在文章《基于Allwinner的Audio子系统分析(Android-5.1)》中已经介绍了Audio的系统架构以及应用层调用的流程，接下来，继续分析AudioRecorder方法中的getMinBufferSize的实现
函数原型：
public static int getMinBufferSize (int sampleRateInHz, int channelConfig, int audioFormat)
作用：
返回成功创建AudioRecord对象所需要的最小缓冲区大小
参数：
sampleRateInHz：默认采样率，单位Hz，这里设置为44100，44100Hz是当前唯一能保证在所有设备上工作的采样率；
channelConfig： 描述音频声道设置，这里设置为AudioFormat.CHANNEL_CONFIGURATION_MONO，CHANNEL_CONFIGURATION_MONO保证能在所有设备上工作；
audioFormat：音频数据的采样精度，这里设置为AudioFormat.ENCODING_16BIT；
返回值：
返回成功创建AudioRecord对象所需要的最小缓冲区大小。 注意：这个大小并不保证在负荷下的流畅录制，应根据预期的频率来选择更高的值，AudioRecord实例在推送新数据时使用此值
如果硬件不支持录制参数，或输入了一个无效的参数，则返回ERROR_BAD_VALUE（-2），如果硬件查询到输出属性没有实现，或最小缓冲区用byte表示，则返回ERROR（-1）
接下来进入系统分析具体实现
frameworks/base/media/java/android/media/AudioRecord.java
static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
int channelCount = 0;
switch (channelConfig) {
case AudioFormat.CHANNEL_IN_DEFAULT: // AudioFormat.CHANNEL_CONFIGURATION_DEFAULT //1
case AudioFormat.CHANNEL_IN_MONO: //16
case AudioFormat.CHANNEL_CONFIGURATION_MONO://2
channelCount = 1;
break;
case AudioFormat.CHANNEL_IN_STEREO: //12
case AudioFormat.CHANNEL_CONFIGURATION_STEREO://3
case (AudioFormat.CHANNEL_IN_FRONT | AudioFormat.CHANNEL_IN_BACK): // 16||32
channelCount = 2;
break;
case AudioFormat.CHANNEL_INVALID://0
default:
loge("getMinBufferSize(): Invalid channel configuration.");
return ERROR_BAD_VALUE;
}
// PCM_8BIT is not supported at the moment
if (audioFormat != AudioFormat.ENCODING_PCM_16BIT) {
loge("getMinBufferSize(): Invalid audio format.");
return ERROR_BAD_VALUE;
}
int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
if (size == 0) {
return ERROR_BAD_VALUE;
}
else if (size == -1) {
return ERROR;
}
else {
return size;
}
}
对音频通道与音频采样精度进行判断，单声道（MONO）时channelCount为1，立体声（STEREO）时channelCount为2，且A64上仅支持PCM_16BIT采样，其值为2，然后继续调用native函数
frameworks/base/core/jni/android_media_AudioRecord.cpp
static jint android_media_AudioRecord_get_min_buff_size(JNIEnv *env, jobject thiz,
jint sampleRateInHertz, jint channelCount, jint audioFormat) {
ALOGV(">> android_media_AudioRecord_get_min_buff_size(%d, %d, %d)",
sampleRateInHertz, channelCount, audioFormat);
size_t frameCount = 0;
//从java转成jni的format类型
audio_format_t format = audioFormatToNative(audioFormat);//AUDIO_FORMAT_PCM_16_BIT=0x1
//获取frameCount，并判断硬件是否支持
status_t result = AudioRecord::getMinFrameCount(&frameCount,
sampleRateInHertz,
format,
audio_channel_in_mask_from_count(channelCount));
if (result == BAD_VALUE) {
return 0;
}
if (result != NO_ERROR) {
return -1;
}
return frameCount * channelCount * audio_bytes_per_sample(format);
}
调用服务端的函数，获取frameCount大小，最后返回了frameCount*声道数*采样精度，其中frameCount表示最小采样帧数，继续分析frameCount的计算方法
frameworks/av/media/libmedia/AudioRecord.cpp
status_t AudioRecord::getMinFrameCount(
size_t* frameCount,
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask)
{
if (frameCount == NULL) {
return BAD_VALUE;
}
size_t size;
status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
if (status != NO_ERROR) {
ALOGE("AudioSystem could not query the input buffer size for sampleRate %u, format %#x, "
"channelMask %#x; status %d", sampleRate, format, channelMask, status);
return status;
}
//计算出最小的frame
// We double the size of input buffer for ping pong use of record buffer.
// Assumes audio_is_linear_pcm(format)
if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) *
audio_bytes_per_sample(format))) == 0) {
ALOGE("Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
return NO_ERROR;
}
此时frameCount= size*2/(声道数*采样精度)，注意这里需要double一下，而size是由hal层得到的，AudioSystem::getInputBufferSize()函数最终会调用到HAL层
frameworks/av/media/libmedia/AudioSystem.cpp
status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask, size_t* buffSize)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) {
return PERMISSION_DENIED;
}
Mutex::Autolock _l(gLockCache);
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
size_t inBuffSize = gInBuffSize;
if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
|| (channelMask != gPrevInChannelMask)) {
gLockCache.unlock();
inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
gLockCache.lock();
if (inBuffSize == 0) {
ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %x",
sampleRate, format, channelMask);
return BAD_VALUE;
}
// A benign race is possible here: we could overwrite a fresher cache entry
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
gPrevInChannelMask = channelMask;
gInBuffSize = inBuffSize;
}
*buffSize = inBuffSize;
return NO_ERROR;
}
这里通过get_audio_flinger获取到了一个AudioFlinger对象
const sp<IAudioFlinger> AudioSystem::get_audio_flinger()
{
sp<IAudioFlinger> af;
sp<AudioFlingerClient> afc;
{
Mutex::Autolock _l(gLock);
if (gAudioFlinger == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
binder = sm->getService(String16("media.audio_flinger"));
if (binder != 0)
break;
ALOGW("AudioFlinger not published, waiting...");
usleep(500000); // 0.5 s
} while (true);
if (gAudioFlingerClient == NULL) {
gAudioFlingerClient = new AudioFlingerClient();
} else {
if (gAudioErrorCallback) {
gAudioErrorCallback(NO_ERROR);
}
}
binder->linkToDeath(gAudioFlingerClient);
gAudioFlinger = interface_cast<IAudioFlinger>(binder);
LOG_ALWAYS_FATAL_IF(gAudioFlinger == 0);
afc = gAudioFlingerClient;
}
af = gAudioFlinger;
}
if (afc != 0) {
af->registerClient(afc);
}
return af;
}
然后判断是否参数是之前配置过的参数，这样做是为了防止重复多次调用getMinBufferSize导致占用硬件资源，所以当第一次调用或更新参数调用后，则调用AF中的getInputBufferSize方法获取BuffSize，而af是IAudioFlinger类型的智能指针，所以实际上会通过binder到达AudioFlinger中
frameworks\av\services\audioflinger\AudioFlinger.cpp
size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
audio_channel_mask_t channelMask) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
audio_config_t config;
memset(&config, 0, sizeof(config));
config.sample_rate = sampleRate;
config.channel_mask = channelMask;
config.format = format;
audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice();
size_t size = dev->get_input_buffer_size(dev, &config);
mHardwareStatus = AUDIO_HW_IDLE;
return size;
}
把参数传递给hal层，获取buffer大小
hardware\aw\audio\tulip\audio_hw.c
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
size_t size;
int channel_count = popcount(config->channel_mask);
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
return 0;
return get_input_buffer_size(config->sample_rate, config->format, channel_count);
}
再次检查一次参数是否正确，为什么在很多函数里面都做一次检查参数呢？可能在其他的地方也调用到了这个函数，所以最好做一次检查，确保万无一失
static size_t get_input_buffer_size(uint32_t sample_rate, int format, int channel_count)
{
size_t size;
size_t device_rate;
if (check_input_parameters(sample_rate, format, channel_count) != 0)
return 0;
/* take resampling into account and return the closest majoring
multiple of 16 frames, as audioflinger expects audio buffers to
be a multiple of 16 frames */
size = (pcm_config_mm_in.period_size * sample_rate) / pcm_config_mm_in.rate;
size = ((size + 15) / 16) * 16;
return size * channel_count * sizeof(short);
}
这里包含一个结构体struct pcm_config，定义了一个周期包含了多少采样帧，并根据结构体的rate数据进行重采样计算，这里的rate是以MM_SAMPLING_RATE为标准，即44100，一个采样周期有1024个采样帧，然后计算出重采样之后的size
同时audioflinger的音频buffer是16的整数倍，所以再次计算得出一个最接近16倍的整数，最后返回size*声道数*1帧数据所占字节数
struct pcm_config pcm_config_mm_in = {
.channels = 2,
.rate = MM_SAMPLING_RATE,
.period_size = 1024,
.period_count = CAPTURE_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
};
总结：
minBuffSize = ((((((((pcm_config_mm_in.period_size * sample_rate) / pcm_config_mm_in.rate) + 15) / 16) * 16) * channel_count * sizeof(short)) * 2) / (audio_channel_count_from_in_mask(channelMask) * audio_bytes_per_sample(format))) * channelCount * audio_bytes_per_sample(format);
=(((((((pcm_config_mm_in.period_size * sample_rate) / pcm_config_mm_in.rate) + 15) / 16) * 16) * channel_count * sizeof(short)) * 2)
其中：pcm_config_mm_in.period_size=1024；pcm_config_mm_in.rate=44100；这里我们可以看到他除掉(channelCount*format)，后面又乘回来了，这个是因为在AudioRecord.cpp对frameCount进行了一次校验，判断是否支持该参数的设置。
以getMinBufferSize(44100, MONO, 16BIT);为例，即sample_rate=44100，channel_count=1， format=2，那么
BufferSize = (((1024*sample_rate/44100)+15)/16)*16*channel_count*sizeof(short)*2 = 4096
即最小缓冲区大小为：周期大小 * 重采样 * 采样声道数 * 2 * 采样精度所占字节数；这里的2的解释为We double the size of input buffer for ping pong use of record buffer，采样精度：PCM_8_BIT为unsigned char，PCM_16_BIT为short，PCM_32_BIT为int。
由于作者内功有限，若文章中存在错误或不足的地方，还请给位大佬指出，不胜感激！