（一）Audio子系统之AudioRecord.getMinBufferSize

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！