL_BOOLEAN_FALSE); CheckErr(res); - fprintf(stdout, "bufferQueue example: after Realize\n"); - - /* Get seek and play interfaces */ - res = (*player)->GetInterface(player, SL_IID_PLAY, (void*)&playItf); - CheckErr(res); - fprintf(stdout, "bufferQueue example: after GetInterface(PLAY)\n"); - - res = (*player)->GetInterface(player, SL_IID_BUFFERQUEUE, - (void*)&bufferQueueItf); CheckErr(res); - - /* Setup to receive buffer queue event callbacks */ - res = (*bufferQueueItf)->RegisterCallback(bufferQueueItf, - BufferQueueCallback, &cntxt); CheckErr(res); - -#if 0 - /* Before we start set volume to -3dB (-300mB) */ - res = (*volumeItf)->SetVolumeLevel(volumeItf, -300); CheckErr(res); -#endif - - /* Enqueue a few buffers to get the ball rolling */ - res = (*bufferQueueItf)->Enqueue(bufferQueueItf, cntxt.pData, - 2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */ - CheckErr(res); - cntxt.pData += AUDIO_DATA_BUFFER_SIZE; - - res = (*bufferQueueItf)->Enqueue(bufferQueueItf, cntxt.pData, - 2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */ - CheckErr(res); - cntxt.pData += AUDIO_DATA_BUFFER_SIZE; - - res = (*bufferQueueItf)->Enqueue(bufferQueueItf, cntxt.pData, - 2 * AUDIO_DATA_BUFFER_SIZE); /* Size given in bytes. */ - CheckErr(res); - cntxt.pData += AUDIO_DATA_BUFFER_SIZE; - - /* Play the PCM samples using a buffer queue */ - fprintf(stdout, "bufferQueue example: starting to play\n"); - res = (*playItf)->SetPlayState( playItf, SL_PLAYSTATE_PLAYING ); - CheckErr(res); - - /* Wait until the PCM data is done playing, the buffer queue callback - will continue to queue buffers until the entire PCM data has been - played. This is indicated by waiting for the count member of the - SLBufferQueueState to go to zero. - */ - res = (*bufferQueueItf)->GetState(bufferQueueItf, &state); - CheckErr(res); - -#if 0 - // while (state.playIndex < 100) { - while (state.count) { - usleep(10000); - (*bufferQueueItf)->GetState(bufferQueueItf, &state); - } - - #endif -} - -SLObjectItf gSoundEngine; - -int startAndroidSound() -{ - SLresult res; - - SLEngineOption EngineOption[] = { - {(SLuint32) SL_ENGINEOPTION_THREADSAFE, - (SLuint32) SL_BOOLEAN_TRUE}}; - - res = slCreateEngine( &gSoundEngine, 1, EngineOption, 0, NULL, NULL); - CheckErr(res); - /* Realizing the SL Engine in synchronous mode. */ - res = (*gSoundEngine)->Realize(gSoundEngine, SL_BOOLEAN_FALSE); CheckErr(res); - - /* Run the test */ - TestPlaySawtoothBufferQueue(gSoundEngine); - return EXIT_SUCCESS; -} - -void finishAndroidSound() -{ - SLresult res; - - if (gSoundEngine == NULL) { - return; - } - - /* Make sure player is stopped */ - if (playItf != NULL) { - res = (*playItf)->SetPlayState(playItf, SL_PLAYSTATE_STOPPED); - CheckErr(res); - playItf = NULL; - } - - if (player != NULL) { - /* Destroy the player */ - (*player)->Destroy(player); - player = NULL; - } - - if (OutputMix != NULL) { - /* Destroy Output Mix object */ - (*OutputMix)->Destroy(OutputMix); - OutputMix = NULL; - } - - /* Shutdown OpenSL ES */ - (*gSoundEngine)->Destroy(gSoundEngine); - gSoundEngine = NULL; -} - -#if 1 - -/* -================== -SNDDMA_Init - -Try to find a sound device to mix for. -Returns false if nothing is found. -================== -*/ -qboolean SNDDMA_Init(void) -{ - // Initialize Quake's idea of a DMA buffer. - - shm = &sn; - memset((void*)&sn, 0, sizeof(sn)); - - shm->splitbuffer = false; // Not used. - shm->samplebits = 16; - shm->speed = 11025; - shm->channels = 2; - shm->samples = TOTAL_BUFFER_SIZE / BYTES_PER_SAMPLE; - shm->samplepos = 0; // Not used. - shm->buffer = (unsigned char*) pcmData; - shm->submission_chunk = 1; // Not used. - - shm->soundalive = true; - - if ( (shm->samples & 0x1ff) != 0 ) { - LOGE("SNDDDMA_Init: samples must be power of two."); - return false; - } - - if ( shm->buffer == 0 ) { - LOGE("SNDDDMA_Init: Could not allocate sound buffer."); - return false; - } - - int result = startAndroidSound(); - return result == EXIT_SUCCESS; -} - -/* -============== -SNDDMA_GetDMAPos - -return the current sample position (in mono samples read) -inside the recirculating dma buffer, so the mixing code will know -how many sample are required to fill it up. -=============== -*/ -int SNDDMA_GetDMAPos(void) -{ - SLresult res; - if (bufferQueueItf != NULL) { - res = (*bufferQueueItf)->GetState(bufferQueueItf, &state); - CheckErr(res); - // Index of the currently playing or filling buffer. - SLuint32 playIndex = state.playIndex; - int ringIndex = playIndex % NUMBER_OF_BUFFERS; - return ringIndex * AUDIO_DATA_BUFFER_SIZE; - } - return 0; -} - -/* -=============== -SNDDMA_ReportWrite - -Report valid data being written into the DMA buffer by the sound mixing code. -This is an Android specific API. -================ -*/ -void SNDDMA_ReportWrite(size_t lengthBytes) { - // TODO: keep track of how much of the sound ring buffer has sound in it, - // detect starvation, and mix silence when we are starving. -} - -/* -============== -SNDDMA_Submit - -Send sound to device if buffer isn't really the dma buffer -=============== -*/ -void SNDDMA_Submit(void) -{ -} - -/* -============== -SNDDMA_Shutdown - -Reset the sound device for exiting -=============== -*/ -void SNDDMA_Shutdown(void) -{ - finishAndroidSound(); -} - - -#else - -// Written by the callback function running in an audio thread. -// index in bytes of where we last read. - -static volatile size_t gDMAByteIndex; - - -// Written by main thread -static size_t gAvailableBytes; -static bool gSoundMixingStarted; - -// The condition is "new data is now available" - -static pthread_mutex_t condition_mutex = PTHREAD_MUTEX_INITIALIZER; -static pthread_cond_t condition_cond = PTHREAD_COND_INITIALIZER; - -/* -================== -SNDDMA_Init - -Try to find a sound device to mix for. -Returns false if nothing is found. -================== -*/ - - -const size_t SAMPLE_RATE = 11025; - - -const size_t BYTES_PER_SAMPLE = 2; -const size_t CHANNEL_COUNT = 2; -const size_t BITS_PER_SAMPLE = 8 * BYTES_PER_SAMPLE; - -const size_t TOTAL_BUFFER_SIZE = 16 * 1024; - -static size_t min(size_t a, size_t b) { - return a < b ? a : b; -} - -static size_t mod(size_t value, size_t mod) { - return value % mod; -} - -static size_t next(size_t value, size_t mod) { - value = value + 1; - if ( value >= mod ) { - value = 0; - } - return value; -} - -static size_t prev(size_t value, size_t mod) { - if ( value <= 0 ) { - value = mod; - } - return value - 1; -} - - -static bool enableSound() { - - if (COM_CheckParm("-nosound")) - return false; - - return true; -} - -// Choose one: - -// #define GENERATE_SINE_WAVE -#define NORMAL_SOUND - -#ifdef GENERATE_SINE_WAVE - -static const float p = 2 * M_PI * 440.0f / SAMPLE_RATE; -static float left = 0.0f; -static float right = 0.0f; - -static float sinef(float x) -{ - const float A = 1.0f / (2.0f*M_PI); - const float B = -16.0f; - const float C = 8.0f; - - // scale angle for easy argument reduction - x *= A; - - if (fabsf(x) >= 0.5f) { - // Argument reduction - x = x - ceilf(x + 0.5f) + 1.0f; - } - - const float y = B*x*fabsf(x) + C*x; - return 0.2215f * (y*fabsf(y) - y) + y; -} - -static -void AndroidQuakeSoundCallback(int event, void* user, void *info) { - - if (event != AudioTrack::EVENT_MORE_DATA) return; - - const AudioTrack::Buffer *buffer = static_cast<const AudioTrack::Buffer *>(info); - size_t bytesToCopy = buffer->size; - size_t framesToCopy = buffer->size / (BYTES_PER_SAMPLE * CHANNEL_COUNT); - short* pData = buffer->i16; - - for(size_t frame = 0; frame < framesToCopy; frame++) { - short leftSample = (short) (32767.0f * sinef(left)); - left += p; - if (left > 2*M_PI) { - left -= 2*M_PI; - } - pData[frame * CHANNEL_COUNT] = leftSample; - - short rightSample = (short) (32767.0f * sinef(right)); - right += 2 * p; - if (right > 2*M_PI) { - right -= 2*M_PI; - } - pData[1 + frame * CHANNEL_COUNT] = rightSample; - } - - gDMAByteIndex = mod(gDMAByteIndex + bytesToCopy, TOTAL_BUFFER_SIZE); - asm volatile ("":::"memory"); -} - -#endif - -#ifdef NORMAL_SOUND - -static bool gWaitingForMixerToRestart; - -// Assumes the mutex is acquired. -// Waits until audio is available or a time period has elapsed. -static bool shouldMixSilence() { - if (!gSoundMixingStarted) { - return true; - } - while (gAvailableBytes == 0) { - if (gWaitingForMixerToRestart) { - return true; - } - timeval tp; - if (gettimeofday(&tp, NULL)) { - return true; - } - const long WAIT_NS = 40 * 1000 * 1000; - const long NS_PER_SECOND = 1000 * 1000 * 1000; - timespec ts; - ts.tv_sec = tp.tv_sec; - ts.tv_nsec = tp.tv_usec * 1000 + WAIT_NS; - if (ts.tv_nsec >= NS_PER_SECOND) { - ts.tv_nsec -= NS_PER_SECOND; - ts.tv_sec += 1; - } - if (ETIMEDOUT == pthread_cond_timedwait( &condition_cond, &condition_mutex, &ts)) { - gWaitingForMixerToRestart = true; - return true; - } - } - gWaitingForMixerToRestart = false; - return false; -} - -static -void AndroidQuakeSoundCallback(int event, void* user, void *info) { - - if (event != AudioTrack::EVENT_MORE_DATA) return; - - const AudioTrack::Buffer *buffer = static_cast<const AudioTrack::Buffer *>(info); - size_t dmaByteIndex = gDMAByteIndex; - size_t size = buffer->size; - unsigned char* pDestBuffer = (unsigned char*) buffer->raw; - - if (size == 0) return; - - if ( ! shm ) { - memset(pDestBuffer, 0, size); - return; - } - - const unsigned char* pSrcBuffer = shm->buffer; - - while(size > 0) { - pthread_mutex_lock( &condition_mutex ); - - if (shouldMixSilence()) { - memset(pDestBuffer, 0, size); - pthread_mutex_unlock( &condition_mutex ); - return; - } - - size_t chunkSize = min(gAvailableBytes, min(TOTAL_BUFFER_SIZE-dmaByteIndex, size)); - gAvailableBytes -= chunkSize; - - pthread_mutex_unlock( &condition_mutex ); - - memcpy(pDestBuffer, pSrcBuffer + dmaByteIndex, chunkSize); - size -= chunkSize; - pDestBuffer += chunkSize; - dmaByteIndex += chunkSize; - if (dmaByteIndex >= TOTAL_BUFFER_SIZE) { - dmaByteIndex = 0; - } - } - - gDMAByteIndex = dmaByteIndex; - asm volatile ("":::"memory"); -} - -#endif - -qboolean SNDDMA_Init(void) -{ - if ( ! enableSound() ) { - return false; - } - - gDMAByteIndex = 0; - - // Initialize the AudioTrack. - - status_t result = gAudioTrack.set( - AUDIO_STREAM_DEFAULT, // stream type - SAMPLE_RATE, // sample rate - BITS_PER_SAMPLE == 16 ? AUDIO_FORMAT_PCM_16_BIT : AUDIO_FORMAT_PCM_8_BIT, // format (8 or 16) - (CHANNEL_COUNT > 1) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO, // channel mask - 0, // default buffer size - 0, // flags - AndroidQuakeSoundCallback, // callback - 0, // user - 0); // default notification size - - LOGI("AudioTrack status = %d (%s)\n", result, result == NO_ERROR ? "success" : "error"); - - if ( result == NO_ERROR ) { - LOGI("AudioTrack latency = %u ms\n", gAudioTrack.latency()); - LOGI("AudioTrack format = %u bits\n", gAudioTrack.format() == AUDIO_FORMAT_PCM_16_BIT ? 16 : 8); - LOGI("AudioTrack sample rate = %u Hz\n", gAudioTrack.getSampleRate()); - LOGI("AudioTrack frame count = %d\n", int(gAudioTrack.frameCount())); - LOGI("AudioTrack channel count = %d\n", gAudioTrack.channelCount()); - - // Initialize Quake's idea of a DMA buffer. - - shm = &sn; - memset((void*)&sn, 0, sizeof(sn)); - - shm->splitbuffer = false; // Not used. - shm->samplebits = gAudioTrack.format() == AUDIO_FORMAT_PCM_16_BIT ? 16 : 8; - shm->speed = gAudioTrack.getSampleRate(); - shm->channels = gAudioTrack.channelCount(); - shm->samples = TOTAL_BUFFER_SIZE / BYTES_PER_SAMPLE; - shm->samplepos = 0; // Not used. - shm->buffer = (unsigned char*) Hunk_AllocName(TOTAL_BUFFER_SIZE, (char*) "shmbuf"); - shm->submission_chunk = 1; // Not used. - - shm->soundalive = true; - - if ( (shm->samples & 0x1ff) != 0 ) { - LOGE("SNDDDMA_Init: samples must be power of two."); - return false; - } - - if ( shm->buffer == 0 ) { - LOGE("SNDDDMA_Init: Could not allocate sound buffer."); - return false; - } - - gAudioTrack.setVolume(1.0f, 1.0f); - gAudioTrack.start(); - } - - return result == NO_ERROR; -} - -/* -============== -SNDDMA_GetDMAPos - -return the current sample position (in mono samples read) -inside the recirculating dma buffer, so the mixing code will know -how many sample are required to fill it up. -=============== -*/ -int SNDDMA_GetDMAPos(void) -{ - int dmaPos = gDMAByteIndex / BYTES_PER_SAMPLE; - asm volatile ("":::"memory"); - return dmaPos; -} - -/* -=============== -SNDDMA_ReportWrite - -Report valid data being written into the DMA buffer by the sound mixing code. -This is an Android specific API. -================ -*/ -void SNDDMA_ReportWrite(size_t lengthBytes) { - pthread_mutex_lock( &condition_mutex ); - gSoundMixingStarted = true; - if (gAvailableBytes == 0) { - pthread_cond_signal( &condition_cond ); - } - gAvailableBytes += lengthBytes; - pthread_mutex_unlock( &condition_mutex ); -} - -/* -============== -SNDDMA_Submit - -Send sound to device if buffer isn't really the dma buffer -=============== -*/ -void SNDDMA_Submit(void) -{ -} - -/* -============== -SNDDMA_Shutdown - -Reset the sound device for exiting -=============== -*/ -void SNDDMA_Shutdown(void) -{ - gAudioTrack.stop(); -} - -#endif |