1 /* 2 * Copyright (C) 2010 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 // Test various buffer queue configurations 18 19 #include <assert.h> 20 #include <math.h> 21 #include <stdio.h> 22 #include <stdlib.h> 23 #include <unistd.h> 24 25 #include <SLES/OpenSLES.h> 26 27 typedef struct { 28 SLuint8 numChannels; 29 SLuint32 milliHz; 30 SLuint8 bitsPerSample; 31 } PCM; 32 33 PCM formats[] = { 34 {1, SL_SAMPLINGRATE_8, 8}, 35 {2, SL_SAMPLINGRATE_8, 8}, 36 {1, SL_SAMPLINGRATE_8, 16}, 37 {2, SL_SAMPLINGRATE_8, 16}, 38 {1, SL_SAMPLINGRATE_11_025, 8}, 39 {2, SL_SAMPLINGRATE_11_025, 8}, 40 {1, SL_SAMPLINGRATE_11_025, 16}, 41 {2, SL_SAMPLINGRATE_11_025, 16}, 42 {1, SL_SAMPLINGRATE_12, 8}, 43 {2, SL_SAMPLINGRATE_12, 8}, 44 {1, SL_SAMPLINGRATE_12, 16}, 45 {2, SL_SAMPLINGRATE_12, 16}, 46 {1, SL_SAMPLINGRATE_16, 8}, 47 {2, SL_SAMPLINGRATE_16, 8}, 48 {1, SL_SAMPLINGRATE_16, 16}, 49 {2, SL_SAMPLINGRATE_16, 16}, 50 {1, SL_SAMPLINGRATE_22_05, 8}, 51 {2, SL_SAMPLINGRATE_22_05, 8}, 52 {1, SL_SAMPLINGRATE_22_05, 16}, 53 {2, SL_SAMPLINGRATE_22_05, 16}, 54 {1, SL_SAMPLINGRATE_24, 8}, 55 {2, SL_SAMPLINGRATE_24, 8}, 56 {1, SL_SAMPLINGRATE_24, 16}, 57 {2, SL_SAMPLINGRATE_24, 16}, 58 {1, SL_SAMPLINGRATE_32, 8}, 59 {2, SL_SAMPLINGRATE_32, 8}, 60 {1, SL_SAMPLINGRATE_32, 16}, 61 {2, SL_SAMPLINGRATE_32, 16}, 62 {1, SL_SAMPLINGRATE_44_1, 8}, 63 {2, SL_SAMPLINGRATE_44_1, 8}, 64 {1, SL_SAMPLINGRATE_44_1, 16}, 65 {2, SL_SAMPLINGRATE_44_1, 16}, 66 {1, SL_SAMPLINGRATE_48, 8}, 67 {2, SL_SAMPLINGRATE_48, 8}, 68 {1, SL_SAMPLINGRATE_48, 16}, 69 {2, SL_SAMPLINGRATE_48, 16}, 70 {0, 0, 0} 71 }; 72 73 int main(int argc, char **argv) 74 { 75 SLresult result; 76 SLObjectItf engineObject; 77 78 // create engine 79 result = slCreateEngine(&engineObject, 0, NULL, 0, NULL, NULL); 80 assert(SL_RESULT_SUCCESS == result); 81 SLEngineItf engineEngine; 82 result = (*engineObject)->Realize(engineObject, SL_BOOLEAN_FALSE); 83 assert(SL_RESULT_SUCCESS == result); 84 result = (*engineObject)->GetInterface(engineObject, SL_IID_ENGINE, &engineEngine); 85 assert(SL_RESULT_SUCCESS == result); 86 87 // create output mix 88 SLObjectItf outputMixObject; 89 result = (*engineEngine)->CreateOutputMix(engineEngine, &outputMixObject, 0, NULL, NULL); 90 assert(SL_RESULT_SUCCESS == result); 91 result = (*outputMixObject)->Realize(outputMixObject, SL_BOOLEAN_FALSE); 92 assert(SL_RESULT_SUCCESS == result); 93 94 // loop over all formats 95 PCM *format; 96 float hzLeft = 440.0; // A440 (Concert A) 97 float hzRight = 440.0; 98 for (format = formats; format->numChannels; ++format) { 99 100 printf("Channels: %d, sample rate: %u, bits: %u\n", format->numChannels, 101 format->milliHz / 1000, format->bitsPerSample); 102 103 // configure audio source 104 SLDataLocator_BufferQueue loc_bufq; 105 loc_bufq.locatorType = SL_DATALOCATOR_BUFFERQUEUE; 106 loc_bufq.numBuffers = 1; 107 SLDataFormat_PCM format_pcm; 108 format_pcm.formatType = SL_DATAFORMAT_PCM; 109 format_pcm.numChannels = format->numChannels; 110 format_pcm.samplesPerSec = format->milliHz; 111 format_pcm.bitsPerSample = format->bitsPerSample; 112 format_pcm.containerSize = format->bitsPerSample; 113 format_pcm.channelMask = 0; 114 format_pcm.endianness = SL_BYTEORDER_LITTLEENDIAN; 115 SLDataSource audioSrc; 116 audioSrc.pLocator = &loc_bufq; 117 audioSrc.pFormat = &format_pcm; 118 119 // configure audio sink 120 SLDataLocator_OutputMix loc_outmix; 121 loc_outmix.locatorType = SL_DATALOCATOR_OUTPUTMIX; 122 loc_outmix.outputMix = outputMixObject; 123 SLDataSink audioSnk; 124 audioSnk.pLocator = &loc_outmix; 125 audioSnk.pFormat = NULL; 126 127 // create audio player 128 SLuint32 numInterfaces = 1; 129 SLInterfaceID ids[1]; 130 SLboolean req[1]; 131 ids[0] = SL_IID_BUFFERQUEUE; 132 req[0] = SL_BOOLEAN_TRUE; 133 SLObjectItf playerObject; 134 result = (*engineEngine)->CreateAudioPlayer(engineEngine, &playerObject, &audioSrc, 135 &audioSnk, numInterfaces, ids, req); 136 if (SL_RESULT_SUCCESS != result) { 137 printf("failed %u\n", result); 138 continue; 139 } 140 141 // realize the player 142 result = (*playerObject)->Realize(playerObject, SL_BOOLEAN_FALSE); 143 assert(SL_RESULT_SUCCESS == result); 144 145 // generate a sine wave buffer, ascending in half-steps for each format 146 #define N (44100*4) 147 static unsigned char buffer[N]; 148 unsigned i; 149 for (i = 0; i < N; ) { 150 float seconds = (((i * 8) / (format->bitsPerSample * format->numChannels)) * 1000.0) / 151 format->milliHz; 152 short sampleLeft = sin(seconds * M_PI_2 * hzLeft) * 32767.0; 153 short sampleRight = sin(seconds * M_PI_2 * hzRight) * 32767.0; 154 if (2 == format->numChannels) { 155 if (8 == format->bitsPerSample) { 156 buffer[i++] = (sampleLeft + 32768) >> 8; 157 buffer[i++] = (sampleRight + 32768) >> 8; 158 } else { 159 assert(16 == format->bitsPerSample); 160 buffer[i++] = sampleLeft & 0xFF; 161 buffer[i++] = sampleLeft >> 8; 162 buffer[i++] = sampleRight & 0xFF; 163 buffer[i++] = sampleRight >> 8; 164 } 165 } else { 166 assert(1 == format->numChannels); 167 // cast to int and divide by 2 are needed to prevent overflow 168 short sampleMono = ((int) sampleLeft + (int) sampleRight) / 2; 169 if (8 == format->bitsPerSample) { 170 buffer[i++] = (sampleMono + 32768) >> 8; 171 } else { 172 assert(16 == format->bitsPerSample); 173 buffer[i++] = sampleMono & 0xFF; 174 buffer[i++] = sampleMono >> 8; 175 } 176 } 177 if (seconds >= 1.0f) 178 break; 179 } 180 181 // get the buffer queue interface and enqueue a buffer 182 SLBufferQueueItf playerBufferQueue; 183 result = (*playerObject)->GetInterface(playerObject, SL_IID_BUFFERQUEUE, 184 &playerBufferQueue); 185 assert(SL_RESULT_SUCCESS == result); 186 result = (*playerBufferQueue)->Enqueue(playerBufferQueue, buffer, i); 187 assert(SL_RESULT_SUCCESS == result); 188 189 // get the play interface 190 SLPlayItf playerPlay; 191 result = (*playerObject)->GetInterface(playerObject, SL_IID_PLAY, &playerPlay); 192 assert(SL_RESULT_SUCCESS == result); 193 194 // set the player's state to playing 195 result = (*playerPlay)->SetPlayState(playerPlay, SL_PLAYSTATE_PLAYING); 196 assert(SL_RESULT_SUCCESS == result); 197 198 // wait for the buffer to be played 199 for (;;) { 200 SLBufferQueueState state; 201 result = (*playerBufferQueue)->GetState(playerBufferQueue, &state); 202 assert(SL_RESULT_SUCCESS == result); 203 if (state.count == 0) 204 break; 205 usleep(20000); 206 } 207 208 // destroy audio player 209 (*playerObject)->Destroy(playerObject); 210 211 //usleep(1000000); 212 hzLeft *= 1.05946309; // twelfth root of 2 213 hzRight /= 1.05946309; 214 } 215 216 // destroy output mix 217 (*outputMixObject)->Destroy(outputMixObject); 218 219 // destroy engine 220 (*engineObject)->Destroy(engineObject); 221 222 return EXIT_SUCCESS; 223 } 224
