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Commit 9098999b authored by Ravi Kumar Alamanda's avatar Ravi Kumar Alamanda Committed by Steve Kondik
Browse files

hal: add support for low-latency capture 

- add support for low-latency capture

Change-Id: Ic8a82854799adfa4eb1fcd323e0177eeadc7e319
parent f37cbc13

hal/Android.mk

+4 −0
Original line number Diff line number Diff line
@@ -161,6 +161,10 @@ ifeq ($(strip $(AUDIO_FEATURE_ENABLED_FLAC_OFFLOAD)),true) 
    LOCAL_CFLAGS += -DFLAC_OFFLOAD_ENABLED

endif



ifeq ($(string $(AUDIO_FEATURE_ENABLED_LOW_LATENCY_CAPTURE)),true)

    LOCAL_CFLAGS += -DLOW_LATENCY_CAPTURE_USE_CASE=1

endif



LOCAL_SHARED_LIBRARIES := \

	liblog \

	libcutils \

hal/audio_hw.c

+76 −21
Original line number Diff line number Diff line
@@ -18,7 +18,7 @@ 
 */



#define LOG_TAG "audio_hw_primary"

/*#define LOG_NDEBUG 0*/

#define LOG_NDEBUG 0

/*#define VERY_VERY_VERBOSE_LOGGING*/

#ifdef VERY_VERY_VERBOSE_LOGGING

#define ALOGVV ALOGV
@@ -69,6 +69,9 @@ 
#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER

#endif



static unsigned int configured_low_latency_capture_period_size =

        LOW_LATENCY_CAPTURE_PERIOD_SIZE;



struct pcm_config pcm_config_deep_buffer = {

    .channels = 2,

    .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
@@ -357,6 +360,19 @@ static int set_snd_card_state(struct audio_device *adev, int snd_scard_state) 
    return 0;

}



int pcm_ioctl(struct pcm *pcm, int request, ...)

{

    va_list ap;

    void * arg;

    int pcm_fd = *(int*)pcm;



    va_start(ap, request);

    arg = va_arg(ap, void *);

    va_end(ap);



    return ioctl(pcm_fd, request, arg);

}



int enable_audio_route(struct audio_device *adev,

                       struct audio_usecase *usecase)

{
@@ -1474,7 +1490,8 @@ static int check_input_parameters(uint32_t sample_rate, 


static size_t get_input_buffer_size(uint32_t sample_rate,

                                    audio_format_t format,

                                    int channel_count)

                                    int channel_count,

                                    bool is_low_latency)

{

    size_t size = 0;
@@ -1482,13 +1499,19 @@ static size_t get_input_buffer_size(uint32_t sample_rate, 
        return 0;



    size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;

    if (is_low_latency)

        size = configured_low_latency_capture_period_size;

    /* ToDo: should use frame_size computed based on the format and

       channel_count here. */

    size *= sizeof(short) * channel_count;



    /* make sure the size is multiple of 64 */

    size += 0x3f;

    size &= ~0x3f;

    /* make sure the size is multiple of 32 bytes

     * At 48 kHz mono 16-bit PCM:

     *  5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)

     *  3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)

     */

    size += 0x1f;

    size &= ~0x1f;



    return size;

}
@@ -3065,7 +3088,8 @@ static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unu 
{

    int channel_count = audio_channel_count_from_in_mask(config->channel_mask);



    return get_input_buffer_size(config->sample_rate, config->format, channel_count);

    return get_input_buffer_size(config->sample_rate, config->format, channel_count,

            false /* is_low_latency: since we don't know, be conservative */);

}



static int adev_open_input_stream(struct audio_hw_device *dev,
@@ -3081,7 +3105,7 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
    struct stream_in *in;

    int ret = 0, buffer_size, frame_size;

    int channel_count = audio_channel_count_from_in_mask(config->channel_mask);



    bool is_low_latency = false;



    *stream_in = NULL;

    if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
@@ -3124,6 +3148,13 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 


    /* Update config params with the requested sample rate and channels */

    in->usecase = USECASE_AUDIO_RECORD;

    if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&

            (flags & AUDIO_INPUT_FLAG_FAST) != 0) {

        is_low_latency = true;

#if LOW_LATENCY_CAPTURE_USE_CASE

        in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;

#endif

    }

    in->config = pcm_config_audio_capture;

    in->config.rate = config->sample_rate;

    in->format = config->format;
@@ -3167,7 +3198,8 @@ static int adev_open_input_stream(struct audio_hw_device *dev, 
        frame_size = audio_stream_in_frame_size(&in->stream);

        buffer_size = get_input_buffer_size(config->sample_rate,

                                            config->format,

                                            channel_count);

                                            channel_count,

                                            is_low_latency);

        in->config.period_size = buffer_size / frame_size;

    }
@@ -3239,6 +3271,23 @@ static int adev_close(hw_device_t *device) 
    return 0;

}



/* This returns 1 if the input parameter looks at all plausible as a low latency period size,

 * or 0 otherwise.  A return value of 1 doesn't mean the value is guaranteed to work,

 * just that it _might_ work.

 */

static int period_size_is_plausible_for_low_latency(int period_size)

{

    switch (period_size) {

    case 160:

    case 240:

    case 320:

    case 480:

        return 1;

    default:

        return 0;

    }

}



static int adev_open(const hw_module_t *module, const char *name,

                     hw_device_t **device)

{
@@ -3357,23 +3406,29 @@ static int adev_open(const hw_module_t *module, const char *name, 
    *device = &adev->device.common;



    audio_device_ref_count++;

    pthread_mutex_unlock(&adev_init_lock);



    ALOGV("%s: exit", __func__);

    return 0;

    char value[PROPERTY_VALUE_MAX];

    int trial;

    if (property_get("audio_hal.period_size", value, NULL) > 0) {

        trial = atoi(value);

        if (period_size_is_plausible_for_low_latency(trial)) {

            pcm_config_low_latency.period_size = trial;

            pcm_config_low_latency.start_threshold = trial / 4;

            pcm_config_low_latency.avail_min = trial / 4;

            configured_low_latency_capture_period_size = trial;

        }

    }

    if (property_get("audio_hal.in_period_size", value, NULL) > 0) {

        trial = atoi(value);

        if (period_size_is_plausible_for_low_latency(trial)) {

            configured_low_latency_capture_period_size = trial;

        }

    }



int pcm_ioctl(struct pcm *pcm, int request, ...)

{

    va_list ap;

    void * arg;

    int pcm_fd = *(int*)pcm;



    va_start(ap, request);

    arg = va_arg(ap, void *);

    va_end(ap);

    pthread_mutex_unlock(&adev_init_lock);



    return ioctl(pcm_fd, request, arg);

    ALOGV("%s: exit", __func__);

    return 0;

}



static struct hw_module_methods_t hal_module_methods = {

hal/msm8960/platform.h

+3 −0
Original line number Diff line number Diff line
@@ -121,4 +121,7 @@ enum { 
#define AFE_PROXY_PLAYBACK_PCM_DEVICE 7

#define AFE_PROXY_RECORD_PCM_DEVICE 8



#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000

#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240



#endif // QCOM_AUDIO_PLATFORM_H

hal/msm8974/platform.h

+3 −0
Original line number Diff line number Diff line
@@ -167,6 +167,9 @@ enum { 
#define AUDIO_CAPTURE_PERIOD_DURATION_MSEC 20

#define AUDIO_CAPTURE_PERIOD_COUNT 2



#define LOW_LATENCY_CAPTURE_SAMPLE_RATE 48000

#define LOW_LATENCY_CAPTURE_PERIOD_SIZE 240



#define DEVICE_NAME_MAX_SIZE 128

#define HW_INFO_ARRAY_MAX_SIZE 32