OpenSL ES无需重新创建音频播放器即可更改音频源

Question

I have a layout that has about 60 buttons and each one, when pressed, plays a different audio file. I have all my audio files as mp3s in my assets folder and to play them I'm basically using the same code as is used in the Google NDK samples "native-audio" project: https://github.com/googlesamples/android-ndk

I have 10 identical native functions (just with uniquely named variables) that work like this..

function to play sound:

jboolean Java_com_example_nativeaudio_Fretboard_player7play(JNIEnv* env, jclass clazz, jobject assetManager, jstring filename)
{
    SLresult result;

    // convert Java string to UTF-8
    const char *utf8 = (*env)->GetStringUTFChars(env, filename, NULL);
    assert(NULL != utf8);
    // use asset manager to open asset by filename
    AAssetManager* mgr = AAssetManager_fromJava(env, assetManager);
    assert(NULL != mgr);
    AAsset* asset = AAssetManager_open(mgr, utf8, AASSET_MODE_UNKNOWN);
    // release the Java string and UTF-8
    (*env)->ReleaseStringUTFChars(env, filename, utf8);
    // the asset might not be found
    if (NULL == asset) {
        return JNI_FALSE;
    }
    // open asset as file descriptor
    off_t start, length;
    int fd = AAsset_openFileDescriptor(asset, &start, &length);
    assert(0 <= fd);
    AAsset_close(asset);

    // configure audio source
    SLDataLocator_AndroidFD loc_fd = {SL_DATALOCATOR_ANDROIDFD, fd, start, length};
    SLDataFormat_MIME format_mime = {SL_DATAFORMAT_MIME, NULL, SL_CONTAINERTYPE_UNSPECIFIED};
    SLDataSource audioSrc = {&loc_fd, &format_mime};
    // configure audio sink
    SLDataLocator_OutputMix loc_outmix = {SL_DATALOCATOR_OUTPUTMIX, outputMixObject};
    SLDataSink audioSnk = {&loc_outmix, NULL};
    // create audio player
    const SLInterfaceID ids[3] = {SL_IID_SEEK, SL_IID_MUTESOLO, SL_IID_VOLUME};
    const SLboolean req[3] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
    result = (*engineEngine)->CreateAudioPlayer(engineEngine, &p7PlayerObject, &audioSrc, &audioSnk,
                                                3, ids, req);
    assert(SL_RESULT_SUCCESS == result);
    (void)result;
    // realize the player
    result = (*p7PlayerObject)->Realize(p7PlayerObject, SL_BOOLEAN_FALSE);
    assert(SL_RESULT_SUCCESS == result);
    (void)result;
    // get the play interface
    result = (*p7PlayerObject)->GetInterface(p7PlayerObject, SL_IID_PLAY, &p7PlayerPlay);
    assert(SL_RESULT_SUCCESS == result);
    (void)result;

    if (NULL != p7PlayerPlay) {
        // play
        result = (*p7PlayerPlay)->SetPlayState(p7PlayerPlay, SL_PLAYSTATE_PLAYING);
        assert(SL_RESULT_SUCCESS == result);
        (void)result;
    }

    return JNI_TRUE;
}

function to stop that sound:

void Java_com_example_nativeaudio_Fretboard_player7stop(JNIEnv* env, jclass clazz)
{
    SLresult result;

    // make sure the asset audio player was created
    if (NULL != p7PlayerPlay) {
        // set the player's state
        result = (*p7PlayerPlay)->SetPlayState(p7PlayerPlay, SL_PLAYSTATE_STOPPED);
        assert(SL_RESULT_SUCCESS == result);
        (void)result;
        // destroy file descriptor audio player object, and invalidate all associated interfaces
        (*p7PlayerObject)->Destroy(p7PlayerObject);
        p7PlayerObject = NULL;
        p7PlayerPlay = NULL;
    }
}

this is easy to deal with, but I want to minimize latency and avoid having to do (*engineEngine)->CreateAudioPlayer() every time I want to play a different file. Is there any way to just change the audioSrc used by the audio player without having to destroy and recreate it from scratch every time?

As a bonus, where can I read more about this stuff? Seems pretty difficult to find any information on OpenSL ES anywhere.

Answer 1

We're in the same boat, I'm currently familiarizing myself too with the NDK and OpenSL ES. My answer is based on my experience entirely consisting of ~2 days of experimentation so there might be better approaches but the information might help you on your way.

I have 10 identical native functions (just with uniquely named variables) that work like this..

If I understood your case correctly, you don't need to have duplicate functions for this. The only thing which differs in these calls is the button pressed and ultimately the sound to play and this can be passed as parameters through the JNI call. You can store the created player and data in a globally accessible structure so you can retrieve it when you need to stop/replay it, maybe using the buttonId as a key to a map.

[..]but I want to minimize latency and avoid having to do (*engineEngine)->CreateAudioPlayer() every time I want to play a different file. Is there any way to just change the audioSrc used by the audio player without having to destroy and recreate it from scratch every time?

Yes, constantly creating and destroying players is costly and can lead to fragmentation of the heap (as stated in the OpenSL ES 1.0 Specification). First, I thought he DynamicSourceItf would allow you to switch data sources but it seems that this interface is not intended to be used like that, at least on Android 6 this returns 'feature unsupported'.

I doubt that creating a player for each unique sound would be a good solution especially since playing the same sound multiple times on top of each other (as it's common in a game for example) would require an arbitrary amount of additional players for that same sound.

Buffer Queues

BufferQueues are queues of individual buffers which a player will process when playing. When all the buffers have been processed, the player 'stops' (it's official state is still 'playing' though) but will resume as soon as new buffers are being enqueued.

What this allows you to do is to create as many players as overlapping sounds you require. When you want to play a sound, you iterate over these players until you've found one which is not currently processing buffers ( BufferQueueItf->GetState(...) provides this information or a callback can be registered so you can tag players as being 'free'). Then, you enqueue as many buffers as your sound needs which will start playing immediately.

The format of a BufferQueue is, as far as I know, locked at creation. So you have to make sure that you either have all your input buffers in the same format or you create different BufferQueue (and players) for each format.

Android Simple BufferQueue

According to the Android NDK documentation, the BufferQueue interface is expected to have significant changes in the future. They have extracted a simplified interface with most of BufferQueue's functionality and called it AndroidSimpleBufferQueue. This interface is not expected to change and thus makes your code more future proof.

The main functionality you loose by using the AndroidSimpleBufferQueue is to be able to use non-PCM source data, so you'd have to decode your files before use. This can be done in OpenSL ES using a AndroidSimpleBufferQueue as a sink. More recent APIs have additional support using the MediaCodec and it's NDK implementation NDKMedia (checkout the native-codec example).

Resources

The NDK documentation does contain some important information which are hard to find anywhere else. Here's the OpenSL ES specific page.

It might be close to 600 pages and hard to digest, but the OpenSL ES 1.0 Specification should be your primary resource of information. I highly recommend reading chapter 4 as it gives a good overview of how things work. Chapter 3 has a bit more information on the specific design. Then, I just jump around using the search function to read up on interfaces and objects as I go.

Understanding OpenSL ES

Once you have understood the basic principles of how OpenSL works, it seems to be quite straightforward. There are media objects (players and recorders, etc) and data sources (inputs) and data sinks (outputs). You essentially connect an input to a media object which routes the processed data to its connected output.

Sources, Sinks and Media Objects are all documented in the specification including their interfaces. With that information, it really is just about picking the building blocks you require and plugging them together.

Update 07/29/16

From my tests, it seems as if both BufferQueue and AndroidSimpleBufferQueue do not support non-PCM data, at least not on the systems I've tested (Nexus 7 @ 6.01, NVidia Shield K1 @ 6.0.1) so you will need to decode your data before you can use it.

I tried using the NDK versions of the MediaExtractor and MediaCodec but there are several caveats to watch out for:

• MediaExtractor does not seem to correctly return the UUID information required for decoding with crypto, at least not for the files I've tested. AMediaExtractor_getPsshInfo returns a nullptr .

• The API does not always behave as the comments in the header claim. Checking for EOS (end of stream) in the MediaExtractor for example seems to be most reliable by checking the amount of bytes returned instead of checking the AMediaExtractor_advance function's return value.

I'd recommend staying in Java for the decoding process as these APIs are more mature, definitely more tested and you might get more functionality out of it. Once you have the buffers of raw PCM data, you can pass it to native code which allows you to reduce latency.