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Verified Commit 36f96cd9 authored by stenkinevgeniy's avatar stenkinevgeniy Committed by Jan Altensen
Browse files

Audio: remove out stream resampler. 

This never use - because -
out->sample_rate and out->config configured in adev_open_output_stream
out->config = can have deep or no pcm config with only 48000 sample rate.

Change-Id: Ie1c4025585ab6e1b7a900486181c1bceeff88139
parent b35f6082

audio/audio_hw.c

+2 −52
Original line number Diff line number Diff line
@@ -2247,14 +2247,6 @@ static int out_close_pcm_devices(struct stream_out *out) 
            pcm_close(pcm_device->pcm);

            pcm_device->pcm = NULL;

        }

        if (pcm_device->resampler) {

            release_resampler(pcm_device->resampler);

            pcm_device->resampler = NULL;

        }

        if (pcm_device->res_buffer) {

            free(pcm_device->res_buffer);

            pcm_device->res_buffer = NULL;

        }

    }



    return 0;
@@ -2293,24 +2285,6 @@ static int out_open_pcm_devices(struct stream_out *out) 
            ret = -EIO;

            goto error_open;

        }

        /*

        * If the stream rate differs from the PCM rate, we need to

        * create a resampler.

        */

        if (out->sample_rate != pcm_device->pcm_profile->config.rate) {

            ALOGV("%s: create_resampler(), pcm_device_card(%d), pcm_device_id(%d), \

                    out_rate(%d), device_rate(%d)",__func__,

                    pcm_device_card, pcm_device_id,

                    out->sample_rate, pcm_device->pcm_profile->config.rate);

            ret = create_resampler(out->sample_rate,

                    pcm_device->pcm_profile->config.rate,

                    audio_channel_count_from_out_mask(out->channel_mask),

                    RESAMPLER_QUALITY_DEFAULT,

                    NULL,

                    &pcm_device->resampler);

            pcm_device->res_byte_count = 0;

            pcm_device->res_buffer = NULL;

        }

    }

    return ret;
@@ -2889,9 +2863,8 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer, 
    ssize_t ret = 0;

    struct pcm_device *pcm_device;

    struct listnode *node;

    size_t frame_size = audio_stream_out_frame_size(stream);

    size_t frames_wr = 0, frames_rq = 0;

#ifdef PREPROCESSING_ENABLED

    size_t frame_size = audio_stream_out_frame_size(stream);

    size_t in_frames = bytes / frame_size;

    size_t out_frames = in_frames;

    struct stream_in *in = NULL;
@@ -2987,24 +2960,6 @@ false_alarm: 
            memset((void *)buffer, 0, bytes);

        list_for_each(node, &out->pcm_dev_list) {

            pcm_device = node_to_item(node, struct pcm_device, stream_list_node);

            if (pcm_device->resampler) {

                if (bytes * pcm_device->pcm_profile->config.rate / out->sample_rate + frame_size

                        > pcm_device->res_byte_count) {

                    pcm_device->res_byte_count =

                        bytes * pcm_device->pcm_profile->config.rate / out->sample_rate + frame_size;

                    pcm_device->res_buffer =

                        realloc(pcm_device->res_buffer, pcm_device->res_byte_count);

                    ALOGV("%s: resampler res_byte_count = %zu", __func__,

                        pcm_device->res_byte_count);

                }

                frames_rq = bytes / frame_size;

                frames_wr = pcm_device->res_byte_count / frame_size;

                ALOGVV("%s: resampler request frames = %d frame_size = %d",

                    __func__, frames_rq, frame_size);

                pcm_device->resampler->resample_from_input(pcm_device->resampler,

                    (int16_t *)buffer, &frames_rq, (int16_t *)pcm_device->res_buffer, &frames_wr);

                ALOGVV("%s: resampler output frames_= %d", __func__, frames_wr);

            }

            if (pcm_device->pcm) {

#ifdef PREPROCESSING_ENABLED

                if (out->echo_reference != NULL && pcm_device->pcm_profile->devices != SND_DEVICE_OUT_SPEAKER) {
@@ -3017,11 +2972,6 @@ false_alarm: 
                 }

#endif

                ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);

                if (pcm_device->resampler && pcm_device->res_buffer)

                    pcm_device->status =

                        pcm_write(pcm_device->pcm, (void *)pcm_device->res_buffer,

                            frames_wr * frame_size);

                else

                pcm_device->status = pcm_write(pcm_device->pcm, (void *)buffer, bytes);

                if (pcm_device->status != 0)

                    ret = pcm_device->status;

audio/audio_hw.h

+0 −4
Original line number Diff line number Diff line
@@ -240,10 +240,6 @@ struct pcm_device { 
    struct pcm_device_profile* pcm_profile;

    struct pcm*                pcm;

    int                        status;

    /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */

    struct resampler_itfe*     resampler;

    int16_t*                   res_buffer;

    size_t                     res_byte_count;

};



struct stream_out {