On Sun, Feb 2, 2020 at 8:38 PM Kővágó, Zoltán <dirty.ice...@gmail.com> wrote:
> This adds proper support for float samples in mixeng by adding a new > audio format for it. > > Limitations: only native endianness is supported. > > Signed-off-by: Kővágó, Zoltán <dirty.ice...@gmail.com> > --- > > This patch is meant to be applied on top of "[PATCH] coreaudio: fix > coreaudio > playback" by Volker Rümelin, available at: > https://lists.nongnu.org/archive/html/qemu-devel/2020-02/msg00114.html > > For more information, please refer to that thread. > > --- > qapi/audio.json | 2 +- > audio/audio_int.h | 3 +- > audio/audio_template.h | 41 ++++++++++++-------- > audio/mixeng.h | 8 ++-- > audio/alsaaudio.c | 17 ++++++++ > audio/audio.c | 56 ++++++++++++++++++--------- > audio/coreaudio.c | 7 +--- > audio/mixeng.c | 88 ++++++++++++++++++++++++++---------------- > audio/paaudio.c | 9 +++++ > audio/sdlaudio.c | 28 ++++++++++++++ > 10 files changed, 180 insertions(+), 79 deletions(-) > > diff --git a/qapi/audio.json b/qapi/audio.json > index 83312b2339..d8c507cced 100644 > --- a/qapi/audio.json > +++ b/qapi/audio.json > @@ -276,7 +276,7 @@ > # Since: 4.0 > ## > { 'enum': 'AudioFormat', > - 'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32' ] } > + 'data': [ 'u8', 's8', 'u16', 's16', 'u32', 's32', 'f32' ] } > > ## > # @AudiodevDriver: > diff --git a/audio/audio_int.h b/audio/audio_int.h > index 5ba2078346..cd92e48163 100644 > --- a/audio/audio_int.h > +++ b/audio/audio_int.h > @@ -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; > diff --git a/audio/audio_template.h b/audio/audio_template.h > index 0336d2670c..7013d3041f 100644 > --- a/audio/audio_template.h > +++ b/audio/audio_template.h > @@ -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 > + sw->conv = mixeng_conv_float[sw->info.nchannels == 2]; > #else > - sw->clip = mixeng_clip > + sw->clip = mixeng_clip_float[sw->info.nchannels == 2]; > #endif > - [sw->info.nchannels == 2] > - [sw->info.sign] > - [sw->info.swap_endianness] > - [audio_bits_to_index (sw->info.bits)]; > + } else { > +#ifdef DAC > + sw->conv = mixeng_conv > +#else > + sw->clip = mixeng_clip > +#endif > + [sw->info.nchannels == 2] > + [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 > + hw->clip = mixeng_clip > #else > - hw->conv = mixeng_conv > + hw->conv = mixeng_conv > #endif > - [hw->info.nchannels == 2] > - [hw->info.sign] > - [hw->info.swap_endianness] > - [audio_bits_to_index (hw->info.bits)]; > + [hw->info.nchannels == 2] > + [hw->info.is_signed] > + [hw->info.swap_endianness] > + [audio_bits_to_index(hw->info.bits)]; > + } > > glue(audio_pcm_hw_alloc_resources_, TYPE)(hw); > > diff --git a/audio/mixeng.h b/audio/mixeng.h > index 7ef61763e8..2dcd6df245 100644 > --- a/audio/mixeng.h > +++ b/audio/mixeng.h > @@ -38,13 +38,13 @@ typedef struct st_sample st_sample; > typedef void (t_sample) (struct st_sample *dst, const void *src, int > samples); > typedef void (f_sample) (void *dst, const struct st_sample *src, int > samples); > > +/* indices: [stereo][signed][swap endiannes][8, 16 or 32-bits] */ > extern t_sample *mixeng_conv[2][2][2][3]; > extern f_sample *mixeng_clip[2][2][2][3]; > > -void conv_natural_float_to_stereo(struct st_sample *dst, const void *src, > - int samples); > -void clip_natural_float_from_stereo(void *dst, const struct st_sample > *src, > - int samples); > +/* indices: [stereo] */ > +extern t_sample *mixeng_conv_float[2]; > +extern f_sample *mixeng_clip_float[2]; > > void *st_rate_start (int inrate, int outrate); > void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf, > diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c > index f37ce1ce85..768b896a93 100644 > --- a/audio/alsaaudio.c > +++ b/audio/alsaaudio.c > @@ -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; > diff --git a/audio/audio.c b/audio/audio.c > index f63f39769a..53fdb42ec7 100644 > --- a/audio/audio.c > +++ b/audio/audio.c > @@ -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 > > @@ -1837,11 +1852,15 @@ CaptureVoiceOut *AUD_add_capture( > > cap->buf = g_malloc0_n(hw->mix_buf->size, > hw->info.bytes_per_frame); > > - hw->clip = mixeng_clip > - [hw->info.nchannels == 2] > - [hw->info.sign] > - [hw->info.swap_endianness] > - [audio_bits_to_index (hw->info.bits)]; > + 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.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); > @@ -2080,6 +2099,7 @@ 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: > diff --git a/audio/coreaudio.c b/audio/coreaudio.c > index 0049db97fa..f1a009610c 100644 > --- a/audio/coreaudio.c > +++ b/audio/coreaudio.c > @@ -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); > diff --git a/audio/mixeng.c b/audio/mixeng.c > index 16b646d48c..c14b0d874c 100644 > --- a/audio/mixeng.c > +++ b/audio/mixeng.c > @@ -267,55 +267,77 @@ f_sample *mixeng_clip[2][2][2][3] = { > } > }; > > -void conv_natural_float_to_stereo(struct st_sample *dst, const void *src, > - int samples) > +#ifdef FLOAT_MIXENG > +#define FLOAT_CONV_TO(x) (x) > +#define FLOAT_CONV_FROM(x) (x) > +#else > +static const float float_scale = UINT_MAX; > +#define FLOAT_CONV_TO(x) ((x) * float_scale) > + > +#ifdef RECIPROCAL > +static const float float_scale_reciprocal = 1.f / UINT_MAX; > +#define FLOAT_CONV_FROM(x) ((x) * float_scale_reciprocal) > +#else > +#define FLOAT_CONV_FROM(x) ((x) / float_scale) > +#endif > +#endif > + > +static void conv_natural_float_to_mono(struct st_sample *dst, const void > *src, > + int samples) > { > float *in = (float *)src; > -#ifndef FLOAT_MIXENG > - const float scale = UINT_MAX; > -#endif > > while (samples--) { > -#ifdef FLOAT_MIXENG > - dst->l = *in++; > - dst->r = *in++; > -#else > - dst->l = *in++ * scale; > - dst->r = *in++ * scale; > -#endif > + dst->r = dst->l = FLOAT_CONV_TO(*in++); > + dst++; > + } > +} > + > +static void conv_natural_float_to_stereo(struct st_sample *dst, const > void *src, > + int samples) > +{ > + float *in = (float *)src; > + > + while (samples--) { > + dst->l = FLOAT_CONV_TO(*in++); > + dst->r = FLOAT_CONV_TO(*in++); > dst++; > } > } > > -void clip_natural_float_from_stereo(void *dst, const struct st_sample > *src, > - int samples) > +t_sample *mixeng_conv_float[2] = { > + conv_natural_float_to_mono, > + conv_natural_float_to_stereo, > +}; > + > +static void clip_natural_float_from_mono(void *dst, const struct > st_sample *src, > + int samples) > { > float *out = (float *)dst; > -#ifndef FLOAT_MIXENG > -#ifdef RECIPROCAL > - const float scale = 1.f / UINT_MAX; > -#else > - const float scale = UINT_MAX; > -#endif > -#endif > > while (samples--) { > -#ifdef FLOAT_MIXENG > - *out++ = src->l; > - *out++ = src->r; > -#else > -#ifdef RECIPROCAL > - *out++ = src->l * scale; > - *out++ = src->r * scale; > -#else > - *out++ = src->l / scale; > - *out++ = src->r / scale; > -#endif > -#endif > + *out++ = FLOAT_CONV_FROM(src->l) + FLOAT_CONV_FROM(src->r); > + src++; > + } > +} > + > +static void clip_natural_float_from_stereo( > + void *dst, const struct st_sample *src, int samples) > +{ > + float *out = (float *)dst; > + > + while (samples--) { > + *out++ = FLOAT_CONV_FROM(src->l); > + *out++ = FLOAT_CONV_FROM(src->r); > src++; > } > } > > +f_sample *mixeng_clip_float[2] = { > + clip_natural_float_from_mono, > + clip_natural_float_from_stereo, > +}; > + > void audio_sample_to_uint64(void *samples, int pos, > uint64_t *left, uint64_t *right) > { > diff --git a/audio/paaudio.c b/audio/paaudio.c > index dbfe48c03a..1278c5a775 100644 > --- a/audio/paaudio.c > +++ b/audio/paaudio.c > @@ -279,6 +279,9 @@ static pa_sample_format_t audfmt_to_pa (AudioFormat > afmt, int endianness) > case AUDIO_FORMAT_U32: > format = endianness ? PA_SAMPLE_S32BE : PA_SAMPLE_S32LE; > break; > + case AUDIO_FORMAT_F32: > + format = endianness ? PA_SAMPLE_FLOAT32BE : PA_SAMPLE_FLOAT32LE; > + break; > default: > dolog ("Internal logic error: Bad audio format %d\n", afmt); > format = PA_SAMPLE_U8; > @@ -304,6 +307,12 @@ static AudioFormat pa_to_audfmt (pa_sample_format_t > fmt, int *endianness) > case PA_SAMPLE_S32LE: > *endianness = 0; > return AUDIO_FORMAT_S32; > + case PA_SAMPLE_FLOAT32BE: > + *endianness = 1; > + return AUDIO_FORMAT_F32; > + case PA_SAMPLE_FLOAT32LE: > + *endianness = 0; > + return AUDIO_FORMAT_F32; > default: > dolog ("Internal logic error: Bad pa_sample_format %d\n", fmt); > return AUDIO_FORMAT_U8; > diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c > index 5c6bcfcb3e..6af1911db9 100644 > --- a/audio/sdlaudio.c > +++ b/audio/sdlaudio.c > @@ -77,6 +77,14 @@ static int aud_to_sdlfmt (AudioFormat fmt) > case AUDIO_FORMAT_U16: > return AUDIO_U16LSB; > > + case AUDIO_FORMAT_S32: > + return AUDIO_S32LSB; > + > + /* no unsigned 32-bit support in SDL */ > + > + case AUDIO_FORMAT_F32: > + return AUDIO_F32LSB; > + > default: > dolog ("Internal logic error: Bad audio format %d\n", fmt); > #ifdef DEBUG_AUDIO > @@ -119,6 +127,26 @@ static int sdl_to_audfmt(int sdlfmt, AudioFormat > *fmt, int *endianness) > *fmt = AUDIO_FORMAT_U16; > break; > > + case AUDIO_S32LSB: > + *endianness = 0; > + *fmt = AUDIO_FORMAT_S32; > + break; > + > + case AUDIO_S32MSB: > + *endianness = 1; > + *fmt = AUDIO_FORMAT_S32; > + break; > + > + case AUDIO_F32LSB: > + *endianness = 0; > + *fmt = AUDIO_FORMAT_F32; > + break; > + > + case AUDIO_F32MSB: > + *endianness = 1; > + *fmt = AUDIO_FORMAT_F32; > + break; > + > default: > dolog ("Unrecognized SDL audio format %d\n", sdlfmt); > return -1; > -- > 2.25.0 > > > Hi, I applied the 2 patches to https://github.com/mcayland/qemu/tree/screamer to test audio support in qemu-system-ppc running Mac OS 9.2 and OSX 10.5. Host is OSX Sierra. Coreaudio seems happy with them. Best, Howard
