Android Audio程式碼分析18 - setSampleRate函數

今天來看看playback rate相關的介面。包括set和get。

*****************************************源碼*************************************************
    //Test case 6: setPlaybackRate() accepts values twice the output sample rate
    @LargeTest
    public void testSetPlaybackRateTwiceOutputSR() throws Exception {
        // constants for test
        final String TEST_NAME = "testSetPlaybackRateTwiceOutputSR";
        final int TEST_SR = 22050;
        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
        final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
        final int TEST_MODE = AudioTrack.MODE_STREAM;
        final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
       
        //-------- initialization --------------
        int minBuffSize = AudioTrack.getMinBufferSize(TEST_SR, TEST_CONF, TEST_FORMAT);
        AudioTrack track = new AudioTrack(TEST_STREAM_TYPE, TEST_SR, TEST_CONF, TEST_FORMAT,
                minBuffSize, TEST_MODE);
        byte data[] = new byte[minBuffSize/2];
        int outputSR = AudioTrack.getNativeOutputSampleRate(TEST_STREAM_TYPE);
        //--------    test        --------------
        track.write(data, 0, data.length);
        track.write(data, 0, data.length);
        assumeTrue(TEST_NAME, track.getState() == AudioTrack.STATE_INITIALIZED);
        track.play();
        assertTrue(TEST_NAME, track.setPlaybackRate(2*outputSR) == AudioTrack.SUCCESS);
        //-------- tear down      --------------
        track.release();
    }
**********************************************************************************************
源碼路徑：
frameworks\base\media\tests\mediaframeworktest\src\com\android\mediaframeworktest\functional\MediaAudioTrackTest.java

#######################說明################################
    //Test case 6: setPlaybackRate() accepts values twice the output sample rate
    @LargeTest
    public void testSetPlaybackRateTwiceOutputSR() throws Exception {
        // constants for test
        final String TEST_NAME = "testSetPlaybackRateTwiceOutputSR";
        final int TEST_SR = 22050;
        final int TEST_CONF = AudioFormat.CHANNEL_OUT_STEREO;
        final int TEST_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
        final int TEST_MODE = AudioTrack.MODE_STREAM;
        final int TEST_STREAM_TYPE = AudioManager.STREAM_MUSIC;
       
        //-------- initialization --------------
        int minBuffSize = AudioTrack.getMinBufferSize(TEST_SR, TEST_CONF, TEST_FORMAT);
        AudioTrack track = new AudioTrack(TEST_STREAM_TYPE, TEST_SR, TEST_CONF, TEST_FORMAT,
                minBuffSize, TEST_MODE);
        byte data[] = new byte[minBuffSize/2];
        int outputSR = AudioTrack.getNativeOutputSampleRate(TEST_STREAM_TYPE);
+++++++++++++++++++++++++++++getNativeOutputSampleRate+++++++++++++++++++++++++++++++++++
    /**
     *  Returns the hardware output sample rate
     */
    static public int getNativeOutputSampleRate(int streamType) {
        return native_get_output_sample_rate(streamType);
+++++++++++++++++++++++++++++android_media_AudioTrack_get_playback_rate+++++++++++++++++++++++++++++++++++
static jint android_media_AudioTrack_get_playback_rate(JNIEnv *env,  jobject thiz) {
    AudioTrack *lpTrack = (AudioTrack *)env->GetIntField(
                thiz, javaAudioTrackFields.nativeTrackInJavaObj);

    if (lpTrack) {
        return (jint) lpTrack->getSampleRate();  
++++++++++++++++++++++++++++AudioTrack::getSampleRate++++++++++++++++++++++++++++++++++++
uint32_t AudioTrack::getSampleRate()
{
// 直接返回了audio_track_cblk_t中的sample rate。
// audio_track_cblk_t對象在AudioFlinger::ThreadBase::TrackBase::TrackBase的建構函式中被建立：new(mCblk) audio_track_cblk_t();
// 建立audio_track_cblk_t對象後，即對其成員變數sampleRate進行了賦值：mCblk->sampleRate = sampleRate;
// 此處的sampleRate其實是建立AudioTrack對象時傳入的sampleRate。
    return mCblk->sampleRate;
}
----------------------------AudioTrack::getSampleRate------------------------------------
    } else {
        jniThrowException(env, "java/lang/IllegalStateException",
            "Unable to retrieve AudioTrack pointer for getSampleRate()");
        return AUDIOTRACK_ERROR;
    }
}
-----------------------------android_media_AudioTrack_get_playback_rate-----------------------------------
    }
-----------------------------getNativeOutputSampleRate-----------------------------------
        //--------    test        --------------
        track.write(data, 0, data.length);
        track.write(data, 0, data.length);
        assumeTrue(TEST_NAME, track.getState() == AudioTrack.STATE_INITIALIZED);
        track.play();
        assertTrue(TEST_NAME, track.setPlaybackRate(2*outputSR) == AudioTrack.SUCCESS);
++++++++++++++++++++++++++++setPlaybackRate+++++++++++++++++++++++++++++++++++
    /**
     * Sets the playback sample rate for this track. This sets the sampling rate at which
     * the audio data will be consumed and played back, not the original sampling rate of the
     * content. Setting it to half the sample rate of the content will cause the playback to
     * last twice as long, but will also result in a negative pitch shift.
     * The valid sample rate range if from 1Hz to twice the value returned by
     * {@link #getNativeOutputSampleRate(int)}.
     * @param sampleRateInHz the sample rate expressed in Hz
     * @return error code or success, see {@link #SUCCESS}, {@link #ERROR_BAD_VALUE},
     *    {@link #ERROR_INVALID_OPERATION}
     */
// 看看這段注釋
// 此處改變的rate，只是播放時的rate，並不是資料本身的rate。
// 例如，如果將rate設定為原來的一半，則播放時間將變為原來的2倍。
// 所設rate的範圍是1Hz到原來rate的2倍。
    public int setPlaybackRate(int sampleRateInHz) {
        if (mState != STATE_INITIALIZED) {
            return ERROR_INVALID_OPERATION;
        }
        if (sampleRateInHz <= 0) {
            return ERROR_BAD_VALUE;
        }
        return native_set_playback_rate(sampleRateInHz);
++++++++++++++++++++++++++++android_media_AudioTrack_set_playback_rate++++++++++++++++++++++++++++++++++++
static jint android_media_AudioTrack_set_playback_rate(JNIEnv *env,  jobject thiz,
        jint sampleRateInHz) {
    AudioTrack *lpTrack = (AudioTrack *)env->GetIntField(
                thiz, javaAudioTrackFields.nativeTrackInJavaObj);

    if (lpTrack) {
        return android_media_translateErrorCode(lpTrack->setSampleRate(sampleRateInHz));
+++++++++++++++++++++++++++AudioTrack::setSampleRate+++++++++++++++++++++++++++++++++++++
status_t AudioTrack::setSampleRate(int rate)
{
    int afSamplingRate;

    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
        return NO_INIT;
+++++++++++++++++++++++++++++AudioSystem::getOutputSamplingRate+++++++++++++++++++++++++++++++++++
// 感覺這麼熟悉！
// 原來已見過多次！
status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
{
    OutputDescriptor *outputDesc;
    audio_io_handle_t output;

    if (streamType == DEFAULT) {
        streamType = MUSIC;
    }

    output = getOutput((stream_type)streamType);
    if (output == 0) {
        return PERMISSION_DENIED;
    }

    gLock.lock();
// AudioSystem::AudioFlingerClient::ioConfigChanged函數有往gOutputs中新增成員
    outputDesc = AudioSystem::gOutputs.valueFor(output);
    if (outputDesc == 0) {
        LOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
        gLock.unlock();
        const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
        if (af == 0) return PERMISSION_DENIED;
        *samplingRate = af->sampleRate(output);
+++++++++++++++++++++++++AudioFlinger::sampleRate+++++++++++++++++++++++++++++++++++++++
uint32_t AudioFlinger::sampleRate(int output) const
{
    Mutex::Autolock _l(mLock);
    PlaybackThread *thread = checkPlaybackThread_l(output);
    if (thread == NULL) {
        LOGW("sampleRate() unknown thread %d", output);
        return 0;
    }
    return thread->sampleRate();
+++++++++++++++++++++++++++AudioFlinger::ThreadBase::sampleRate+++++++++++++++++++++++++++++++++++++
uint32_t AudioFlinger::ThreadBase::sampleRate() const
{
// 函數AudioFlinger::PlaybackThread::readOutputParameters中會給mSampleRate賦值： mSampleRate = mOutput->sampleRate();
    return mSampleRate;
}
---------------------------AudioFlinger::ThreadBase::sampleRate-------------------------------------
}
-------------------------AudioFlinger::sampleRate---------------------------------------
    } else {
        LOGV("getOutputSamplingRate() reading from output desc");
        *samplingRate = outputDesc->samplingRate;
        gLock.unlock();
    }

    LOGV("getOutputSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, *samplingRate);

    return NO_ERROR;
}
-----------------------------AudioSystem::getOutputSamplingRate-----------------------------------
    }
    // Resampler implementation limits input sampling rate to 2 x output sampling rate.
    if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;

// 將rate設定到audio_track_cblk_t對象中
    mCblk->sampleRate = rate;
    return NO_ERROR;
}
---------------------------AudioTrack::setSampleRate-------------------------------------
    } else {
        jniThrowException(env, "java/lang/IllegalStateException",
            "Unable to retrieve AudioTrack pointer for setSampleRate()");
        return AUDIOTRACK_ERROR;
    }
}
----------------------------android_media_AudioTrack_set_playback_rate------------------------------------
    }
----------------------------setPlaybackRate------------------------------------
        //-------- tear down      --------------
        track.release();
    }
###########################################################

&&&&&&&&&&&&&&&&&&&&&&&總結&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
set rate改變的只是播放時的rate，而不是資料本身的rate。
也就是說，set rate若與原來的rate不同的話，播放時間會改變。
函數AudioFlinger::MixerThread::threadLoop中會根據rate計算max period。
&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&

摘自：江風的專欄