From: Howard Spoelstra <hsp.cat7@gmail.com>
To: "Kővágó, Zoltán" <dirty.ice.hu@gmail.com>
Cc: Markus Armbruster <armbru@redhat.com>,
qemu-devel qemu-devel <qemu-devel@nongnu.org>,
Gerd Hoffmann <kraxel@redhat.com>
Subject: Re: [RFC PATCH] audio: proper support for float samples in mixeng
Date: Sun, 2 Feb 2020 23:14:06 +0100 [thread overview]
Message-ID: <CABLmASEHUkq7sCr8_Re68u-FOVPt+nym6vqP+8HYhjqTv20AhA@mail.gmail.com> (raw)
In-Reply-To: <8a8b0b5698401b78d3c4c8ec90aef83b95babb06.1580672076.git.DirtY.iCE.hu@gmail.com>
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On Sun, Feb 2, 2020 at 8:38 PM Kővágó, Zoltán <dirty.ice.hu@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.hu@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
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next prev parent reply other threads:[~2020-02-02 22:15 UTC|newest]
Thread overview: 11+ messages / expand[flat|nested] mbox.gz Atom feed top
2020-02-02 19:38 [RFC PATCH] audio: proper support for float samples in mixeng Kővágó, Zoltán
2020-02-02 22:14 ` Howard Spoelstra [this message]
2020-02-03 6:21 ` Markus Armbruster
2020-02-03 15:34 ` Eric Blake
2020-02-04 6:48 ` Markus Armbruster
2020-02-03 8:59 ` Volker Rümelin
2020-02-03 10:00 ` Peter Maydell
2020-02-03 20:38 ` Zoltán Kővágó
2020-02-04 10:24 ` Peter Maydell
2020-02-06 13:37 ` Gerd Hoffmann
2020-03-09 18:36 ` Alexander Bulekov
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