Android Audio代码分析18 - setSampleRate函数

本文详细解读了Android系统中AudioTrack类中setPlaybackRate方法的内部实现,包括初始化配置、测试流程、以及关键代码解析。着重解释了如何通过该方法调整音频播放速率,以及它对实际播放效果的影响。通过分析AudioTrack的内部工作原理,读者可以更好地理解Android音频系统的运行机制。

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今天来看看playback rate相关的接口。包括set和get。


*****************************************源码*************************************************
//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();
}
**********************************************************************************************
源码路径:
frameworks\base\media\tests\mediaframeworktest\src\com\android\mediaframeworktest\functional\MediaAudioTrackTest.java


#######################说明################################
//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对象在AudioFlinger::ThreadBase::TrackBase::TrackBase的构造函数中被创建:new(mCblk) audio_track_cblk_t();
// 创建audio_track_cblk_t对象后,即对其成员变量sampleRate进行了赋值:mCblk->sampleRate = sampleRate;
// 此处的sampleRate其实是创建AudioTrack对象时传入的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}
*/
// 看看这段注释
// 此处改变的rate,只是播放时的rate,并不是数据本身的rate。
// 例如,如果将rate设置为原来的一半,则播放时间将变为原来的2倍。
// 所设rate的范围是1Hz到原来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+++++++++++++++++++++++++++++++++++
// 感觉这么熟悉!
// 原来已见过多次!
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函数有往gOutputs中添加成员
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
{
// 函数AudioFlinger::PlaybackThread::readOutputParameters中会给mSampleRate赋值: 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;


// 将rate设置到audio_track_cblk_t对象中
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();
}
###########################################################


&&&&&&&&&&&&&&&&&&&&&&&总结&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
set rate改变的只是播放时的rate,而不是数据本身的rate。
也就是说,set rate若与原来的rate不同的话,播放时间会改变。
函数AudioFlinger::MixerThread::threadLoop中会根据rate计算max period。
&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&
### 配置AudioRecord使用特定输入设备 在Android平台中,`AudioRecord`类用于从音频输入硬件捕获原始音频数据。为了配置`AudioRecord`以使用特定的输入设备,可以利用`AudioAttributes.Builder`和`AudioFormat.Builder`来设定录音属性。 #### 使用AudioAttributes和AudioFormat构建器 通过`AudioAttributes.Builder`能够指定媒体使用的场景以及内容类型,这对于优化音频处理路径非常重要。而`AudioFormat.Builder`则允许更精细地控制音频格式特性,比如采样率、声道掩码等。 ```java // 构建音频属性 AudioAttributes audioAttrs = new AudioAttributes.Builder() .setContentType(AudioAttributes.CONTENT_TYPE_MUSIC) .build(); // 设置音频格式 AudioFormat audioFormat = new AudioFormat.Builder() .setEncoding(AudioFormat.ENCODING_PCM_16BIT) .setSampleRate(sampleRateInHz) .setChannelMask(channelConfig) .build(); ``` #### 获取并选择合适的音频源 对于不同的应用场景可以选择不同类型的音频源(` AudioSource`),例如麦克风(MIC),远端蓝牙SCO连接(REMOTE_SUBMIX)或者其他可能存在的外部声卡等。这一步骤决定了实际的数据来源[^1]。 ```java int audioSource; if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M && useSpecificDevice) { // 对于API Level 23及以上版本支持更多选项 List<AudioDeviceInfo> devices = am.getDevices(AudioManager.GET_DEVICES_INPUTS); for (AudioDeviceInfo device : devices) { if (device.getType() == AudioDeviceInfo.TYPE_USB_DEVICE || device.getType() == AudioDeviceInfo.TYPE_BLUETOOTH_A2DP || device.getType() == AudioDeviceInfo.TYPE_WIRED_HEADSET) { int source = getBestMatchingSource(device); // 自定义函数获取最佳匹配的source ID break; } } } else { // 默认情况下采用MIC作为音频源 audioSource = MediaRecorder.AudioSource.MIC; } ``` #### 初始化AudioRecord对象 最后基于前面准备好的参数初始化`AudioRecord`实例: ```java audioRecord = new AudioRecord(audioSource, sampleRateInHz, channelConfig, audioFormat.getEncoding(), bufferSizeInBytes); ```
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