[PULL,12/12] audio: proper support for float samples in mixeng

Message ID	20200207074557.26073-13-kraxel@redhat.com (mailing list archive)
State	New, archived
Headers	show Return-Path: <SRS0=8NHN=33=nongnu.org=qemu-devel-bounces+patchwork-qemu-devel=patchwork.kernel.org@kernel.org> DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org 7CA1021741 From: Gerd Hoffmann <kraxel@redhat.com> To: qemu-devel@nongnu.org Subject: [PULL 12/12] audio: proper support for float samples in mixeng Date: Fri, 7 Feb 2020 08:45:57 +0100 Message-Id: <20200207074557.26073-13-kraxel@redhat.com> In-Reply-To: <20200207074557.26073-1-kraxel@redhat.com> References: <20200207074557.26073-1-kraxel@redhat.com> MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: quoted-printable Precedence: list Cc: K =?utf-8?b?xZE=?= v =?utf-8?b?w6E=?= g =?utf-8?b?w7M=?= @redhat.com, =?utf-8?q?Zolt=C3=A1n?= <DirtY.iCE.hu@gmail.com>, Markus Armbruster <armbru@redhat.com>, Gerd Hoffmann <kraxel@redhat.com> Errors-To: qemu-devel-bounces+patchwork-qemu-devel=patchwork.kernel.org@nongnu.org Sender: "Qemu-devel" <qemu-devel-bounces+patchwork-qemu-devel=patchwork.kernel.org@nongnu.org>
Series	[PULL,01/12] audio/oss: fix buffer pos calculation \| expand [PULL,01/12] audio/oss: fix buffer pos calculation [PULL,02/12] audio: fix audio_generic_write [PULL,03/12] audio: fix audio_generic_read [PULL,04/12] paaudio: remove unused variables [PULL,05/12] audio: prevent SIGSEGV in AUD_get_buffer_size_out [PULL,06/12] audio: fix bug 1858488 [PULL,07/12] ossaudio: prevent SIGSEGV in oss_enable_out [PULL,08/12] ossaudio: disable poll mode can't be reached [PULL,09/12] audio: audio_generic_get_buffer_in should honor *size [PULL,10/12] audio/dsound: fix invalid parameters error [PULL,11/12] coreaudio: fix coreaudio playback [PULL,12/12] audio: proper support for float samples in mixeng

diff --git a/audio/audio_int.h b/audio/audio_int.h index 3c8e48b55b1c..4775857bf245 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 0336d2670cf6..7013d3041f91 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 7ef61763e8f9..2dcd6df24561 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 4ef26818be57..a23a5a0b60a1 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 3bfd808bc6f1..9ac9a20c41ba 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 @@ -1832,11 +1847,15 @@ 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); @@ -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: diff --git a/audio/coreaudio.c b/audio/coreaudio.c index e3620b274bd6..4b4365660fcf 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 16b646d48cdf..c14b0d874ce5 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 8f37c6185161..b05208469831 100644 --- a/audio/paaudio.c +++ b/audio/paaudio.c @@ -277,6 +277,9 @@ 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; @@ -302,6 +305,12 @@ 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 c00e7d784523..21b7a0484bec 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; diff --git a/qapi/audio.json b/qapi/audio.json index 83312b23391e..d8c507ccedae 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:

[PULL,12/12] audio: proper support for float samples in mixeng

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