Commit 2b9cce8c authored by Kővágó, Zoltán's avatar Kővágó, Zoltán Committed by Gerd Hoffmann
Browse files

audio: replace shift in audio_pcm_info with bytes_per_frame 



The bit shifting trick worked because the number of bytes per frame was
always a power-of-two (since QEMU only supports mono, stereo and 8, 16
and 32 bit samples).  But if we want to add support for surround sound,
this no longer holds true.

Signed-off-by: default avatarKővágó, Zoltán <DirtY.iCE.hu@gmail.com>
Message-id: 1351fd9bcce0ff20d81850c5292722194329de02.1570996490.git.DirtY.iCE.hu@gmail.com
Signed-off-by: default avatarGerd Hoffmann <kraxel@redhat.com>
parent cecc1e79

audio/alsaaudio.c

+6 −5
Original line number Diff line number Diff line
@@ -602,7 +602,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) 
{

    ALSAVoiceOut *alsa = (ALSAVoiceOut *) hw;

    size_t pos = 0;

    size_t len_frames = len >> hw->info.shift;

    size_t len_frames = len / hw->info.bytes_per_frame;



    while (len_frames) {

        char *src = advance(buf, pos);
@@ -648,7 +648,7 @@ static size_t alsa_write(HWVoiceOut *hw, void *buf, size_t len) 
            }

        }



        pos += written << hw->info.shift;

        pos += written * hw->info.bytes_per_frame;

        if (written < len_frames) {

            break;

        }
@@ -802,7 +802,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) 
        void *dst = advance(buf, pos);

        snd_pcm_sframes_t nread;



        nread = snd_pcm_readi(alsa->handle, dst, len >> hw->info.shift);

        nread = snd_pcm_readi(

            alsa->handle, dst, len / hw->info.bytes_per_frame);



        if (nread <= 0) {

            switch (nread) {
@@ -828,8 +829,8 @@ static size_t alsa_read(HWVoiceIn *hw, void *buf, size_t len) 
            }

        }



        pos += nread << hw->info.shift;

        len -= nread << hw->info.shift;

        pos += nread * hw->info.bytes_per_frame;

        len -= nread * hw->info.bytes_per_frame;

    }



    return pos;

audio/audio.c

+37 −37
Original line number Diff line number Diff line
@@ -299,12 +299,13 @@ static int audio_pcm_info_eq (struct audio_pcm_info *info, struct audsettings *a 


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

{

    int bits = 8, sign = 0, shift = 0;

    int bits = 8, sign = 0, mul;



    switch (as->fmt) {

    case AUDIO_FORMAT_S8:

        sign = 1;

    case AUDIO_FORMAT_U8:

        mul = 1;

        break;



    case AUDIO_FORMAT_S16:
@@ -312,7 +313,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) 
        /* fall through */

    case AUDIO_FORMAT_U16:

        bits = 16;

        shift = 1;

        mul = 2;

        break;



    case AUDIO_FORMAT_S32:
@@ -320,7 +321,7 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) 
        /* fall through */

    case AUDIO_FORMAT_U32:

        bits = 32;

        shift = 2;

        mul = 4;

        break;



    default:
@@ -331,9 +332,8 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as) 
    info->bits = bits;

    info->sign = sign;

    info->nchannels = as->nchannels;

    info->shift = (as->nchannels == 2) + shift;

    info->align = (1 << info->shift) - 1;

    info->bytes_per_second = info->freq << info->shift;

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

    info->bytes_per_second = info->freq * info->bytes_per_frame;

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

}
@@ -344,26 +344,25 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len) 
    }



    if (info->sign) {

        memset (buf, 0x00, len << info->shift);

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

    }

    else {

        switch (info->bits) {

        case 8:

            memset (buf, 0x80, len << info->shift);

            memset(buf, 0x80, len * info->bytes_per_frame);

            break;



        case 16:

            {

                int i;

                uint16_t *p = buf;

                int shift = info->nchannels - 1;

                short s = INT16_MAX;



                if (info->swap_endianness) {

                    s = bswap16 (s);

                }



                for (i = 0; i < len << shift; i++) {

                for (i = 0; i < len * info->nchannels; i++) {

                    p[i] = s;

                }

            }
@@ -373,14 +372,13 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len) 
            {

                int i;

                uint32_t *p = buf;

                int shift = info->nchannels - 1;

                int32_t s = INT32_MAX;



                if (info->swap_endianness) {

                    s = bswap32 (s);

                }



                for (i = 0; i < len << shift; i++) {

                for (i = 0; i < len * info->nchannels; i++) {

                    p[i] = s;

                }

            }
@@ -558,7 +556,7 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) 


    while (len) {

        st_sample *src = hw->mix_buf->samples + pos;

        uint8_t *dst = advance(pcm_buf, clipped << hw->info.shift);

        uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame);

        size_t samples_till_end_of_buf = hw->mix_buf->size - pos;

        size_t samples_to_clip = MIN(len, samples_till_end_of_buf);
@@ -607,7 +605,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) 
        return 0;

    }



    samples = size >> sw->info.shift;

    samples = size / sw->info.bytes_per_frame;

    if (!live) {

        return 0;

    }
@@ -642,7 +640,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) 


    sw->clip (buf, sw->buf, ret);

    sw->total_hw_samples_acquired += total;

    return ret << sw->info.shift;

    return ret * sw->info.bytes_per_frame;

}



/*
@@ -715,7 +713,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) 
    }



    wpos = (sw->hw->mix_buf->pos + live) % hwsamples;

    samples = size >> sw->info.shift;

    samples = size / sw->info.bytes_per_frame;



    dead = hwsamples - live;

    swlim = ((int64_t) dead << 32) / sw->ratio;
@@ -759,13 +757,13 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) 
    dolog (

        "%s: write size %zu ret %zu total sw %zu\n",

        SW_NAME (sw),

        size >> sw->info.shift,

        size / sw->info.bytes_per_frame,

        ret,

        sw->total_hw_samples_mixed

        );

#endif



    return ret << sw->info.shift;

    return ret * sw->info.bytes_per_frame;

}



#ifdef DEBUG_AUDIO
@@ -882,7 +880,7 @@ size_t AUD_read(SWVoiceIn *sw, void *buf, size_t size) 


int AUD_get_buffer_size_out (SWVoiceOut *sw)

{

    return sw->hw->mix_buf->size << sw->hw->info.shift;

    return sw->hw->mix_buf->size * sw->hw->info.bytes_per_frame;

}



void AUD_set_active_out (SWVoiceOut *sw, int on)
@@ -998,10 +996,10 @@ static size_t audio_get_avail (SWVoiceIn *sw) 
    ldebug (

        "%s: get_avail live %d ret %" PRId64 "\n",

        SW_NAME (sw),

        live, (((int64_t) live << 32) / sw->ratio) << sw->info.shift

        live, (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame

        );



    return (((int64_t) live << 32) / sw->ratio) << sw->info.shift;

    return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame;

}



static size_t audio_get_free(SWVoiceOut *sw)
@@ -1025,10 +1023,11 @@ static size_t audio_get_free(SWVoiceOut *sw) 
#ifdef DEBUG_OUT

    dolog ("%s: get_free live %d dead %d ret %" PRId64 "\n",

           SW_NAME (sw),

           live, dead, (((int64_t) dead << 32) / sw->ratio) << sw->info.shift);

           live, dead, (((int64_t) dead << 32) / sw->ratio) *

           sw->info.bytes_per_frame);

#endif



    return (((int64_t) dead << 32) / sw->ratio) << sw->info.shift;

    return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame;

}



static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos,
@@ -1047,7 +1046,7 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, 
            while (n) {

                size_t till_end_of_hw = hw->mix_buf->size - rpos2;

                size_t to_write = MIN(till_end_of_hw, n);

                size_t bytes = to_write << hw->info.shift;

                size_t bytes = to_write * hw->info.bytes_per_frame;

                size_t written;



                sw->buf = hw->mix_buf->samples + rpos2;
@@ -1082,10 +1081,11 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) 
            return clipped + live;

        }



        decr = MIN(size >> hw->info.shift, live);

        decr = MIN(size / hw->info.bytes_per_frame, live);

        audio_pcm_hw_clip_out(hw, buf, decr);

        proc = hw->pcm_ops->put_buffer_out(hw, buf, decr << hw->info.shift) >>

            hw->info.shift;

        proc = hw->pcm_ops->put_buffer_out(hw, buf,

                                           decr * hw->info.bytes_per_frame) /

            hw->info.bytes_per_frame;



        live -= proc;

        clipped += proc;
@@ -1234,16 +1234,16 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples) 


    while (samples) {

        size_t proc;

        size_t size = samples << hw->info.shift;

        size_t size = samples * hw->info.bytes_per_frame;

        void *buf = hw->pcm_ops->get_buffer_in(hw, &size);



        assert((size & hw->info.align) == 0);

        assert(size % hw->info.bytes_per_frame == 0);

        if (size == 0) {

            hw->pcm_ops->put_buffer_in(hw, buf, size);

            break;

        }



        proc = MIN(size >> hw->info.shift,

        proc = MIN(size / hw->info.bytes_per_frame,

                   conv_buf->size - conv_buf->pos);



        hw->conv(conv_buf->samples + conv_buf->pos, buf, proc);
@@ -1251,7 +1251,7 @@ static size_t audio_pcm_hw_run_in(HWVoiceIn *hw, size_t samples) 


        samples -= proc;

        conv += proc;

        hw->pcm_ops->put_buffer_in(hw, buf, proc << hw->info.shift);

        hw->pcm_ops->put_buffer_in(hw, buf, proc * hw->info.bytes_per_frame);

    }



    return conv;
@@ -1325,7 +1325,7 @@ static void audio_run_capture (AudioState *s) 


            for (cb = cap->cb_head.lh_first; cb; cb = cb->entries.le_next) {

                cb->ops.capture (cb->opaque, cap->buf,

                                 to_capture << hw->info.shift);

                                 to_capture * hw->info.bytes_per_frame);

            }

            rpos = (rpos + to_capture) % hw->mix_buf->size;

            live -= to_capture;
@@ -1378,7 +1378,7 @@ void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size) 
    ssize_t start;



    if (unlikely(!hw->buf_emul)) {

        size_t calc_size = hw->conv_buf->size << hw->info.shift;

        size_t calc_size = hw->conv_buf->size * hw->info.bytes_per_frame;

        hw->buf_emul = g_malloc(calc_size);

        hw->size_emul = calc_size;

        hw->pos_emul = hw->pending_emul = 0;
@@ -1414,7 +1414,7 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size) 
void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size)

{

    if (unlikely(!hw->buf_emul)) {

        size_t calc_size = hw->mix_buf->size << hw->info.shift;

        size_t calc_size = hw->mix_buf->size * hw->info.bytes_per_frame;



        hw->buf_emul = g_malloc(calc_size);

        hw->size_emul = calc_size;
@@ -1833,7 +1833,7 @@ CaptureVoiceOut *AUD_add_capture( 


        audio_pcm_init_info (&hw->info, as);



        cap->buf = g_malloc0_n(hw->mix_buf->size, 1 << hw->info.shift);

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



        hw->clip = mixeng_clip

            [hw->info.nchannels == 2]
@@ -2153,14 +2153,14 @@ size_t audio_rate_get_bytes(struct audio_pcm_info *info, RateCtl *rate, 
    now = qemu_clock_get_ns(QEMU_CLOCK_VIRTUAL);

    ticks = now - rate->start_ticks;

    bytes = muldiv64(ticks, info->bytes_per_second, NANOSECONDS_PER_SECOND);

    samples = (bytes - rate->bytes_sent) >> info->shift;

    samples = (bytes - rate->bytes_sent) / info->bytes_per_frame;

    if (samples < 0 || samples > 65536) {

        AUD_log(NULL, "Resetting rate control (%" PRId64 " samples)\n", samples);

        audio_rate_start(rate);

        samples = 0;

    }



    ret = MIN(samples << info->shift, bytes_avail);

    ret = MIN(samples * info->bytes_per_frame, bytes_avail);

    rate->bytes_sent += ret;

    return ret;

}

audio/audio_int.h

+1 −2
Original line number Diff line number Diff line
@@ -43,8 +43,7 @@ struct audio_pcm_info { 
    int sign;

    int freq;

    int nchannels;

    int align;

    int shift;

    int bytes_per_frame;

    int bytes_per_second;

    int swap_endianness;

};

audio/coreaudio.c

+2 −2
Original line number Diff line number Diff line
@@ -440,7 +440,7 @@ static OSStatus audioDeviceIOProc( 
    }



    frameCount = core->audioDevicePropertyBufferFrameSize;

    pending_frames = hw->pending_emul >> hw->info.shift;

    pending_frames = hw->pending_emul / hw->info.bytes_per_frame;



    /* if there are not enough samples, set signal and return */

    if (pending_frames < frameCount) {
@@ -449,7 +449,7 @@ static OSStatus audioDeviceIOProc( 
        return 0;

    }



    len = frameCount << hw->info.shift;

    len = frameCount * hw->info.bytes_per_frame;

    while (len) {

        size_t write_len;

        ssize_t start = ((ssize_t) hw->pos_emul) - hw->pending_emul;

audio/dsound_template.h

+5 −5
Original line number Diff line number Diff line
@@ -98,8 +98,8 @@ static int glue (dsound_lock_, TYPE) ( 
        goto fail;

    }



    if ((p1p && *p1p && (*blen1p & info->align)) ||

        (p2p && *p2p && (*blen2p & info->align))) {

    if ((p1p && *p1p && (*blen1p % info->bytes_per_frame)) ||

        (p2p && *p2p && (*blen2p % info->bytes_per_frame))) {

        dolog("DirectSound returned misaligned buffer %ld %ld\n",

              *blen1p, *blen2p);

        glue(dsound_unlock_, TYPE)(buf, *p1p, p2p ? *p2p : NULL, *blen1p,
@@ -247,14 +247,14 @@ static int dsound_init_out(HWVoiceOut *hw, struct audsettings *as, 
    obt_as.endianness = 0;

    audio_pcm_init_info (&hw->info, &obt_as);



    if (bc.dwBufferBytes & hw->info.align) {

    if (bc.dwBufferBytes % hw->info.bytes_per_frame) {

        dolog (

            "GetCaps returned misaligned buffer size %ld, alignment %d\n",

            bc.dwBufferBytes, hw->info.align + 1

            bc.dwBufferBytes, hw->info.bytes_per_frame

            );

    }

    hw->size_emul = bc.dwBufferBytes;

    hw->samples = bc.dwBufferBytes >> hw->info.shift;

    hw->samples = bc.dwBufferBytes / hw->info.bytes_per_frame;

    ds->s = s;



#ifdef DEBUG_DSOUND