Dear Glenn Kasten, We are concern audio low latency of Android 4.2 Compatibility Definition of (android-4.2-cdd), which make a requirement: * 1. cold output latency of 100 milliseconds or less* * 2. continuous output latency of 45 milliseconds or less* In the document, *"If a device implementation meets the requirements of this section after any initial calibration when using the* *OpenSL ES PCM buffer queue API"*, we have a question of this definition, what is the state of opensles Audioplayer when we start the stopwatch? If there is a background sound playing, we should start the stopwatch from *"enable an audioplayer"* or from *"enqueue buffer in callback function".*
As attachment (google opensles example), we should start audio latency measurement from line #264, or from line #95? Thank you! Sincerely, Glenn Kasten於 2012年9月7日星期五UTC+8下午11時44分29秒寫道： > > 1. You didn't mention if you're developing Android apps or the platform. > If you're an Android app developer, you should be using only documented > public APIs. For audio output, that's Java language > android.media.AudioTrack in SDK and C language OpenSL ES AudioPlayer with > PCM buffer queue in NDK. The AUDIO_OUTPUT_FLAG_FAST is an internal symbol > that's used only at the AudioTrack C++ level, and that's not a documented > public API. So you should not need to deal with AUDIO_OUTPUT_FLAG_FAST. > > But if you're doing platform development such as porting, it can be > helpful to understand the internal implementation in JB ... > AUDIO_OUTPUT_FLAG_FAST is a hint from the API level that this application > would like to use a lower latency, fewer feature, audio track if one is > available. The request is not guaranteed to be accepted by the audio > server (AudioFlinger). The fewer features that are not available include > effects, as you said, and also sample rate conversion. If AudioFlinger > can handle the request it will create a "fast track", otherwise a normal > track. > > 2. The "fast" in FastMixer means that it executes more often, and that it > uses less CPU time each time it runs, than the normal mixer thread. The > normal mixer thread runs about once every 20 ms, and the FastMixer thread > runs at rate of once per HAL buffer (which is ideally less than 20 ms). The > FastMixer thread supports up to 7 fast tracks, and does not support sample > rate conversion of effects. So it uses a limited amount of CPU each time it > runs. The normal mixer thread supports more tracks (up to 32), and supports > sample rate conversion and effects. So it can use more CPU each time it > runs. The main purpose of FastMixer design was not to take advantage of > multi-core. > > On Tuesday, September 4, 2012 6:59:27 PM UTC-7, big_fish_ wrote: >> >> I am a android developer, I just read the FastMixer code of Jellybean. >> >> I have some questions, >> >> 1, If submit AudioTrack with AUDIO_OUTPUT_FLAG_FAST flag, then this Track >> can't do AudioEffect handle, right? >> >> I noticed that FastMixer thread handle all FastTacks without >> AudioEffect. Except mFastTracks[0], because the zero FastTrack is passed >> from MixerThread which was already through mixer and effect handled. right? >> >> 2, About the performance, why FastMixer is faster then before? >> >> If we have 20 tracks, we set 8 tracks as FastMixer, and 12 as normal tracks, >> then there are two threads to do mixer. So if we run on dual core CPU, then >> we have multithreading adventage. >> >> But if we have 32 tracks are all as FastTrack, then MixerThread will not >> do mixer. then there will have no multithreading adventage. >> >> >> >> -- -- unsubscribe: [email protected] website: http://groups.google.com/group/android-porting --- You received this message because you are subscribed to the Google Groups "android-porting" group. To unsubscribe from this group and stop receiving emails from it, send an email to [email protected]. For more options, visit https://groups.google.com/groups/opt_out.
/* * Copyright (C) 2010 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ /* * Copyright (c) 2009 The Khronos Group Inc. * * Permission is hereby granted, free of charge, to any person obtaining a copy of this * software and /or associated documentation files (the "Materials "), to deal in the * Materials without restriction, including without limitation the rights to use, copy, * modify, merge, publish, distribute, sublicense, and/or sell copies of the Materials, * and to permit persons to whom the Materials are furnished to do so, subject to * the following conditions: * * The above copyright notice and this permission notice shall be included * in all copies or substantial portions of the Materials. * * THE MATERIALS ARE PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS * BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN * ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN * CONNECTION WITH THE MATERIALS OR THE USE OR OTHER DEALINGS IN THE * MATERIALS. */ #include <stdlib.h> #include <stdio.h> #include <string.h> #include <unistd.h> #include <sys/time.h> #include <SLES/OpenSLES.h> #define MAX_NUMBER_INTERFACES 3 /* Local storage for Audio data in 16 bit words */ #define AUDIO_DATA_STORAGE_SIZE 4096 * 100 /* Audio data buffer size in 16 bit words. 8 data segments are used in this simple example */ #define AUDIO_DATA_BUFFER_SIZE 4096/8 /* 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)) { //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"); } /* Play some audio from a buffer queue */ void TestPlaySawtoothBufferQueue( SLObjectItf sl ) { SLEngineItf EngineItf; SLint32 numOutputs = 0; SLuint32 deviceID = 0; SLresult res; SLDataSource audioSource; SLDataLocator_BufferQueue bufferQueue; SLDataFormat_PCM pcm; SLDataSink audioSink; SLDataLocator_OutputMix locator_outputmix; SLObjectItf player; SLPlayItf playItf; SLBufferQueueItf bufferQueueItf; SLBufferQueueState state; SLObjectItf OutputMix; SLVolumeItf volumeItf; int i; SLboolean required[MAX_NUMBER_INTERFACES]; SLInterfaceID iidArray[MAX_NUMBER_INTERFACES]; /* Callback context for the buffer queue callback function */ CallbackCntxt cntxt; /* 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 = 1;//2; pcm.samplesPerSec = SL_SAMPLINGRATE_44_1; 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 play */ unsigned int j; for (j = 0; j < sizeof(pcmData)/sizeof(pcmData[0]); ++j) { pcmData[j] = j*(100 + j / 200);// % 1000; } /* 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); // while (state.playIndex < 100) { while (state.count) { usleep(10000); (*bufferQueueItf)->GetState(bufferQueueItf, &state); } /* Make sure player is stopped */ res = (*playItf)->SetPlayState(playItf, SL_PLAYSTATE_STOPPED); CheckErr(res); /* Destroy the player */ (*player)->Destroy(player); /* Destroy Output Mix object */ (*OutputMix)->Destroy(OutputMix); } int main(int argc, char* const argv[]) { SLresult res; SLObjectItf sl; SLEngineOption EngineOption[] = { {(SLuint32) SL_ENGINEOPTION_THREADSAFE, (SLuint32) SL_BOOLEAN_TRUE}}; res = slCreateEngine( &sl, 1, EngineOption, 0, NULL, NULL); CheckErr(res); /* Realizing the SL Engine in synchronous mode. */ res = (*sl)->Realize(sl, SL_BOOLEAN_FALSE); CheckErr(res); /* Run the test */ TestPlaySawtoothBufferQueue(sl); /* Shutdown OpenSL ES */ (*sl)->Destroy(sl); return EXIT_SUCCESS; }
