AudioTrack的使用示例中，用到瞭函數getMinBufferSize，今天把它倒出來，再嚼嚼。
*****************************************源碼*************************************************
static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
int channelCount = 0;
switch(channelConfig) {
case AudioFormat.CHANNEL_OUT_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
channelCount = 1;
break;
case AudioFormat.CHANNEL_OUT_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
channelCount = 2;
break;
default:
loge("getMinBufferSize(): Invalid channel configuration.");
return AudioTrack.ERROR_BAD_VALUE;
}
if ((audioFormat != AudioFormat.ENCODING_PCM_16BIT)
&& (audioFormat != AudioFormat.ENCODING_PCM_8BIT)) {
loge("getMinBufferSize(): Invalid audio format.");
return AudioTrack.ERROR_BAD_VALUE;
}
if ( (sampleRateInHz < 4000) || (sampleRateInHz > 48000) ) {
loge("getMinBufferSize(): " + sampleRateInHz +"Hz is not a supported sample rate.");
return AudioTrack.ERROR_BAD_VALUE;
}
int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
if ((size == -1) || (size == 0)) {
loge("getMinBufferSize(): error querying hardware");
return AudioTrack.ERROR;
}
else {
return size;
}
} www.aiwalls.com
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源碼路徑：
frameworks\base\media\java\android\media\AudioTrack.java
###########################################說明##############################################################
先把自帶的註釋拿來看看吧：
/**
* Returns the minimum buffer size required for the successful creation of an AudioTrack
* object to be created in the {@link #MODE_STREAM} mode. Note that this size doesn't
* guarantee a smooth playback under load, and higher values should be chosen according to
* the expected frequency at which the buffer will be refilled with additional data to play.
* @param sampleRateInHz the sample rate expressed in Hertz.
* @param channelConfig describes the configuration of the audio channels.
* See {@link AudioFormat#CHANNEL_OUT_MONO} and
* {@link AudioFormat#CHANNEL_OUT_STEREO}
* @param audioFormat the format in which the audio data is represented.
* See {@link AudioFormat#ENCODING_PCM_16BIT} and
* {@link AudioFormat#ENCODING_PCM_8BIT}
* @return {@link #ERROR_BAD_VALUE} if an invalid parameter was passed,
* or {@link #ERROR} if the implementation was unable to query the hardware for its output
* properties,
* or the minimum buffer size expressed in bytes.
*/
從註釋可以看出，通過該函數獲取的最小buffer size，隻是保證在MODE_STREAM模式下成功地創建一個AudioTrack對象。
並不能保證流暢地播放。
1、參數就不說瞭，可以參考上面註釋，上一篇文章中也有說。
2、定義瞭一個內部變量：
int channelCount = 0;
用來記錄聲道數量。
調用native函數native_get_min_buff_size時會用。
可見buffer size也是由native層來決定的。
3、接下來根據Channel類型，計算聲道數量：
switch(channelConfig) {
case AudioFormat.CHANNEL_OUT_MONO:
case AudioFormat.CHANNEL_CONFIGURATION_MONO:
channelCount = 1;
break;
case AudioFormat.CHANNEL_OUT_STEREO:
case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
channelCount = 2;
break;
default:
loge("getMinBufferSize(): Invalid channel configuration.");
return AudioTrack.ERROR_BAD_VALUE;
}
MONO都是1，Stereo的都是2。
不過，我們之前看過，Channel類型不止這幾種。有以下一堆呢：
public static final int CHANNEL_OUT_FRONT_LEFT = 0x4;
public static final int CHANNEL_OUT_FRONT_RIGHT = 0x8;
public static final int CHANNEL_OUT_FRONT_CENTER = 0x10;
public static final int CHANNEL_OUT_LOW_FREQUENCY = 0x20;
public static final int CHANNEL_OUT_BACK_LEFT = 0x40;
public static final int CHANNEL_OUT_BACK_RIGHT = 0x80;
public static final int CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100;
public static final int CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200;
public static final int CHANNEL_OUT_BACK_CENTER = 0x400;
public static final int CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT;
public static final int CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT);
public static final int CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT);
public static final int CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER);
public static final int CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT);
public static final int CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT |
CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT |
CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER);
並且，AudioFormat.CHANNEL_CONFIGURATION_MONO和AudioFormat.CHANNEL_CONFIGURATION_STEREO的定義還不包含在這一堆之中，而是在它們之前定義：
/** Mono audio configuration */
/** @deprecated use CHANNEL_OUT_MONO or CHANNEL_IN_MONO instead */
@Deprecated public static final int CHANNEL_CONFIGURATION_MONO = 2;
/** Stereo (2 channel) audio configuration */
/** @deprecated use CHANNEL_OUT_STEREO or CHANNEL_IN_STEREO instead */
@Deprecated public static final int CHANNEL_CONFIGURATION_STEREO = 3;
難道其他的Channel類型都不需要獲取這個min buffer size？？？
還是說，目前隻支持單聲道和雙聲道？？？
4、下面判斷音頻格式，即采樣點數據所占的bit數：
if ((audioFormat != AudioFormat.ENCODING_PCM_16BIT)
&& (audioFormat != AudioFormat.ENCODING_PCM_8BIT)) {
loge("getMinBufferSize(): Invalid audio format.");
return AudioTrack.ERROR_BAD_VALUE;
}
可見，隻支持16bit和8bit兩種。
5、判斷采用率：
if ( (sampleRateInHz < 4000) || (sampleRateInHz > 48000) ) {
loge("getMinBufferSize(): " + sampleRateInHz +"Hz is not a supported sample rate.");
return AudioTrack.ERROR_BAD_VALUE;
}
隻支持4000Hz到48000Hz之間。
6、接下來調到native中去：
int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
if ((size == -1) || (size == 0)) {
loge("getMinBufferSize(): error querying hardware");
return AudioTrack.ERROR;
}
else {
return size;
}
可見，真正幹活的是在native中，java層中隻是做些輔助操作。
通過前文中JNI的函數對照表，可知native_get_min_buff_size函數對應的是native中的android_media_AudioTrack_get_min_buff_size函數。
路徑：frameworks\base\core\jni\android_media_AudioTrack.cpp
函數android_media_AudioTrack_get_min_buff_size的實現：
// returns the minimum required size for the successful creation of a streaming AudioTrack
// returns -1 if there was an error querying the hardware.
static jint android_media_AudioTrack_get_min_buff_size(JNIEnv *env, jobject thiz,
jint sampleRateInHertz, jint nbChannels, jint audioFormat) {
int frameCount = 0;
if (AudioTrack::getMinFrameCount(&frameCount, AudioSystem::DEFAULT,
sampleRateInHertz) != NO_ERROR) {
return -1;
}
return frameCount * nbChannels * (audioFormat == javaAudioTrackFields.PCM16 ? 2 : 1);
}
可見，最小buffer size是frameCoun乘以聲道個數，在根據音頻格式乘以1或2得到。
聲道個數和音頻格式都是傳入的，不再說。
frameCount是調用函數AudioTrack::getMinFrameCount取得的。從函數名可知，此處取得的應該是最小frame數。
傳入的三個參數：
&frameCount是用來保存frame計數的。
sampleRateInHertz是采樣率。
AudioSystem::DEFAULT是寫死的。其定義在類AudioSystem中，其他的定義如下：
enum stream_type {
DEFAULT =-1,
VOICE_CALL = 0,
SYSTEM = 1,
RING = 2,
MUSIC = 3,
ALARM = 4,
NOTIFICATION = 5,
BLUETOOTH_SCO = 6,
ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker
DTMF = 8,
TTS = 9,
NUM_STREAM_TYPES
};
原來是stream的類型。
為什麼不在調用getMinBufferSize的時候傳入stream類型，而在此處使用DEFAULT呢？？？
先放放，繼續看函數AudioTrack::getMinFrameCount。
函數AudioTrack::getMinFrameCount的實現：
status_t AudioTrack::getMinFrameCount(
int* frameCount,
int streamType,
uint32_t sampleRate)
{
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
if (minBufCount < 2) minBufCount = 2;
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
return NO_ERROR;
}
開始，調用瞭三個AudioSystem的函數，似曾謀面，不過當時被無視瞭，今天看看吧。
函數AudioSystem::getOutputSamplingRate的實現：
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();
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);
} 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;
}
判斷流的類型，如果是DEFAULT，將其設置為MUSIC！
納爐嚎啕！！！
DEFAULT的流類型原來是這麼用的。
接下來根據stream type獲取output。
然後獲取output的描述。
若獲取成功，則output描述中的采樣率就是要獲取的采樣率。
否則，嘗試從AudioFlinger中獲取采樣率。
函數AudioSystem::getOutputFrameCount，AudioSystem::getOutputLatency，與函數AudioSystem::getOutputSamplingRate的處理類似。
至此，采樣率，frameCount和延遲都取得瞭。
接下來計算minBufCount：
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
if (minBufCount < 2) minBufCount = 2;
從註釋可知，buff大小應至少能覆蓋audio 硬件的延遲。
公式不太明白。
先看看從鏈接：https://blog.csdn.net/innost/article/details/6125779
中摘過來的frame的說明：
一個frame就是1個采樣點的字節數*聲道。為啥搞個frame出來？因為對於多聲道的話，用1個采樣點的字節數表示不全，
因為播放的時候肯定是多個聲道的數據都要播出來才行。所以為瞭方便，就說1秒鐘有多少個frame，這樣就能拋開聲道數，把意思表示全瞭。
還不是很明白。先放放。
猜瞭半天也猜不出來。哪位大俠指點指點。
下面計算frameCount：
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
我們的sampleRate肯定不為0，所以最後的計算應該為：afFrameCount * minBufCount * sampleRate / afSampleRate
摘自：江風的專欄