今天來看看playback rate相關的接口。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()
{
// 直接返回瞭audio_track_cblk_t中的sample rate。
// audio_track_cblk_t對象在AudioFlinger::ThreadBase::TrackBase::TrackBase的構造函數中被創建：new(mCblk) audio_track_cblk_t();
// 創建audio_track_cblk_t對象後，即對其成員變量sampleRate進行瞭賦值：mCblk->sampleRate = sampleRate;
// 此處的sampleRate其實是創建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}
*/
// 看看這段註釋
// 此處改變的rate，隻是播放時的rate，並不是數據本身的rate。
// 例如，如果將rate設置為原來的一半，則播放時間將變為原來的2倍。
// 所設rate的范圍是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中添加成員
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設置到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();
}
###########################################################
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set rate改變的隻是播放時的rate，而不是數據本身的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。
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摘自：江風的專欄
