Commit ed2a4a79 authored by Kővágó, Zoltán's avatar Kővágó, Zoltán Committed by Gerd Hoffmann
Browse files

audio: proper support for float samples in mixeng 



This adds proper support for float samples in mixeng by adding a new
audio format for it.

Limitations: only native endianness is supported.  None of the virtual
sound cards support float samples (it looks like most of them only
support 8 and 16 bit, only hda supports 32 bit), it is only used for the
audio backends (i.e. host side).

Signed-off-by: default avatarKővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Acked-by: default avatarMarkus Armbruster <armbru@redhat.com>
Message-id: 8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com
Signed-off-by: default avatarGerd Hoffmann <kraxel@redhat.com>
parent 180b044f

audio/alsaaudio.c

+17 −0
Original line number Diff line number Diff line
@@ -307,6 +307,13 @@ static snd_pcm_format_t aud_to_alsafmt (AudioFormat fmt, int endianness) 
            return SND_PCM_FORMAT_U32_LE;

        }



    case AUDIO_FORMAT_F32:

        if (endianness) {

            return SND_PCM_FORMAT_FLOAT_BE;

        } else {

            return SND_PCM_FORMAT_FLOAT_LE;

        }



    default:

        dolog ("Internal logic error: Bad audio format %d\n", fmt);

#ifdef DEBUG_AUDIO
@@ -370,6 +377,16 @@ static int alsa_to_audfmt (snd_pcm_format_t alsafmt, AudioFormat *fmt, 
        *fmt = AUDIO_FORMAT_U32;

        break;



    case SND_PCM_FORMAT_FLOAT_LE:

        *endianness = 0;

        *fmt = AUDIO_FORMAT_F32;

        break;



    case SND_PCM_FORMAT_FLOAT_BE:

        *endianness = 1;

        *fmt = AUDIO_FORMAT_F32;

        break;



    default:

        dolog ("Unrecognized audio format %d\n", alsafmt);

        return -1;

audio/audio.c

+38 −18
Original line number Diff line number Diff line
@@ -218,6 +218,9 @@ static void audio_print_settings (struct audsettings *as) 
    case AUDIO_FORMAT_U32:

        AUD_log (NULL, "U32");

        break;

    case AUDIO_FORMAT_F32:

        AUD_log (NULL, "F32");

        break;

    default:

        AUD_log (NULL, "invalid(%d)", as->fmt);

        break;
@@ -252,6 +255,7 @@ static int audio_validate_settings (struct audsettings *as) 
    case AUDIO_FORMAT_U16:

    case AUDIO_FORMAT_S32:

    case AUDIO_FORMAT_U32:

    case AUDIO_FORMAT_F32:

        break;

    default:

        invalid = 1;
@@ -264,24 +268,28 @@ static int audio_validate_settings (struct audsettings *as) 


static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *as)

{

    int bits = 8, sign = 0;

    int bits = 8;

    bool is_signed = false, is_float = false;



    switch (as->fmt) {

    case AUDIO_FORMAT_S8:

        sign = 1;

        is_signed = true;

        /* fall through */

    case AUDIO_FORMAT_U8:

        break;



    case AUDIO_FORMAT_S16:

        sign = 1;

        is_signed = true;

        /* fall through */

    case AUDIO_FORMAT_U16:

        bits = 16;

        break;



    case AUDIO_FORMAT_F32:

        is_float = true;

        /* fall through */

    case AUDIO_FORMAT_S32:

        sign = 1;

        is_signed = true;

        /* fall through */

    case AUDIO_FORMAT_U32:

        bits = 32;
@@ -292,33 +300,38 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a 
    }

    return info->freq == as->freq

        && info->nchannels == as->nchannels

        && info->sign == sign

        && info->is_signed == is_signed

        && info->is_float == is_float

        && info->bits == bits

        && info->swap_endianness == (as->endianness != AUDIO_HOST_ENDIANNESS);

}



void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as)

{

    int bits = 8, sign = 0, mul;

    int bits = 8, mul;

    bool is_signed = false, is_float = false;



    switch (as->fmt) {

    case AUDIO_FORMAT_S8:

        sign = 1;

        is_signed = true;

        /* fall through */

    case AUDIO_FORMAT_U8:

        mul = 1;

        break;



    case AUDIO_FORMAT_S16:

        sign = 1;

        is_signed = true;

        /* fall through */

    case AUDIO_FORMAT_U16:

        bits = 16;

        mul = 2;

        break;



    case AUDIO_FORMAT_F32:

        is_float = true;

        /* fall through */

    case AUDIO_FORMAT_S32:

        sign = 1;

        is_signed = true;

        /* fall through */

    case AUDIO_FORMAT_U32:

        bits = 32;
@@ -331,7 +344,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) 


    info->freq = as->freq;

    info->bits = bits;

    info->sign = sign;

    info->is_signed = is_signed;

    info->is_float = is_float;

    info->nchannels = as->nchannels;

    info->bytes_per_frame = as->nchannels * mul;

    info->bytes_per_second = info->freq * info->bytes_per_frame;
@@ -344,7 +358,7 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len) 
        return;

    }



    if (info->sign) {

    if (info->is_signed || info->is_float) {

        memset(buf, 0x00, len * info->bytes_per_frame);

    }

    else {
@@ -770,8 +784,9 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) 
#ifdef DEBUG_AUDIO

static void audio_pcm_print_info (const char *cap, struct audio_pcm_info *info)

{

    dolog ("%s: bits %d, sign %d, freq %d, nchan %d\n",

           cap, info->bits, info->sign, info->freq, info->nchannels);

    dolog("%s: bits %d, sign %d, float %d, freq %d, nchan %d\n",

          cap, info->bits, info->is_signed, info->is_float, info->freq,

          info->nchannels);

}

#endif
@@ -1832,11 +1847,15 @@ CaptureVoiceOut *AUD_add_capture( 


        cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame);



        if (hw->info.is_float) {

            hw->clip = mixeng_clip_float[hw->info.nchannels == 2];

        } else {

            hw->clip = mixeng_clip

                [hw->info.nchannels == 2]

            [hw->info.sign]

                [hw->info.is_signed]

                [hw->info.swap_endianness]

                [audio_bits_to_index(hw->info.bits)];

        }



        QLIST_INSERT_HEAD (&s->cap_head, cap, entries);

        QLIST_INSERT_HEAD (&cap->cb_head, cb, entries);
@@ -2075,6 +2094,7 @@ int audioformat_bytes_per_sample(AudioFormat fmt) 


    case AUDIO_FORMAT_U32:

    case AUDIO_FORMAT_S32:

    case AUDIO_FORMAT_F32:

        return 4;



    case AUDIO_FORMAT__MAX:

audio/audio_int.h

+2 −1
Original line number Diff line number Diff line
@@ -40,7 +40,8 @@ struct audio_callback { 


struct audio_pcm_info {

    int bits;

    int sign;

    bool is_signed;

    bool is_float;

    int freq;

    int nchannels;

    int bytes_per_frame;

audio/audio_template.h

+25 −16
Original line number Diff line number Diff line
@@ -153,15 +153,23 @@ static int glue (audio_pcm_sw_init_, TYPE) ( 
    sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq;

#endif



    if (sw->info.is_float) {

#ifdef DAC

        sw->conv = mixeng_conv_float[sw->info.nchannels == 2];

#else

        sw->clip = mixeng_clip_float[sw->info.nchannels == 2];

#endif

    } else {

#ifdef DAC

        sw->conv = mixeng_conv

#else

        sw->clip = mixeng_clip

#endif

            [sw->info.nchannels == 2]

        [sw->info.sign]

            [sw->info.is_signed]

            [sw->info.swap_endianness]

            [audio_bits_to_index(sw->info.bits)];

    }



    sw->name = g_strdup (name);

    err = glue (audio_pcm_sw_alloc_resources_, TYPE) (sw);
@@ -276,22 +284,23 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s, 
        goto err1;

    }



    if (s->dev->driver == AUDIODEV_DRIVER_COREAUDIO) {

    if (hw->info.is_float) {

#ifdef DAC

        hw->clip = clip_natural_float_from_stereo;

        hw->clip = mixeng_clip_float[hw->info.nchannels == 2];

#else

        hw->conv = conv_natural_float_to_stereo;

        hw->conv = mixeng_conv_float[hw->info.nchannels == 2];

#endif

    } else

    } else {

#ifdef DAC

        hw->clip = mixeng_clip

#else

        hw->conv = mixeng_conv

#endif

            [hw->info.nchannels == 2]

        [hw->info.sign]

            [hw->info.is_signed]

            [hw->info.swap_endianness]

            [audio_bits_to_index(hw->info.bits)];

    }



    glue(audio_pcm_hw_alloc_resources_, TYPE)(hw);

audio/coreaudio.c

+1 −6
Original line number Diff line number Diff line
@@ -491,14 +491,9 @@ static int coreaudio_init_out(HWVoiceOut *hw, struct audsettings *as, 
        return -1;

    }



    /*

     * The canonical audio format for CoreAudio on macOS is float. Currently

     * there is no generic code for AUDIO_FORMAT_F32 in qemu. Here we select

     * AUDIO_FORMAT_S32 instead because only the sample size has to match.

     */

    fake_as = *as;

    as = &fake_as;

    as->fmt = AUDIO_FORMAT_S32;

    as->fmt = AUDIO_FORMAT_F32;

    audio_pcm_init_info (&hw->info, as);



    status = coreaudio_get_voice(&core->outputDeviceID);