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Commit bd08d63c authored by Ralf Herz's avatar Ralf Herz
Browse files

hal: Fix input buffer size for fractional rate 24bit capture 

The input buffer size from ADSP must be a multiple of 32 bytes and a
multiple of the frame_size. For fractional sample rates (eg. 44.1kHz)
with 24bit packed samples, 32 is not the smallest common multiple.

Change-Id: Ifcef1daa4f43e90fdb30873192c0047677ae3c00
parent 6225a270

hal/audio_hw.c

+14 −4
Original line number Diff line number Diff line
@@ -3289,6 +3289,8 @@ static size_t get_input_buffer_size(uint32_t sample_rate, 
                                    int channel_count,

                                    bool is_low_latency)

{

    int i = 0;

    size_t frame_size = 0;

    size_t size = 0;



    if (check_input_parameters(sample_rate, format, channel_count) != 0)
@@ -3298,15 +3300,23 @@ static size_t get_input_buffer_size(uint32_t sample_rate, 
    if (is_low_latency)

        size = configured_low_latency_capture_period_size;



    size *= audio_bytes_per_sample(format) * channel_count;

    frame_size = audio_bytes_per_sample(format) * channel_count;

    size *= frame_size;



    /* make sure the size is multiple of 32 bytes

    /* make sure the size is multiple of 32 bytes and additionally multiple of

     * the frame_size (required for 24bit samples and non-power-of-2 channel counts)

     * 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)

     *

     *  The loop reaches result within 32 iterations, as initial size is

     *  already a multiple of frame_size

     */

    size += 0x1f;

    size &= ~0x1f;

    for (i=0; i<32; i++) {

        if ((size & 0x1f) == 0)

            break;

        size += frame_size;

    }



    return size;

}