Review of android source code reference about AudioRecord.

DragonEmperorG commented 5 years ago

git clone https://android.googlesource.com/platform/frameworks/base git clone https://android.googlesource.com/platform/system/core

DragonEmperorG commented 5 years ago

// ----------------------------------------------------------------------------
// returns the minimum required size for the successful creation of an AudioRecord instance.
// returns 0 if the parameter combination is not supported.
// return -1 if there was an error querying the buffer size.
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;
    audio_format_t format = audioFormatToNative(audioFormat);
    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);
}

DragonEmperorG commented 5 years ago

static inline audio_format_t audioFormatToNative(int audioFormat)
{
    switch (audioFormat) {
    case ENCODING_PCM_16BIT:
        return AUDIO_FORMAT_PCM_16_BIT;
    case ENCODING_PCM_8BIT:
        return AUDIO_FORMAT_PCM_8_BIT;
    case ENCODING_PCM_FLOAT:
        return AUDIO_FORMAT_PCM_FLOAT;
    case ENCODING_AC3:
        return AUDIO_FORMAT_AC3;
    case ENCODING_E_AC3:
        return AUDIO_FORMAT_E_AC3;
    case ENCODING_DTS:
        return AUDIO_FORMAT_DTS;
    case ENCODING_DTS_HD:
        return AUDIO_FORMAT_DTS_HD;
    case ENCODING_MP3:
        return AUDIO_FORMAT_MP3;
    case ENCODING_AAC_LC:
        return AUDIO_FORMAT_AAC_LC;
    case ENCODING_AAC_HE_V1:
        return AUDIO_FORMAT_AAC_HE_V1;
    case ENCODING_AAC_HE_V2:
        return AUDIO_FORMAT_AAC_HE_V2;
    case ENCODING_DOLBY_TRUEHD:
        return AUDIO_FORMAT_DOLBY_TRUEHD;
    case ENCODING_IEC61937:
        return AUDIO_FORMAT_IEC61937;
    case ENCODING_AAC_ELD:
        return AUDIO_FORMAT_AAC_ELD;
    case ENCODING_AAC_XHE:
        return AUDIO_FORMAT_AAC_XHE;
    case ENCODING_AC4:
        return AUDIO_FORMAT_AC4;
    case ENCODING_E_AC3_JOC:
        return AUDIO_FORMAT_E_AC3_JOC;
    case ENCODING_DEFAULT:
        return AUDIO_FORMAT_DEFAULT;
    default:
        return AUDIO_FORMAT_INVALID;
    }
}

/* Audio format consists of a main format field (upper 8 bits) and a sub format
 * field (lower 24 bits).
 *
 * The main format indicates the main codec type. The sub format field
 * indicates options and parameters for each format. The sub format is mainly
 * used for record to indicate for instance the requested bitrate or profile.
 * It can also be used for certain formats to give informations not present in
 * the encoded audio stream (e.g. octet alignement for AMR).
 */
typedef enum {
    AUDIO_FORMAT_INVALID             = 0xFFFFFFFFUL,
    AUDIO_FORMAT_DEFAULT             = 0,
    AUDIO_FORMAT_PCM                 = 0x00000000UL, /* DO NOT CHANGE */
    AUDIO_FORMAT_MP3                 = 0x01000000UL,
    AUDIO_FORMAT_AMR_NB              = 0x02000000UL,
    AUDIO_FORMAT_AMR_WB              = 0x03000000UL,
    AUDIO_FORMAT_AAC                 = 0x04000000UL,
    AUDIO_FORMAT_HE_AAC_V1           = 0x05000000UL, /* Deprecated, Use AUDIO_FORMAT_AAC_HE_V1*/
    AUDIO_FORMAT_HE_AAC_V2           = 0x06000000UL, /* Deprecated, Use AUDIO_FORMAT_AAC_HE_V2*/
    AUDIO_FORMAT_VORBIS              = 0x07000000UL,
    AUDIO_FORMAT_OPUS                = 0x08000000UL,
    AUDIO_FORMAT_AC3                 = 0x09000000UL,
    AUDIO_FORMAT_E_AC3               = 0x0A000000UL,
    AUDIO_FORMAT_MAIN_MASK           = 0xFF000000UL,
    AUDIO_FORMAT_SUB_MASK            = 0x00FFFFFFUL,
    /* Aliases */
    /* note != AudioFormat.ENCODING_PCM_16BIT */
    AUDIO_FORMAT_PCM_16_BIT          = (AUDIO_FORMAT_PCM |
                                        AUDIO_FORMAT_PCM_SUB_16_BIT),
    /* note != AudioFormat.ENCODING_PCM_8BIT */
    AUDIO_FORMAT_PCM_8_BIT           = (AUDIO_FORMAT_PCM |
                                        AUDIO_FORMAT_PCM_SUB_8_BIT),
    AUDIO_FORMAT_PCM_32_BIT          = (AUDIO_FORMAT_PCM |
                                        AUDIO_FORMAT_PCM_SUB_32_BIT),
    AUDIO_FORMAT_PCM_8_24_BIT        = (AUDIO_FORMAT_PCM |
                                        AUDIO_FORMAT_PCM_SUB_8_24_BIT),
    AUDIO_FORMAT_PCM_FLOAT           = (AUDIO_FORMAT_PCM |
                                        AUDIO_FORMAT_PCM_SUB_FLOAT),
    AUDIO_FORMAT_PCM_24_BIT_PACKED   = (AUDIO_FORMAT_PCM |
                                        AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED),
    AUDIO_FORMAT_AAC_MAIN            = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_MAIN),
    AUDIO_FORMAT_AAC_LC              = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_LC),
    AUDIO_FORMAT_AAC_SSR             = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_SSR),
    AUDIO_FORMAT_AAC_LTP             = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_LTP),
    AUDIO_FORMAT_AAC_HE_V1           = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_HE_V1),
    AUDIO_FORMAT_AAC_SCALABLE        = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_SCALABLE),
    AUDIO_FORMAT_AAC_ERLC            = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_ERLC),
    AUDIO_FORMAT_AAC_LD              = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_LD),
    AUDIO_FORMAT_AAC_HE_V2           = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_HE_V2),
    AUDIO_FORMAT_AAC_ELD             = (AUDIO_FORMAT_AAC |
                                        AUDIO_FORMAT_AAC_SUB_ELD),
} audio_format_t;

/* PCM sub formats */
typedef enum {
    /* All of these are in native byte order */
    AUDIO_FORMAT_PCM_SUB_16_BIT          = 0x1, /* DO NOT CHANGE - PCM signed 16 bits */
    AUDIO_FORMAT_PCM_SUB_8_BIT           = 0x2, /* DO NOT CHANGE - PCM unsigned 8 bits */
    AUDIO_FORMAT_PCM_SUB_32_BIT          = 0x3, /* PCM signed .31 fixed point */
    AUDIO_FORMAT_PCM_SUB_8_24_BIT        = 0x4, /* PCM signed 7.24 fixed point */
    AUDIO_FORMAT_PCM_SUB_FLOAT           = 0x5, /* PCM single-precision floating point */
    AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED   = 0x6, /* PCM signed .23 fixed point packed in 3 bytes */
} audio_format_pcm_sub_fmt_t;

DragonEmperorG commented 5 years ago

// static
status_t AudioRecord::getMinFrameCount(
        int* frameCount,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask)
{
    if (frameCount == NULL) return BAD_VALUE;
    // default to 0 in case of error
    *frameCount = 0;
    size_t size = 0;
    if (AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size)
            != NO_ERROR) {
        ALOGE("AudioSystem could not query the input buffer size.");
        return NO_INIT;
    }
    if (size == 0) {
        ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
            sampleRate, format, channelMask);
        return BAD_VALUE;
    }
    // We double the size of input buffer for ping pong use of record buffer.
    size <<= 1;
    if (audio_is_linear_pcm(format)) {
        int channelCount = popcount(channelMask);
        size /= channelCount * audio_bytes_per_sample(format);
    }
    *frameCount = size;
    return NO_ERROR;
}

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;
}

/* Derive an input channel mask for position assignment from a channel count.
 * Currently handles only mono and stereo.
 * Returns the matching channel mask,
 * or AUDIO_CHANNEL_NONE if the channel count is zero,
 * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the
 * configurations for which a default input channel mask is defined.
 */
static inline audio_channel_mask_t audio_channel_in_mask_from_count(uint32_t channel_count)
{
    uint32_t bits;
    switch (channel_count) {
    case 0:
        return AUDIO_CHANNEL_NONE;
    case 1:
        bits = AUDIO_CHANNEL_IN_MONO;
        break;
    case 2:
        bits = AUDIO_CHANNEL_IN_STEREO;
        break;
    default:
        return AUDIO_CHANNEL_INVALID;
    }
    return audio_channel_mask_from_representation_and_bits(
            AUDIO_CHANNEL_REPRESENTATION_POSITION, bits);
}

/* For the channel mask for position assignment representation */
enum {
/* These can be a complete audio_channel_mask_t. */
    AUDIO_CHANNEL_NONE                      = 0x0,
    AUDIO_CHANNEL_INVALID                   = 0xC0000000,
/* These can be the bits portion of an audio_channel_mask_t
 * with representation AUDIO_CHANNEL_REPRESENTATION_POSITION.
 * Using these bits as a complete audio_channel_mask_t is deprecated.
 */
    /* output channels */
    AUDIO_CHANNEL_OUT_FRONT_LEFT            = 0x1,
    AUDIO_CHANNEL_OUT_FRONT_RIGHT           = 0x2,
    AUDIO_CHANNEL_OUT_FRONT_CENTER          = 0x4,
    AUDIO_CHANNEL_OUT_LOW_FREQUENCY         = 0x8,
    AUDIO_CHANNEL_OUT_BACK_LEFT             = 0x10,
    AUDIO_CHANNEL_OUT_BACK_RIGHT            = 0x20,
    AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER  = 0x40,
    AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x80,
    AUDIO_CHANNEL_OUT_BACK_CENTER           = 0x100,
    AUDIO_CHANNEL_OUT_SIDE_LEFT             = 0x200,
    AUDIO_CHANNEL_OUT_SIDE_RIGHT            = 0x400,
    AUDIO_CHANNEL_OUT_TOP_CENTER            = 0x800,
    AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT        = 0x1000,
    AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER      = 0x2000,
    AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT       = 0x4000,
    AUDIO_CHANNEL_OUT_TOP_BACK_LEFT         = 0x8000,
    AUDIO_CHANNEL_OUT_TOP_BACK_CENTER       = 0x10000,
    AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT        = 0x20000,
/* TODO: should these be considered complete channel masks, or only bits? */
    AUDIO_CHANNEL_OUT_MONO     = AUDIO_CHANNEL_OUT_FRONT_LEFT,
    AUDIO_CHANNEL_OUT_STEREO   = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT),
    AUDIO_CHANNEL_OUT_QUAD     = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
                                  AUDIO_CHANNEL_OUT_BACK_RIGHT),
    AUDIO_CHANNEL_OUT_QUAD_BACK = AUDIO_CHANNEL_OUT_QUAD,
    /* like AUDIO_CHANNEL_OUT_QUAD_BACK with *_SIDE_* instead of *_BACK_* */
    AUDIO_CHANNEL_OUT_QUAD_SIDE = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
                                  AUDIO_CHANNEL_OUT_SIDE_LEFT |
                                  AUDIO_CHANNEL_OUT_SIDE_RIGHT),
    AUDIO_CHANNEL_OUT_5POINT1  = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
                                  AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
                                  AUDIO_CHANNEL_OUT_BACK_RIGHT),
    AUDIO_CHANNEL_OUT_5POINT1_BACK = AUDIO_CHANNEL_OUT_5POINT1,
    /* like AUDIO_CHANNEL_OUT_5POINT1_BACK with *_SIDE_* instead of *_BACK_* */
    AUDIO_CHANNEL_OUT_5POINT1_SIDE = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
                                  AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
                                  AUDIO_CHANNEL_OUT_SIDE_LEFT |
                                  AUDIO_CHANNEL_OUT_SIDE_RIGHT),
    // matches the correct AudioFormat.CHANNEL_OUT_7POINT1_SURROUND definition for 7.1
    AUDIO_CHANNEL_OUT_7POINT1  = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
                                  AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
                                  AUDIO_CHANNEL_OUT_BACK_RIGHT |
                                  AUDIO_CHANNEL_OUT_SIDE_LEFT |
                                  AUDIO_CHANNEL_OUT_SIDE_RIGHT),
    AUDIO_CHANNEL_OUT_ALL      = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
                                  AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
                                  AUDIO_CHANNEL_OUT_BACK_RIGHT |
                                  AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER |
                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER |
                                  AUDIO_CHANNEL_OUT_BACK_CENTER|
                                  AUDIO_CHANNEL_OUT_SIDE_LEFT|
                                  AUDIO_CHANNEL_OUT_SIDE_RIGHT|
                                  AUDIO_CHANNEL_OUT_TOP_CENTER|
                                  AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT|
                                  AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER|
                                  AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT|
                                  AUDIO_CHANNEL_OUT_TOP_BACK_LEFT|
                                  AUDIO_CHANNEL_OUT_TOP_BACK_CENTER|
                                  AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT),
/* These are bits only, not complete values */
    /* input channels */
    AUDIO_CHANNEL_IN_LEFT            = 0x4,
    AUDIO_CHANNEL_IN_RIGHT           = 0x8,
    AUDIO_CHANNEL_IN_FRONT           = 0x10,
    AUDIO_CHANNEL_IN_BACK            = 0x20,
    AUDIO_CHANNEL_IN_LEFT_PROCESSED  = 0x40,
    AUDIO_CHANNEL_IN_RIGHT_PROCESSED = 0x80,
    AUDIO_CHANNEL_IN_FRONT_PROCESSED = 0x100,
    AUDIO_CHANNEL_IN_BACK_PROCESSED  = 0x200,
    AUDIO_CHANNEL_IN_PRESSURE        = 0x400,
    AUDIO_CHANNEL_IN_X_AXIS          = 0x800,
    AUDIO_CHANNEL_IN_Y_AXIS          = 0x1000,
    AUDIO_CHANNEL_IN_Z_AXIS          = 0x2000,
    AUDIO_CHANNEL_IN_VOICE_UPLINK    = 0x4000,
    AUDIO_CHANNEL_IN_VOICE_DNLINK    = 0x8000,
/* TODO: should these be considered complete channel masks, or only bits, or deprecated? */
    AUDIO_CHANNEL_IN_MONO   = AUDIO_CHANNEL_IN_FRONT,
    AUDIO_CHANNEL_IN_STEREO = (AUDIO_CHANNEL_IN_LEFT | AUDIO_CHANNEL_IN_RIGHT),
    AUDIO_CHANNEL_IN_FRONT_BACK = (AUDIO_CHANNEL_IN_FRONT | AUDIO_CHANNEL_IN_BACK),
    AUDIO_CHANNEL_IN_ALL    = (AUDIO_CHANNEL_IN_LEFT |
                               AUDIO_CHANNEL_IN_RIGHT |
                               AUDIO_CHANNEL_IN_FRONT |
                               AUDIO_CHANNEL_IN_BACK|
                               AUDIO_CHANNEL_IN_LEFT_PROCESSED |
                               AUDIO_CHANNEL_IN_RIGHT_PROCESSED |
                               AUDIO_CHANNEL_IN_FRONT_PROCESSED |
                               AUDIO_CHANNEL_IN_BACK_PROCESSED|
                               AUDIO_CHANNEL_IN_PRESSURE |
                               AUDIO_CHANNEL_IN_X_AXIS |
                               AUDIO_CHANNEL_IN_Y_AXIS |
                               AUDIO_CHANNEL_IN_Z_AXIS |
                               AUDIO_CHANNEL_IN_VOICE_UPLINK |
                               AUDIO_CHANNEL_IN_VOICE_DNLINK),
};

DragonEmperorG commented 5 years ago

// establish binder interface to AudioFlinger service
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;
}

IAudioFlinger.h

class IAudioFlinger : public IInterface
{
public:
    // retrieve the audio recording buffer size
    virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const = 0;
};

IAudioFlinger.cpp

class BpAudioFlinger : public BpInterface<IAudioFlinger>
{
public:

    virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
            audio_channel_mask_t channelMask) const
    {
        Parcel data, reply;
        data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
        data.writeInt32(sampleRate);
        data.writeInt32(format);
        data.writeInt32(channelMask);
        remote()->transact(GET_INPUTBUFFERSIZE, data, &reply);
        return reply.readInt64();
    }
};

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;
    }
    if ((sampleRate == 0) ||
            !audio_is_valid_format(format) || !audio_has_proportional_frames(format) ||
            !audio_is_input_channel(channelMask)) {
        return 0;
    }

    AutoMutex lock(mHardwareLock);
    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
    audio_config_t config, proposed;
    memset(&proposed, 0, sizeof(proposed));
    proposed.sample_rate = sampleRate;
    proposed.channel_mask = channelMask;
    proposed.format = format;

    sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
    size_t frames;
    for (;;) {
        // Note: config is currently a const parameter for get_input_buffer_size()
        // but we use a copy from proposed in case config changes from the call.
        config = proposed;
        status_t result = dev->getInputBufferSize(&config, &frames);
        if (result == OK && frames != 0) {
            break; // hal success, config is the result
        }
        // change one parameter of the configuration each iteration to a more "common" value
        // to see if the device will support it.
        if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) {
            proposed.format = AUDIO_FORMAT_PCM_16_BIT;
        } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as
            proposed.sample_rate = 44100;           // legacy AudioRecord.java. TODO: Query hw?
        } else {
            ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, "
                    "format %#x, channelMask 0x%X",
                    sampleRate, format, channelMask);
            break; // retries failed, break out of loop with frames == 0.
        }
    }
    mHardwareStatus = AUDIO_HW_IDLE;
    if (frames > 0 && config.sample_rate != sampleRate) {
        frames = destinationFramesPossible(frames, sampleRate, config.sample_rate);
    }
    return frames; // may be converted to bytes at the Java level.
}

DragonEmperorG / AcousticNativeModule

Review of android source code reference about AudioRecord. #1