From cc99139bae83cdc6116d6435b594837d54bd035c Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Marc-Andr=C3=A9=20Lureau?= Date: Tue, 7 Feb 2023 12:56:10 +0400 Subject: [PATCH 01/27] MAINTAINERS: add myself to ui/ and audio/ MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Helping out with patch review & queue handling. Signed-off-by: Marc-André Lureau Reviewed-by: Gerd Hoffmann Reviewed-by: Philippe Mathieu-Daudé Message-Id: <20230207085610.1033536-1-marcandre.lureau@redhat.com> --- MAINTAINERS | 2 ++ 1 file changed, 2 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 011fd85a09..da29661b37 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2490,6 +2490,7 @@ Subsystems ---------- Overall Audio backends M: Gerd Hoffmann +M: Marc-André Lureau S: Odd Fixes F: audio/ X: audio/alsaaudio.c @@ -2785,6 +2786,7 @@ F: docs/spice-port-fqdn.txt Graphics M: Gerd Hoffmann +M: Marc-André Lureau S: Odd Fixes F: ui/ F: include/ui/ From b9ae74e2e497154f77b109743c77fa97c71d71c1 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:25 +0100 Subject: [PATCH 02/27] audio: log unimplemented audio device sample rates MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Some emulated audio devices allow guests to select very low sample rates that the audio subsystem doesn't support. The lowest supported sample rate depends on the audio backend used and in most cases can be changed with various -audiodev arguments. Until now, the audio_bug function emits an error message similar to the following error message A bug was just triggered in audio_calloc Save all your work and restart without audio I am sorry Context: audio_pcm_sw_alloc_resources_out passed invalid arguments to audio_calloc nmemb=0 size=16 (len=0) audio: Could not allocate buffer for `ac97.po' (0 samples) and the audio subsystem continues without sound for the affected device. The fact that the selected sample rate is not supported is not a guest error. Instead of displaying an error message, the missing audio support is now logged. Simply continuing without sound is correct, since the audio stream won't transport anything reasonable at such high resample ratios anyway. The AUD_open_* functions return NULL like before. The opened audio device will not be registered in the audio subsystem and consequently the audio frontend callback functions will not be called. The AUD_read and AUD_write functions return early in this case. This is necessary because, for example, the Sound Blaster 16 emulation calls AUD_write from the DMA callback function. Acked-by: Christian Schoenebeck Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-1-vr_qemu@t-online.de> --- audio/audio.c | 1 + audio/audio_template.h | 13 +++++++++++++ 2 files changed, 14 insertions(+) diff --git a/audio/audio.c b/audio/audio.c index 4290309d18..81f5c0bb1e 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -33,6 +33,7 @@ #include "qapi/qapi-visit-audio.h" #include "qapi/qapi-commands-audio.h" #include "qemu/cutils.h" +#include "qemu/log.h" #include "qemu/module.h" #include "qemu/help_option.h" #include "sysemu/sysemu.h" diff --git a/audio/audio_template.h b/audio/audio_template.h index 42b4712acb..dbfb4fee4c 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -115,6 +115,19 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) #else samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; #endif + if (samples == 0) { + HW *hw = sw->hw; + size_t f_fe_min; + + /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ + f_fe_min = (hw->info.freq + HWBUF->size - 1) / HWBUF->size; + qemu_log_mask(LOG_UNIMP, + AUDIO_CAP ": The guest selected a " NAME " sample rate" + " of %d Hz for %s. Only sample rates >= %zu Hz are" + " supported.\n", + sw->info.freq, sw->name, f_fe_min); + return -1; + } sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample)); if (!sw->buf) { From 90394fe15f95f58ad34684541cd480422b31d0af Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:26 +0100 Subject: [PATCH 03/27] audio: don't show unnecessary error messages MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Let the audio_pcm_create_voice_pair_* functions handle error reporting. This avoids an additional error message in case the guest selected an unimplemented sample rate. Reviewed-by: Philippe Mathieu-Daudé Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-2-vr_qemu@t-online.de> --- audio/audio_template.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/audio/audio_template.h b/audio/audio_template.h index dbfb4fee4c..f0ef262ab3 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -441,6 +441,7 @@ static SW *glue(audio_pcm_create_voice_pair_, TYPE)( hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as); if (!hw) { + dolog("Could not create a backend for voice `%s'\n", sw_name); goto err2; } @@ -540,7 +541,6 @@ SW *glue (AUD_open_, TYPE) ( } else { sw = glue(audio_pcm_create_voice_pair_, TYPE)(s, name, as); if (!sw) { - dolog ("Failed to create voice `%s'\n", name); return NULL; } } From b637a61c6f27f8023905223314fe18819540162f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:27 +0100 Subject: [PATCH 04/27] audio: rename hardware store to backend MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Use a consistent friendly name for the HWVoiceOut and HWVoiceIn structures. Reviewed-by: Thomas Huth Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Reviewed-by: Philippe Mathieu-Daudé Message-Id: <20230121094735.11644-3-vr_qemu@t-online.de> --- audio/audio_template.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/audio/audio_template.h b/audio/audio_template.h index f0ef262ab3..33af42ed8b 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -529,8 +529,8 @@ SW *glue (AUD_open_, TYPE) ( HW *hw = sw->hw; if (!hw) { - dolog ("Internal logic error voice `%s' has no hardware store\n", - SW_NAME (sw)); + dolog("Internal logic error: voice `%s' has no backend\n", + SW_NAME(sw)); goto fail; } From f8f8a8ac57723a0c3c06d18ed69e1cd209db5309 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:28 +0100 Subject: [PATCH 05/27] audio: remove unused #define AUDIO_STRINGIFY MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Remove the unused #define AUDIO_STRINGIFY. It was last used before commit 470bcabd8f ("audio: Replace AUDIO_FUNC with __func__"). Reviewed-by: Philippe Mathieu-Daudé Reviewed-by: Thomas Huth Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-4-vr_qemu@t-online.de> --- audio/audio_int.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/audio/audio_int.h b/audio/audio_int.h index e87ce014a0..4632cdf9cc 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -294,9 +294,6 @@ static inline size_t audio_ring_posb(size_t pos, size_t dist, size_t len) #define ldebug(fmt, ...) (void)0 #endif -#define AUDIO_STRINGIFY_(n) #n -#define AUDIO_STRINGIFY(n) AUDIO_STRINGIFY_(n) - typedef struct AudiodevListEntry { Audiodev *dev; QSIMPLEQ_ENTRY(AudiodevListEntry) next; From 25bf0c2d35f7caa2cab212340aea95e8245eae28 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:29 +0100 Subject: [PATCH 06/27] audio/mixeng: use g_new0() instead of audio_calloc() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Replace audio_calloc() with the equivalent g_new0(). With a n_structs argument of 1, g_new0() never returns NULL. Also remove the unnecessary NULL checks. Reviewed-by: Richard Henderson Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-5-vr_qemu@t-online.de> --- audio/audio.c | 5 ----- audio/audio_template.h | 6 +----- audio/mixeng.c | 7 +------ 3 files changed, 2 insertions(+), 16 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 81f5c0bb1e..012d10996b 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -509,11 +509,6 @@ static int audio_attach_capture (HWVoiceOut *hw) sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq; sw->vol = nominal_volume; sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq); - if (!sw->rate) { - dolog ("Could not start rate conversion for `%s'\n", SW_NAME (sw)); - g_free (sw); - return -1; - } QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries); QLIST_INSERT_HEAD (&hw->cap_head, sc, entries); #ifdef DEBUG_CAPTURE diff --git a/audio/audio_template.h b/audio/audio_template.h index 33af42ed8b..dfa440f778 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -141,11 +141,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) #else sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq); #endif - if (!sw->rate) { - g_free (sw->buf); - sw->buf = NULL; - return -1; - } + return 0; } diff --git a/audio/mixeng.c b/audio/mixeng.c index 100a306d6f..fe454e0725 100644 --- a/audio/mixeng.c +++ b/audio/mixeng.c @@ -414,12 +414,7 @@ struct rate { */ void *st_rate_start (int inrate, int outrate) { - struct rate *rate = audio_calloc(__func__, 1, sizeof(*rate)); - - if (!rate) { - dolog ("Could not allocate resampler (%zu bytes)\n", sizeof (*rate)); - return NULL; - } + struct rate *rate = g_new0(struct rate, 1); rate->opos = 0; From d1def19fa3689127f551f8740e5c8b3b743dc33e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:30 +0100 Subject: [PATCH 07/27] audio/alsaaudio: use g_new0() instead of audio_calloc() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Replace audio_calloc() with the equivalent g_new0(). The value of the g_new0() argument count is >= 1, which means g_new0() will never return NULL. Also remove the unnecessary NULL check. Reviewed-by: Philippe Mathieu-Daudé Reviewed-by: Richard Henderson Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-6-vr_qemu@t-online.de> --- audio/alsaaudio.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 714bfb6453..5f50dfa0bf 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -222,11 +222,7 @@ static int alsa_poll_helper (snd_pcm_t *handle, struct pollhlp *hlp, int mask) return -1; } - pfds = audio_calloc ("alsa_poll_helper", count, sizeof (*pfds)); - if (!pfds) { - dolog ("Could not initialize poll mode\n"); - return -1; - } + pfds = g_new0(struct pollfd, count); err = snd_pcm_poll_descriptors (handle, pfds, count); if (err < 0) { From 3724ab3b34ff3430f315272429188a1b3dfdd247 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:31 +0100 Subject: [PATCH 08/27] audio/audio_template: use g_malloc0() to replace audio_calloc() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Use g_malloc0() as a direct replacement for audio_calloc(). Since the type of the parameter n_bytes of the function g_malloc0() is unsigned, the type of the variables voice_size_out and voice_size_in has been changed to size_t. This means that the function argument no longer has to be checked for negative values. Signed-off-by: Volker Rümelin Reviewed-by: Daniel P. Berrangé Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-7-vr_qemu@t-online.de> --- audio/audio_int.h | 4 ++-- audio/audio_template.h | 18 ++++++++---------- 2 files changed, 10 insertions(+), 12 deletions(-) diff --git a/audio/audio_int.h b/audio/audio_int.h index 4632cdf9cc..ce2d6bf92c 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -151,8 +151,8 @@ struct audio_driver { int can_be_default; int max_voices_out; int max_voices_in; - int voice_size_out; - int voice_size_in; + size_t voice_size_out; + size_t voice_size_in; QLIST_ENTRY(audio_driver) next; }; diff --git a/audio/audio_template.h b/audio/audio_template.h index dfa440f778..592866f14a 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -40,7 +40,7 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, struct audio_driver *drv) { int max_voices = glue (drv->max_voices_, TYPE); - int voice_size = glue (drv->voice_size_, TYPE); + size_t voice_size = glue(drv->voice_size_, TYPE); if (glue (s->nb_hw_voices_, TYPE) > max_voices) { if (!max_voices) { @@ -63,8 +63,8 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, } if (audio_bug(__func__, voice_size && !max_voices)) { - dolog ("drv=`%s' voice_size=%d max_voices=0\n", - drv->name, voice_size); + dolog("drv=`%s' voice_size=%zu max_voices=0\n", + drv->name, voice_size); } } @@ -273,13 +273,11 @@ static HW *glue(audio_pcm_hw_add_new_, TYPE)(AudioState *s, return NULL; } - hw = audio_calloc(__func__, 1, glue(drv->voice_size_, TYPE)); - if (!hw) { - dolog ("Can not allocate voice `%s' size %d\n", - drv->name, glue (drv->voice_size_, TYPE)); - return NULL; - } - + /* + * Since glue(s->nb_hw_voices_, TYPE) is != 0, glue(drv->voice_size_, TYPE) + * is guaranteed to be != 0. See the audio_init_nb_voices_* functions. + */ + hw = g_malloc0(glue(drv->voice_size_, TYPE)); hw->s = s; hw->pcm_ops = drv->pcm_ops; From c6b69a814a2dd9f8da08d55f36ed73597ee0c612 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:32 +0100 Subject: [PATCH 09/27] audio/audio_template: use g_new0() to replace audio_calloc() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Replace audio_calloc() with the equivalent g_new0(). With a n_structs argument >= 1, g_new0() never returns NULL. Also remove the unnecessary NULL checks. Signed-off-by: Volker Rümelin Reviewed-by: Daniel P. Berrangé Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-8-vr_qemu@t-online.de> --- audio/audio_template.h | 29 ++++++++++++----------------- 1 file changed, 12 insertions(+), 17 deletions(-) diff --git a/audio/audio_template.h b/audio/audio_template.h index 592866f14a..980e1f4bd0 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -115,6 +115,12 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) #else samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; #endif + if (audio_bug(__func__, samples < 0)) { + dolog("Can not allocate buffer for `%s' (%d samples)\n", + SW_NAME(sw), samples); + return -1; + } + if (samples == 0) { HW *hw = sw->hw; size_t f_fe_min; @@ -129,12 +135,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) return -1; } - sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample)); - if (!sw->buf) { - dolog ("Could not allocate buffer for `%s' (%d samples)\n", - SW_NAME (sw), samples); - return -1; - } + sw->buf = g_new0(st_sample, samples); #ifdef DAC sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq); @@ -425,34 +426,28 @@ static SW *glue(audio_pcm_create_voice_pair_, TYPE)( hw_as = *as; } - sw = audio_calloc(__func__, 1, sizeof(*sw)); - if (!sw) { - dolog ("Could not allocate soft voice `%s' (%zu bytes)\n", - sw_name ? sw_name : "unknown", sizeof (*sw)); - goto err1; - } + sw = g_new0(SW, 1); sw->s = s; hw = glue(audio_pcm_hw_add_, TYPE)(s, &hw_as); if (!hw) { dolog("Could not create a backend for voice `%s'\n", sw_name); - goto err2; + goto err1; } glue (audio_pcm_hw_add_sw_, TYPE) (hw, sw); if (glue (audio_pcm_sw_init_, TYPE) (sw, hw, sw_name, as)) { - goto err3; + goto err2; } return sw; -err3: +err2: glue (audio_pcm_hw_del_sw_, TYPE) (sw); glue (audio_pcm_hw_gc_, TYPE) (&hw); -err2: - g_free (sw); err1: + g_free(sw); return NULL; } From 2d2ccb6060a12380644d2027252bc2e0ae145daf Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:33 +0100 Subject: [PATCH 10/27] audio: remove audio_calloc() function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Now that the last call site of audio_calloc() was removed, remove the unused audio_calloc() function. Reviewed-by: Philippe Mathieu-Daudé Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-9-vr_qemu@t-online.de> --- audio/audio.c | 20 -------------------- audio/audio_int.h | 1 - 2 files changed, 21 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 012d10996b..772c3cc320 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -149,26 +149,6 @@ static inline int audio_bits_to_index (int bits) } } -void *audio_calloc (const char *funcname, int nmemb, size_t size) -{ - int cond; - size_t len; - - len = nmemb * size; - cond = !nmemb || !size; - cond |= nmemb < 0; - cond |= len < size; - - if (audio_bug ("audio_calloc", cond)) { - AUD_log (NULL, "%s passed invalid arguments to audio_calloc\n", - funcname); - AUD_log (NULL, "nmemb=%d size=%zu (len=%zu)\n", nmemb, size, len); - return NULL; - } - - return g_malloc0 (len); -} - void AUD_vlog (const char *cap, const char *fmt, va_list ap) { if (cap) { diff --git a/audio/audio_int.h b/audio/audio_int.h index ce2d6bf92c..5028f2354a 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -251,7 +251,6 @@ void audio_pcm_init_info (struct audio_pcm_info *info, struct audsettings *as); void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len); int audio_bug (const char *funcname, int cond); -void *audio_calloc (const char *funcname, int nmemb, size_t size); void audio_run(AudioState *s, const char *msg); From 467447320a1c28448e48bdcbfa939654729b28d1 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:34 +0100 Subject: [PATCH 11/27] alsaaudio: change default playback settings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The currently used default playback settings in the ALSA audio backend are a bit unfortunate. With a few emulated audio devices, audio playback does not work properly. Here is a short part of the debug log while audio is playing (elapsed time in seconds). audio: Elapsed since last alsa run (running): 0.046244 audio: Elapsed since last alsa run (running): 0.023137 audio: Elapsed since last alsa run (running): 0.023170 audio: Elapsed since last alsa run (running): 0.023650 audio: Elapsed since last alsa run (running): 0.060802 audio: Elapsed since last alsa run (running): 0.031931 For some audio devices the time of more than 23ms between updates is too long. Set the period time to 5.8ms so that the maximum time between two updates typically does not exceed 11ms. This roughly matches the 10ms period time when doing playback with the audio timer. After this patch the debug log looks like this. audio: Elapsed since last alsa run (running): 0.011919 audio: Elapsed since last alsa run (running): 0.005788 audio: Elapsed since last alsa run (running): 0.005995 audio: Elapsed since last alsa run (running): 0.011069 audio: Elapsed since last alsa run (running): 0.005901 audio: Elapsed since last alsa run (running): 0.006084 Acked-by: Christian Schoenebeck Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-10-vr_qemu@t-online.de> --- audio/alsaaudio.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 5f50dfa0bf..0cc982e61f 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -913,17 +913,14 @@ static void *alsa_audio_init(Audiodev *dev) alsa_init_per_direction(aopts->in); alsa_init_per_direction(aopts->out); - /* - * need to define them, as otherwise alsa produces no sound - * doesn't set has_* so alsa_open can identify it wasn't set by the user - */ + /* don't set has_* so alsa_open can identify it wasn't set by the user */ if (!dev->u.alsa.out->has_period_length) { - /* 1024 frames assuming 44100Hz */ - dev->u.alsa.out->period_length = 1024 * 1000000 / 44100; + /* 256 frames assuming 44100Hz */ + dev->u.alsa.out->period_length = 5805; } if (!dev->u.alsa.out->has_buffer_length) { /* 4096 frames assuming 44100Hz */ - dev->u.alsa.out->buffer_length = 4096ll * 1000000 / 44100; + dev->u.alsa.out->buffer_length = 92880; } /* From 5140ad8279c746182c80fa867e0c3674250df5bc Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Sat, 21 Jan 2023 10:47:35 +0100 Subject: [PATCH 12/27] alsaaudio: reintroduce default recording settings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Audio recording with ALSA default settings currently doesn't work. The debug log shows updates every 0.75s and 1.5s. audio: Elapsed since last alsa run (running): 0.743030 audio: Elapsed since last alsa run (running): 1.486048 audio: Elapsed since last alsa run (running): 0.743008 audio: Elapsed since last alsa run (running): 1.485878 audio: Elapsed since last alsa run (running): 1.486040 audio: Elapsed since last alsa run (running): 1.485886 The time between updates should be in the 10ms range. Audio recording with ALSA has the same timing contraints as playback. Reintroduce the default recording settings and use the same default settings for recording as for playback. The term "reintroduce" is correct because commit a93f328177 ("alsaaudio: port to -audiodev config") removed the default settings for recording. Signed-off-by: Volker Rümelin Reviewed-by: Marc-André Lureau Message-Id: <20230121094735.11644-11-vr_qemu@t-online.de> --- audio/alsaaudio.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c index 0cc982e61f..057571dd1e 100644 --- a/audio/alsaaudio.c +++ b/audio/alsaaudio.c @@ -923,15 +923,13 @@ static void *alsa_audio_init(Audiodev *dev) dev->u.alsa.out->buffer_length = 92880; } - /* - * OptsVisitor sets unspecified optional fields to zero, but do not depend - * on it... - */ if (!dev->u.alsa.in->has_period_length) { - dev->u.alsa.in->period_length = 0; + /* 256 frames assuming 44100Hz */ + dev->u.alsa.in->period_length = 5805; } if (!dev->u.alsa.in->has_buffer_length) { - dev->u.alsa.in->buffer_length = 0; + /* 4096 frames assuming 44100Hz */ + dev->u.alsa.in->buffer_length = 92880; } return dev; From 8dbd3d1795da6f39e90a911e9acaae74c9ff120d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:41 +0100 Subject: [PATCH 13/27] audio: change type of mix_buf and conv_buf MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Change the type of mix_buf in struct HWVoiceOut and conv_buf in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer. However, a buffer pointer is still needed. For this reason in struct STSampleBuffer samples[] is changed to *buffer. This is a preparation for the next patch. The next patch will add this line, which is not possible with the current struct STSampleBuffer definition. + sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; There are no functional changes. Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-1-vr_qemu@t-online.de> --- audio/audio.c | 106 ++++++++++++++++++++--------------------- audio/audio_int.h | 6 +-- audio/audio_template.h | 19 ++++---- 3 files changed, 67 insertions(+), 64 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 772c3cc320..a0b54e4a2e 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -523,8 +523,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw) static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) { size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw); - if (audio_bug(__func__, live > hw->conv_buf->size)) { - dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); + if (audio_bug(__func__, live > hw->conv_buf.size)) { + dolog("live=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); return 0; } return live; @@ -533,13 +533,13 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples) { size_t conv = 0; - STSampleBuffer *conv_buf = hw->conv_buf; + STSampleBuffer *conv_buf = &hw->conv_buf; while (samples) { uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame); size_t proc = MIN(samples, conv_buf->size - conv_buf->pos); - hw->conv(conv_buf->samples + conv_buf->pos, src, proc); + hw->conv(conv_buf->buffer + conv_buf->pos, src, proc); conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size; samples -= proc; conv += proc; @@ -561,12 +561,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) if (!live) { return 0; } - if (audio_bug(__func__, live > hw->conv_buf->size)) { - dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); + if (audio_bug(__func__, live > hw->conv_buf.size)) { + dolog("live_in=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); return 0; } - rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size); + rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size); samples = size / sw->info.bytes_per_frame; @@ -574,11 +574,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) swlim = MIN (swlim, samples); while (swlim) { - src = hw->conv_buf->samples + rpos; - if (hw->conv_buf->pos > rpos) { - isamp = hw->conv_buf->pos - rpos; + src = hw->conv_buf.buffer + rpos; + if (hw->conv_buf.pos > rpos) { + isamp = hw->conv_buf.pos - rpos; } else { - isamp = hw->conv_buf->size - rpos; + isamp = hw->conv_buf.size - rpos; } if (!isamp) { @@ -588,7 +588,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) st_rate_flow (sw->rate, src, dst, &isamp, &osamp); swlim -= osamp; - rpos = (rpos + isamp) % hw->conv_buf->size; + rpos = (rpos + isamp) % hw->conv_buf.size; dst += osamp; ret += osamp; total += isamp; @@ -636,8 +636,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) if (nb_live1) { size_t live = smin; - if (audio_bug(__func__, live > hw->mix_buf->size)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); + if (audio_bug(__func__, live > hw->mix_buf.size)) { + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); return 0; } return live; @@ -654,17 +654,17 @@ static size_t audio_pcm_hw_get_free(HWVoiceOut *hw) static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) { size_t clipped = 0; - size_t pos = hw->mix_buf->pos; + size_t pos = hw->mix_buf.pos; while (len) { - st_sample *src = hw->mix_buf->samples + pos; + st_sample *src = hw->mix_buf.buffer + pos; 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_till_end_of_buf = hw->mix_buf.size - pos; size_t samples_to_clip = MIN(len, samples_till_end_of_buf); hw->clip(dst, src, samples_to_clip); - pos = (pos + samples_to_clip) % hw->mix_buf->size; + pos = (pos + samples_to_clip) % hw->mix_buf.size; len -= samples_to_clip; clipped += samples_to_clip; } @@ -683,11 +683,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) return size; } - hwsamples = sw->hw->mix_buf->size; + hwsamples = sw->hw->mix_buf.size; live = sw->total_hw_samples_mixed; if (audio_bug(__func__, live > hwsamples)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples); + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples); return 0; } @@ -698,7 +698,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) return 0; } - wpos = (sw->hw->mix_buf->pos + live) % hwsamples; + wpos = (sw->hw->mix_buf.pos + live) % hwsamples; dead = hwsamples - live; hw_free = audio_pcm_hw_get_free(sw->hw); @@ -725,7 +725,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) st_rate_flow_mix ( sw->rate, sw->buf + pos, - sw->hw->mix_buf->samples + wpos, + sw->hw->mix_buf.buffer + wpos, &isamp, &osamp ); @@ -989,9 +989,9 @@ static size_t audio_get_avail (SWVoiceIn *sw) } live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired; - if (audio_bug(__func__, live > sw->hw->conv_buf->size)) { - dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live, - sw->hw->conv_buf->size); + if (audio_bug(__func__, live > sw->hw->conv_buf.size)) { + dolog("live=%zu sw->hw->conv_buf.size=%zu\n", live, + sw->hw->conv_buf.size); return 0; } @@ -1026,13 +1026,13 @@ static size_t audio_get_free(SWVoiceOut *sw) live = sw->total_hw_samples_mixed; - if (audio_bug(__func__, live > sw->hw->mix_buf->size)) { - dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live, - sw->hw->mix_buf->size); + if (audio_bug(__func__, live > sw->hw->mix_buf.size)) { + dolog("live=%zu sw->hw->mix_buf.size=%zu\n", live, + sw->hw->mix_buf.size); return 0; } - dead = sw->hw->mix_buf->size - live; + dead = sw->hw->mix_buf.size - live; #ifdef DEBUG_OUT dolog("%s: get_free live %zu dead %zu frontend frames %zu\n", @@ -1056,12 +1056,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, n = samples; while (n) { - size_t till_end_of_hw = hw->mix_buf->size - rpos2; + 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.bytes_per_frame; size_t written; - sw->buf = hw->mix_buf->samples + rpos2; + sw->buf = hw->mix_buf.buffer + rpos2; written = audio_pcm_sw_write (sw, NULL, bytes); if (written - bytes) { dolog("Could not mix %zu bytes into a capture " @@ -1070,14 +1070,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, break; } n -= to_write; - rpos2 = (rpos2 + to_write) % hw->mix_buf->size; + rpos2 = (rpos2 + to_write) % hw->mix_buf.size; } } } - n = MIN(samples, hw->mix_buf->size - rpos); - mixeng_clear(hw->mix_buf->samples + rpos, n); - mixeng_clear(hw->mix_buf->samples, samples - n); + n = MIN(samples, hw->mix_buf.size - rpos); + mixeng_clear(hw->mix_buf.buffer + rpos, n); + mixeng_clear(hw->mix_buf.buffer, samples - n); } static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) @@ -1103,7 +1103,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) live -= proc; clipped += proc; - hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size; + hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size; if (proc == 0 || proc < decr) { break; @@ -1174,8 +1174,8 @@ static void audio_run_out (AudioState *s) live = 0; } - if (audio_bug(__func__, live > hw->mix_buf->size)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); + if (audio_bug(__func__, live > hw->mix_buf.size)) { + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); continue; } @@ -1203,13 +1203,13 @@ static void audio_run_out (AudioState *s) continue; } - prev_rpos = hw->mix_buf->pos; + prev_rpos = hw->mix_buf.pos; played = audio_pcm_hw_run_out(hw, live); replay_audio_out(&played); - if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) { - dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n", - hw->mix_buf->pos, hw->mix_buf->size, played); - hw->mix_buf->pos = 0; + if (audio_bug(__func__, hw->mix_buf.pos >= hw->mix_buf.size)) { + dolog("hw->mix_buf.pos=%zu hw->mix_buf.size=%zu played=%zu\n", + hw->mix_buf.pos, hw->mix_buf.size, played); + hw->mix_buf.pos = 0; } #ifdef DEBUG_OUT @@ -1290,10 +1290,10 @@ static void audio_run_in (AudioState *s) if (replay_mode != REPLAY_MODE_PLAY) { captured = audio_pcm_hw_run_in( - hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw)); + hw, hw->conv_buf.size - audio_pcm_hw_get_live_in(hw)); } - replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos, - hw->conv_buf->size); + replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos, + hw->conv_buf.size); min = audio_pcm_hw_find_min_in (hw); hw->total_samples_captured += captured - min; @@ -1326,14 +1326,14 @@ static void audio_run_capture (AudioState *s) SWVoiceOut *sw; captured = live = audio_pcm_hw_get_live_out (hw, NULL); - rpos = hw->mix_buf->pos; + rpos = hw->mix_buf.pos; while (live) { - size_t left = hw->mix_buf->size - rpos; + size_t left = hw->mix_buf.size - rpos; size_t to_capture = MIN(live, left); struct st_sample *src; struct capture_callback *cb; - src = hw->mix_buf->samples + rpos; + src = hw->mix_buf.buffer + rpos; hw->clip (cap->buf, src, to_capture); mixeng_clear (src, to_capture); @@ -1341,10 +1341,10 @@ static void audio_run_capture (AudioState *s) cb->ops.capture (cb->opaque, cap->buf, to_capture * hw->info.bytes_per_frame); } - rpos = (rpos + to_capture) % hw->mix_buf->size; + rpos = (rpos + to_capture) % hw->mix_buf.size; live -= to_capture; } - hw->mix_buf->pos = rpos; + hw->mix_buf.pos = rpos; for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { if (!sw->active && sw->empty) { @@ -1903,7 +1903,7 @@ CaptureVoiceOut *AUD_add_capture( audio_pcm_init_info (&hw->info, as); - cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame); + 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]; @@ -1955,7 +1955,7 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque) sw = sw1; } QLIST_REMOVE (cap, entries); - g_free (cap->hw.mix_buf); + g_free(cap->hw.mix_buf.buffer); g_free (cap->buf); g_free (cap); } diff --git a/audio/audio_int.h b/audio/audio_int.h index 5028f2354a..061845dcc2 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -58,7 +58,7 @@ typedef struct SWVoiceCap SWVoiceCap; typedef struct STSampleBuffer { size_t pos, size; - st_sample samples[]; + st_sample *buffer; } STSampleBuffer; typedef struct HWVoiceOut { @@ -71,7 +71,7 @@ typedef struct HWVoiceOut { f_sample *clip; uint64_t ts_helper; - STSampleBuffer *mix_buf; + STSampleBuffer mix_buf; void *buf_emul; size_t pos_emul, pending_emul, size_emul; @@ -93,7 +93,7 @@ typedef struct HWVoiceIn { size_t total_samples_captured; uint64_t ts_helper; - STSampleBuffer *conv_buf; + STSampleBuffer conv_buf; void *buf_emul; size_t pos_emul, pending_emul, size_emul; diff --git a/audio/audio_template.h b/audio/audio_template.h index 980e1f4bd0..dd87170cbd 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -71,8 +71,9 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw) { g_free(hw->buf_emul); - g_free (HWBUF); - HWBUF = NULL; + g_free(HWBUF.buffer); + HWBUF.buffer = NULL; + HWBUF.size = 0; } static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) @@ -83,10 +84,12 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) dolog("Attempted to allocate empty buffer\n"); } - HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples); - HWBUF->size = samples; + HWBUF.buffer = g_new0(st_sample, samples); + HWBUF.size = samples; + HWBUF.pos = 0; } else { - HWBUF = NULL; + HWBUF.buffer = NULL; + HWBUF.size = 0; } } @@ -111,9 +114,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) } #ifdef DAC - samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio; + samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; #else - samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; + samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; #endif if (audio_bug(__func__, samples < 0)) { dolog("Can not allocate buffer for `%s' (%d samples)\n", @@ -126,7 +129,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) size_t f_fe_min; /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ - f_fe_min = (hw->info.freq + HWBUF->size - 1) / HWBUF->size; + f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size; qemu_log_mask(LOG_UNIMP, AUDIO_CAP ": The guest selected a " NAME " sample rate" " of %d Hz for %s. Only sample rates >= %zu Hz are" From 2c3f9a0a92ac98c31d501c2a73119853cb6739f1 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:42 +0100 Subject: [PATCH 14/27] audio: change type and name of the resample buffer MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Change the type of the resample buffer from struct st_sample * to STSampleBuffer. Also change the name from buf to resample_buf for better readability. The new variables resample_buf.size and resample_buf.pos will be used after the next patches. There is no functional change. Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-2-vr_qemu@t-online.de> --- audio/audio.c | 15 ++++++++------- audio/audio_int.h | 4 ++-- audio/audio_template.h | 10 ++++++---- 3 files changed, 16 insertions(+), 13 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index a0b54e4a2e..a399147486 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -555,7 +555,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) { HWVoiceIn *hw = sw->hw; size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0; - struct st_sample *src, *dst = sw->buf; + struct st_sample *src, *dst = sw->resample_buf.buffer; live = hw->total_samples_captured - sw->total_hw_samples_acquired; if (!live) { @@ -595,10 +595,10 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) } if (!hw->pcm_ops->volume_in) { - mixeng_volume (sw->buf, ret, &sw->vol); + mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol); } - sw->clip (buf, sw->buf, ret); + sw->clip(buf, sw->resample_buf.buffer, ret); sw->total_hw_samples_acquired += total; return ret * sw->info.bytes_per_frame; } @@ -706,10 +706,10 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio; samples = MIN(samples, size / sw->info.bytes_per_frame); if (samples) { - sw->conv(sw->buf, buf, samples); + sw->conv(sw->resample_buf.buffer, buf, samples); if (!sw->hw->pcm_ops->volume_out) { - mixeng_volume(sw->buf, samples, &sw->vol); + mixeng_volume(sw->resample_buf.buffer, samples, &sw->vol); } } @@ -724,7 +724,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) osamp = blck; st_rate_flow_mix ( sw->rate, - sw->buf + pos, + sw->resample_buf.buffer + pos, sw->hw->mix_buf.buffer + wpos, &isamp, &osamp @@ -1061,7 +1061,8 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, size_t bytes = to_write * hw->info.bytes_per_frame; size_t written; - sw->buf = hw->mix_buf.buffer + rpos2; + sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; + sw->resample_buf.size = to_write; written = audio_pcm_sw_write (sw, NULL, bytes); if (written - bytes) { dolog("Could not mix %zu bytes into a capture " diff --git a/audio/audio_int.h b/audio/audio_int.h index 061845dcc2..8b163e1759 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -109,7 +109,7 @@ struct SWVoiceOut { struct audio_pcm_info info; t_sample *conv; int64_t ratio; - struct st_sample *buf; + STSampleBuffer resample_buf; void *rate; size_t total_hw_samples_mixed; int active; @@ -129,7 +129,7 @@ struct SWVoiceIn { int64_t ratio; void *rate; size_t total_hw_samples_acquired; - struct st_sample *buf; + STSampleBuffer resample_buf; f_sample *clip; HWVoiceIn *hw; char *name; diff --git a/audio/audio_template.h b/audio/audio_template.h index dd87170cbd..a0b653f52c 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -95,13 +95,13 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw) { - g_free (sw->buf); + g_free(sw->resample_buf.buffer); + sw->resample_buf.buffer = NULL; + sw->resample_buf.size = 0; if (sw->rate) { st_rate_stop (sw->rate); } - - sw->buf = NULL; sw->rate = NULL; } @@ -138,7 +138,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) return -1; } - sw->buf = g_new0(st_sample, samples); + sw->resample_buf.buffer = g_new0(st_sample, samples); + sw->resample_buf.size = samples; + sw->resample_buf.pos = 0; #ifdef DAC sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq); From 8933882da9c63a0b60706828e11f3c09e2a903c7 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:43 +0100 Subject: [PATCH 15/27] audio: make the resampling code greedy MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Read the maximum possible number of audio frames instead of the minimum necessary number of frames when the audio stream is downsampled and the output buffer is limited. This makes the function symmetrical to upsampling when the input buffer is limited. The maximum possible number of frames is written here. With this change it's easier to calculate the exact number of audio frames the resample function will read or write. These two functions will be introduced later. Acked-by: Mark Cave-Ayland Acked-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-3-vr_qemu@t-online.de> --- audio/rate_template.h | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/audio/rate_template.h b/audio/rate_template.h index b432719ebb..6648f0d2e5 100644 --- a/audio/rate_template.h +++ b/audio/rate_template.h @@ -40,8 +40,6 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, int64_t t; #endif - ilast = rate->ilast; - istart = ibuf; iend = ibuf + *isamp; @@ -59,15 +57,17 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, return; } - while (obuf < oend) { + /* without input samples, there's nothing to do */ + if (ibuf >= iend) { + *osamp = 0; + return; + } - /* Safety catch to make sure we have input samples. */ - if (ibuf >= iend) { - break; - } + ilast = rate->ilast; + + while (true) { /* read as many input samples so that ipos > opos */ - while (rate->ipos <= (rate->opos >> 32)) { ilast = *ibuf++; rate->ipos++; @@ -78,6 +78,11 @@ void NAME (void *opaque, struct st_sample *ibuf, struct st_sample *obuf, } } + /* make sure that the next output sample can be written */ + if (obuf >= oend) { + break; + } + icur = *ibuf; /* wrap ipos and opos around long before they overflow */ From 8a81abeeb2e17630c366fccbd99ce20eac328819 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:44 +0100 Subject: [PATCH 16/27] audio: replace the resampling loop in audio_pcm_sw_write() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Replace the resampling loop in audio_pcm_sw_write() with the new function audio_pcm_sw_resample_out(). Unlike the old resample loop the new function will try to consume input frames even if the output buffer is full. This is necessary when downsampling to avoid reading less audio frames than calculated in advance. The loop was unrolled to avoid complicated loop control conditions in this case. Acked-by: Mark Cave-Ayland Acked-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-4-vr_qemu@t-online.de> --- audio/audio.c | 63 +++++++++++++++++++++++++++++---------------------- 1 file changed, 36 insertions(+), 27 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index a399147486..4412b5fad8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -673,11 +673,44 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) /* * Soft voice (playback) */ +static void audio_pcm_sw_resample_out(SWVoiceOut *sw, + size_t frames_in_max, size_t frames_out_max, + size_t *total_in, size_t *total_out) +{ + HWVoiceOut *hw = sw->hw; + struct st_sample *src, *dst; + size_t live, wpos, frames_in, frames_out; + + live = sw->total_hw_samples_mixed; + wpos = (hw->mix_buf.pos + live) % hw->mix_buf.size; + + /* write to mix_buf from wpos to end of buffer */ + src = sw->resample_buf.buffer; + frames_in = frames_in_max; + dst = hw->mix_buf.buffer + wpos; + frames_out = MIN(frames_out_max, hw->mix_buf.size - wpos); + st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out); + wpos += frames_out; + *total_in = frames_in; + *total_out = frames_out; + + /* write to mix_buf from start of buffer if there are input frames left */ + if (frames_in_max - frames_in > 0 && wpos == hw->mix_buf.size) { + src += frames_in; + frames_in = frames_in_max - frames_in; + dst = hw->mix_buf.buffer; + frames_out = frames_out_max - frames_out; + st_rate_flow_mix(sw->rate, src, dst, &frames_in, &frames_out); + *total_in += frames_in; + *total_out += frames_out; + } +} + static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) { - size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck; + size_t hwsamples, samples, live, dead; size_t hw_free; - size_t ret = 0, pos = 0, total = 0; + size_t ret, total; if (!sw) { return size; @@ -698,8 +731,6 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) return 0; } - wpos = (sw->hw->mix_buf.pos + live) % hwsamples; - dead = hwsamples - live; hw_free = audio_pcm_hw_get_free(sw->hw); hw_free = hw_free > live ? hw_free - live : 0; @@ -713,29 +744,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) } } - while (samples) { - dead = hwsamples - live; - left = hwsamples - wpos; - blck = MIN (dead, left); - if (!blck) { - break; - } - isamp = samples; - osamp = blck; - st_rate_flow_mix ( - sw->rate, - sw->resample_buf.buffer + pos, - sw->hw->mix_buf.buffer + wpos, - &isamp, - &osamp - ); - ret += isamp; - samples -= isamp; - pos += isamp; - live += osamp; - wpos = (wpos + osamp) % hwsamples; - total += osamp; - } + audio_pcm_sw_resample_out(sw, samples, MIN(dead, hw_free), &ret, &total); sw->total_hw_samples_mixed += total; sw->empty = sw->total_hw_samples_mixed == 0; From b8fc5638788d2f1200e0db2c1575e37be5e474f9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:45 +0100 Subject: [PATCH 17/27] audio: remove sw == NULL check MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit All call sites of audio_pcm_sw_write() guarantee that sw is not NULL. Remove the unnecessary NULL check. Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-5-vr_qemu@t-online.de> --- audio/audio.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 4412b5fad8..8f1c0e77b0 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -712,10 +712,6 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) size_t hw_free; size_t ret, total; - if (!sw) { - return size; - } - hwsamples = sw->hw->mix_buf.size; live = sw->total_hw_samples_mixed; From d5647bd95831d9c98284033bd992f0a8b6deb103 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:46 +0100 Subject: [PATCH 18/27] audio: rename variables in audio_pcm_sw_write() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The audio_pcm_sw_write() function uses a lot of very unspecific variable names. Rename them for better readability. ret => total_in total => total_out size => buf_len hwsamples => hw->mix_buf.size samples => frames_in_max Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-6-vr_qemu@t-online.de> --- audio/audio.c | 45 ++++++++++++++++++++++----------------------- 1 file changed, 22 insertions(+), 23 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 8f1c0e77b0..cd10f1ec10 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -706,56 +706,55 @@ static void audio_pcm_sw_resample_out(SWVoiceOut *sw, } } -static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) +static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) { - size_t hwsamples, samples, live, dead; - size_t hw_free; - size_t ret, total; - - hwsamples = sw->hw->mix_buf.size; + HWVoiceOut *hw = sw->hw; + size_t live, dead, hw_free; + size_t frames_in_max, total_in, total_out; live = sw->total_hw_samples_mixed; - if (audio_bug(__func__, live > hwsamples)) { - dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples); + if (audio_bug(__func__, live > hw->mix_buf.size)) { + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); return 0; } - if (live == hwsamples) { + if (live == hw->mix_buf.size) { #ifdef DEBUG_OUT dolog ("%s is full %zu\n", sw->name, live); #endif return 0; } - dead = hwsamples - live; - hw_free = audio_pcm_hw_get_free(sw->hw); + dead = hw->mix_buf.size - live; + hw_free = audio_pcm_hw_get_free(hw); hw_free = hw_free > live ? hw_free - live : 0; - samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio; - samples = MIN(samples, size / sw->info.bytes_per_frame); - if (samples) { - sw->conv(sw->resample_buf.buffer, buf, samples); + frames_in_max = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio; + frames_in_max = MIN(frames_in_max, buf_len / sw->info.bytes_per_frame); + if (frames_in_max) { + sw->conv(sw->resample_buf.buffer, buf, frames_in_max); if (!sw->hw->pcm_ops->volume_out) { - mixeng_volume(sw->resample_buf.buffer, samples, &sw->vol); + mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol); } } - audio_pcm_sw_resample_out(sw, samples, MIN(dead, hw_free), &ret, &total); + audio_pcm_sw_resample_out(sw, frames_in_max, MIN(dead, hw_free), + &total_in, &total_out); - sw->total_hw_samples_mixed += total; + sw->total_hw_samples_mixed += total_out; sw->empty = sw->total_hw_samples_mixed == 0; #ifdef DEBUG_OUT dolog ( - "%s: write size %zu ret %zu total sw %zu\n", - SW_NAME (sw), - size / sw->info.bytes_per_frame, - ret, + "%s: write size %zu written %zu total mixed %zu\n", + SW_NAME(sw), + buf_len / sw->info.bytes_per_frame, + total_in, sw->total_hw_samples_mixed ); #endif - return ret * sw->info.bytes_per_frame; + return total_in * sw->info.bytes_per_frame; } #ifdef DEBUG_AUDIO From 671cca352075c9651e5c2e64bb0d2c70465082a3 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:47 +0100 Subject: [PATCH 19/27] audio: don't misuse audio_pcm_sw_write() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The audio_pcm_sw_write() function is intended to convert a PCM audio stream to the internal representation, adjust the volume, and then mix it with the other audio streams with a possibly changed sample rate in mix_buf. In order for the audio_capture_mix_and_clear() function to use audio_pcm_sw_write(), it must bypass the first two tasks of audio_pcm_sw_write(). Since patch "audio: split out the resampling loop in audio_pcm_sw_write()" this is no longer necessary, because now the audio_pcm_sw_resample_out() function can be used instead of audio_pcm_sw_write(). Acked-by: Mark Cave-Ayland Acked-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-7-vr_qemu@t-online.de> --- audio/audio.c | 29 ++++++++++++++++++----------- 1 file changed, 18 insertions(+), 11 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index cd10f1ec10..44eb7b63b4 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -1056,26 +1056,33 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, for (sc = hw->cap_head.lh_first; sc; sc = sc->entries.le_next) { SWVoiceOut *sw = &sc->sw; - int rpos2 = rpos; + size_t rpos2 = rpos; n = samples; 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.bytes_per_frame; - size_t written; + size_t to_read = MIN(till_end_of_hw, n); + size_t live, frames_in, frames_out; sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; - sw->resample_buf.size = to_write; - written = audio_pcm_sw_write (sw, NULL, bytes); - if (written - bytes) { - dolog("Could not mix %zu bytes into a capture " + sw->resample_buf.size = to_read; + live = sw->total_hw_samples_mixed; + + audio_pcm_sw_resample_out(sw, + to_read, sw->hw->mix_buf.size - live, + &frames_in, &frames_out); + + sw->total_hw_samples_mixed += frames_out; + sw->empty = sw->total_hw_samples_mixed == 0; + + if (to_read - frames_in) { + dolog("Could not mix %zu frames into a capture " "buffer, mixed %zu\n", - bytes, written); + to_read, frames_in); break; } - n -= to_write; - rpos2 = (rpos2 + to_write) % hw->mix_buf.size; + n -= to_read; + rpos2 = (rpos2 + to_read) % hw->mix_buf.size; } } } From 1fe3cae39f059c9fc2010e3c51c0bbd696cbf880 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:48 +0100 Subject: [PATCH 20/27] audio: remove unused noop_conv() function MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The function audio_capture_mix_and_clear() no longer uses audio_pcm_sw_write() to resample audio frames from one internal buffer to another. For this reason, the noop_conv() function is now unused. Remove it. Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-8-vr_qemu@t-online.de> --- audio/audio.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 44eb7b63b4..556696b095 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -381,13 +381,6 @@ void audio_pcm_info_clear_buf (struct audio_pcm_info *info, void *buf, int len) /* * Capture */ -static void noop_conv (struct st_sample *dst, const void *src, int samples) -{ - (void) src; - (void) dst; - (void) samples; -} - static CaptureVoiceOut *audio_pcm_capture_find_specific(AudioState *s, struct audsettings *as) { @@ -485,7 +478,6 @@ static int audio_attach_capture (HWVoiceOut *hw) sw->info = hw->info; sw->empty = 1; sw->active = hw->enabled; - sw->conv = noop_conv; sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq; sw->vol = nominal_volume; sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq); From 1a01df3db89010d40eb43889c3272d864b3b9430 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:49 +0100 Subject: [PATCH 21/27] audio: make playback packet length calculation exact MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Introduce the new function st_rate_frames_in() to calculate the exact number of audio input frames needed to get a given number of audio output frames. The exact number of frames depends only on the difference of opos - ipos and the number of output frames. When downsampling, this function returns the maximum number of input frames needed. This new function replaces the audio_frontend_frames_out() function, which calculated the average number of input frames rounded down to the nearest integer. Because audio_frontend_frames_out() also limited the number of input frames to the size of the resample buffer, st_rate_frames_in() is not a direct replacement and two additional MIN() functions are needed. One to prevent resample buffer overflows and one to limit the available bytes for the audio frontends. After this patch the audio packet length calculation for playback is exact. When upsampling, it's still possible that the audio frontends can't write the last audio frame. This will be fixed later. Acked-by: Mark Cave-Ayland Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-9-vr_qemu@t-online.de> --- audio/audio.c | 43 ++++++++++++++++++------------------------- audio/mixeng.c | 39 +++++++++++++++++++++++++++++++++++++++ audio/mixeng.h | 1 + 3 files changed, 58 insertions(+), 25 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 556696b095..e18b5e98c5 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -701,8 +701,8 @@ static void audio_pcm_sw_resample_out(SWVoiceOut *sw, static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) { HWVoiceOut *hw = sw->hw; - size_t live, dead, hw_free; - size_t frames_in_max, total_in, total_out; + size_t live, dead, hw_free, sw_max, fe_max; + size_t frames_in_max, frames_out_max, total_in, total_out; live = sw->total_hw_samples_mixed; if (audio_bug(__func__, live > hw->mix_buf.size)) { @@ -720,17 +720,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) dead = hw->mix_buf.size - live; hw_free = audio_pcm_hw_get_free(hw); hw_free = hw_free > live ? hw_free - live : 0; - frames_in_max = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio; - frames_in_max = MIN(frames_in_max, buf_len / sw->info.bytes_per_frame); - if (frames_in_max) { - sw->conv(sw->resample_buf.buffer, buf, frames_in_max); + frames_out_max = MIN(dead, hw_free); + sw_max = st_rate_frames_in(sw->rate, frames_out_max); + fe_max = MIN(buf_len / sw->info.bytes_per_frame, sw->resample_buf.size); + frames_in_max = MIN(sw_max, fe_max); - if (!sw->hw->pcm_ops->volume_out) { - mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol); - } + if (!frames_in_max) { + return 0; } - audio_pcm_sw_resample_out(sw, frames_in_max, MIN(dead, hw_free), + sw->conv(sw->resample_buf.buffer, buf, frames_in_max); + if (!sw->hw->pcm_ops->volume_out) { + mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol); + } + + audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max, &total_in, &total_out); sw->total_hw_samples_mixed += total_out; @@ -1000,18 +1004,6 @@ static size_t audio_get_avail (SWVoiceIn *sw) return live; } -/** - * audio_frontend_frames_out() - returns the number of frames needed to - * get frames_out frames after resampling - * - * @sw: audio playback frontend - * @frames_out: number of frames - */ -static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out) -{ - return ((int64_t)frames_out << 32) / sw->ratio; -} - static size_t audio_get_free(SWVoiceOut *sw) { size_t live, dead; @@ -1031,8 +1023,8 @@ static size_t audio_get_free(SWVoiceOut *sw) dead = sw->hw->mix_buf.size - live; #ifdef DEBUG_OUT - dolog("%s: get_free live %zu dead %zu frontend frames %zu\n", - SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead)); + dolog("%s: get_free live %zu dead %zu frontend frames %u\n", + SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead)); #endif return dead; @@ -1161,12 +1153,13 @@ static void audio_run_out (AudioState *s) size_t free; if (hw_free > sw->total_hw_samples_mixed) { - free = audio_frontend_frames_out(sw, + free = st_rate_frames_in(sw->rate, MIN(sw_free, hw_free - sw->total_hw_samples_mixed)); } else { free = 0; } if (free > 0) { + free = MIN(free, sw->resample_buf.size); sw->callback.fn(sw->callback.opaque, free * sw->info.bytes_per_frame); } diff --git a/audio/mixeng.c b/audio/mixeng.c index fe454e0725..a24c8c45a7 100644 --- a/audio/mixeng.c +++ b/audio/mixeng.c @@ -440,6 +440,45 @@ void st_rate_stop (void *opaque) g_free (opaque); } +/** + * st_rate_frames_in() - returns the number of frames needed to + * get frames_out frames after resampling + * + * @opaque: pointer to struct rate + * @frames_out: number of frames + * + * When downsampling, there may be more than one correct result. In this + * case, the function returns the maximum number of input frames needed. + */ +uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out) +{ + struct rate *rate = opaque; + uint64_t opos_start, opos_end; + uint32_t ipos_start, ipos_end; + + if (rate->opos_inc == 1ULL << 32) { + return frames_out; + } + + if (frames_out) { + opos_start = rate->opos; + ipos_start = rate->ipos; + } else { + uint64_t offset; + + /* add offset = ceil(opos_inc) to opos and ipos to avoid an underflow */ + offset = (rate->opos_inc + (1ULL << 32) - 1) & ~((1ULL << 32) - 1); + opos_start = rate->opos + offset; + ipos_start = rate->ipos + (offset >> 32); + } + /* last frame written was at opos_start - rate->opos_inc */ + opos_end = opos_start - rate->opos_inc + rate->opos_inc * frames_out; + ipos_end = (opos_end >> 32) + 1; + + /* last frame read was at ipos_start - 1 */ + return ipos_end + 1 > ipos_start ? ipos_end + 1 - ipos_start : 0; +} + void mixeng_clear (struct st_sample *buf, int len) { memset (buf, 0, len * sizeof (struct st_sample)); diff --git a/audio/mixeng.h b/audio/mixeng.h index 2dcd6df245..64c1e231cc 100644 --- a/audio/mixeng.h +++ b/audio/mixeng.h @@ -52,6 +52,7 @@ void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf, void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf, size_t *isamp, size_t *osamp); void st_rate_stop (void *opaque); +uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out); void mixeng_clear (struct st_sample *buf, int len); void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol); From 1c49c5f19e53cf29ae3356c0f0b3b124d44a3664 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:50 +0100 Subject: [PATCH 22/27] audio: replace the resampling loop in audio_pcm_sw_read() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Replace the resampling loop in audio_pcm_sw_read() with the new function audio_pcm_sw_resample_in(). Unlike the old resample loop the new function will try to consume input frames even if the output buffer is full. This is necessary when downsampling to avoid reading less audio frames than calculated in advance. The loop was unrolled to avoid complicated loop control conditions in this case. Acked-by: Mark Cave-Ayland Acked-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-10-vr_qemu@t-online.de> --- audio/audio.c | 59 ++++++++++++++++++++++++++++++--------------------- 1 file changed, 35 insertions(+), 24 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index e18b5e98c5..9e9c03a42e 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -543,11 +543,43 @@ static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples) /* * Soft voice (capture) */ +static void audio_pcm_sw_resample_in(SWVoiceIn *sw, + size_t frames_in_max, size_t frames_out_max, + size_t *total_in, size_t *total_out) +{ + HWVoiceIn *hw = sw->hw; + struct st_sample *src, *dst; + size_t live, rpos, frames_in, frames_out; + + live = hw->total_samples_captured - sw->total_hw_samples_acquired; + rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size); + + /* resample conv_buf from rpos to end of buffer */ + src = hw->conv_buf.buffer + rpos; + frames_in = MIN(frames_in_max, hw->conv_buf.size - rpos); + dst = sw->resample_buf.buffer; + frames_out = frames_out_max; + st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out); + rpos += frames_in; + *total_in = frames_in; + *total_out = frames_out; + + /* resample conv_buf from start of buffer if there are input frames left */ + if (frames_in_max - frames_in && rpos == hw->conv_buf.size) { + src = hw->conv_buf.buffer; + frames_in = frames_in_max - frames_in; + dst += frames_out; + frames_out = frames_out_max - frames_out; + st_rate_flow(sw->rate, src, dst, &frames_in, &frames_out); + *total_in += frames_in; + *total_out += frames_out; + } +} + static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) { HWVoiceIn *hw = sw->hw; - size_t samples, live, ret = 0, swlim, isamp, osamp, rpos, total = 0; - struct st_sample *src, *dst = sw->resample_buf.buffer; + size_t samples, live, ret, swlim, total; live = hw->total_samples_captured - sw->total_hw_samples_acquired; if (!live) { @@ -558,33 +590,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) return 0; } - rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size); - samples = size / sw->info.bytes_per_frame; swlim = (live * sw->ratio) >> 32; swlim = MIN (swlim, samples); - while (swlim) { - src = hw->conv_buf.buffer + rpos; - if (hw->conv_buf.pos > rpos) { - isamp = hw->conv_buf.pos - rpos; - } else { - isamp = hw->conv_buf.size - rpos; - } - - if (!isamp) { - break; - } - osamp = swlim; - - st_rate_flow (sw->rate, src, dst, &isamp, &osamp); - swlim -= osamp; - rpos = (rpos + isamp) % hw->conv_buf.size; - dst += osamp; - ret += osamp; - total += isamp; - } + audio_pcm_sw_resample_in(sw, live, swlim, &total, &ret); if (!hw->pcm_ops->volume_in) { mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol); From fbde1edf06dad792ef3e9f51e3f52a49669bdd40 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:51 +0100 Subject: [PATCH 23/27] audio: rename variables in audio_pcm_sw_read() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The audio_pcm_sw_read() function uses a few very unspecific variable names. Rename them for better readability. ret => total_out total => total_in size => buf_len samples => frames_out_max Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-11-vr_qemu@t-online.de> --- audio/audio.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 9e9c03a42e..22c36d6660 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -576,10 +576,10 @@ static void audio_pcm_sw_resample_in(SWVoiceIn *sw, } } -static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) +static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len) { HWVoiceIn *hw = sw->hw; - size_t samples, live, ret, swlim, total; + size_t live, frames_out_max, swlim, total_in, total_out; live = hw->total_samples_captured - sw->total_hw_samples_acquired; if (!live) { @@ -590,20 +590,20 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) return 0; } - samples = size / sw->info.bytes_per_frame; + frames_out_max = buf_len / sw->info.bytes_per_frame; swlim = (live * sw->ratio) >> 32; - swlim = MIN (swlim, samples); + swlim = MIN(swlim, frames_out_max); - audio_pcm_sw_resample_in(sw, live, swlim, &total, &ret); + audio_pcm_sw_resample_in(sw, live, swlim, &total_in, &total_out); if (!hw->pcm_ops->volume_in) { - mixeng_volume(sw->resample_buf.buffer, ret, &sw->vol); + mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol); } + sw->clip(buf, sw->resample_buf.buffer, total_out); - sw->clip(buf, sw->resample_buf.buffer, ret); - sw->total_hw_samples_acquired += total; - return ret * sw->info.bytes_per_frame; + sw->total_hw_samples_acquired += total_in; + return total_out * sw->info.bytes_per_frame; } /* From a9ea567873ba8d532520f194413ff28f37065c00 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:52 +0100 Subject: [PATCH 24/27] audio: make recording packet length calculation exact MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Introduce the new function st_rate_frames_out() to calculate the exact number of audio output frames the resampling code can generate from a given number of audio input frames. When upsampling, this function returns the maximum number of output frames. This new function replaces the audio_frontend_frames_in() function, which calculated the average number of output frames rounded down to the nearest integer. The audio_frontend_frames_in() function was additionally used to limit the number of output frames to the resample buffer size. In audio_pcm_sw_read() the variable resample_buf.size replaces the open coded audio_frontend_frames_in() function. In audio_run_in() an additional MIN() function is necessary. After this patch the audio packet length calculation for audio recording is exact. Acked-by: Mark Cave-Ayland Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-12-vr_qemu@t-online.de> --- audio/audio.c | 29 ++++++++--------------------- audio/mixeng.c | 41 +++++++++++++++++++++++++++++++++++++++++ audio/mixeng.h | 1 + 3 files changed, 50 insertions(+), 21 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 22c36d6660..dad17e59b8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -579,7 +579,7 @@ static void audio_pcm_sw_resample_in(SWVoiceIn *sw, static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len) { HWVoiceIn *hw = sw->hw; - size_t live, frames_out_max, swlim, total_in, total_out; + size_t live, frames_out_max, total_in, total_out; live = hw->total_samples_captured - sw->total_hw_samples_acquired; if (!live) { @@ -590,12 +590,10 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len) return 0; } - frames_out_max = buf_len / sw->info.bytes_per_frame; + frames_out_max = MIN(buf_len / sw->info.bytes_per_frame, + sw->resample_buf.size); - swlim = (live * sw->ratio) >> 32; - swlim = MIN(swlim, frames_out_max); - - audio_pcm_sw_resample_in(sw, live, swlim, &total_in, &total_out); + audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out); if (!hw->pcm_ops->volume_in) { mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol); @@ -979,18 +977,6 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) } } -/** - * audio_frontend_frames_in() - returns the number of frames the resampling - * code generates from frames_in frames - * - * @sw: audio recording frontend - * @frames_in: number of frames - */ -static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in) -{ - return (int64_t)frames_in * sw->ratio >> 32; -} - static size_t audio_get_avail (SWVoiceIn *sw) { size_t live; @@ -1007,9 +993,9 @@ static size_t audio_get_avail (SWVoiceIn *sw) } ldebug ( - "%s: get_avail live %zu frontend frames %zu\n", + "%s: get_avail live %zu frontend frames %u\n", SW_NAME (sw), - live, audio_frontend_frames_in(sw, live) + live, st_rate_frames_out(sw->rate, live) ); return live; @@ -1314,8 +1300,9 @@ static void audio_run_in (AudioState *s) size_t sw_avail = audio_get_avail(sw); size_t avail; - avail = audio_frontend_frames_in(sw, sw_avail); + avail = st_rate_frames_out(sw->rate, sw_avail); if (avail > 0) { + avail = MIN(avail, sw->resample_buf.size); sw->callback.fn(sw->callback.opaque, avail * sw->info.bytes_per_frame); } diff --git a/audio/mixeng.c b/audio/mixeng.c index a24c8c45a7..69f6549224 100644 --- a/audio/mixeng.c +++ b/audio/mixeng.c @@ -440,6 +440,47 @@ void st_rate_stop (void *opaque) g_free (opaque); } +/** + * st_rate_frames_out() - returns the number of frames the resampling code + * generates from frames_in frames + * + * @opaque: pointer to struct rate + * @frames_in: number of frames + * + * When upsampling, there may be more than one correct result. In this case, + * the function returns the maximum number of output frames the resampling + * code can generate. + */ +uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in) +{ + struct rate *rate = opaque; + uint64_t opos_end, opos_delta; + uint32_t ipos_end; + uint32_t frames_out; + + if (rate->opos_inc == 1ULL << 32) { + return frames_in; + } + + /* no output frame without at least one input frame */ + if (!frames_in) { + return 0; + } + + /* last frame read was at rate->ipos - 1 */ + ipos_end = rate->ipos - 1 + frames_in; + opos_end = (uint64_t)ipos_end << 32; + + /* last frame written was at rate->opos - rate->opos_inc */ + if (opos_end + rate->opos_inc <= rate->opos) { + return 0; + } + opos_delta = opos_end - rate->opos + rate->opos_inc; + frames_out = opos_delta / rate->opos_inc; + + return opos_delta % rate->opos_inc ? frames_out : frames_out - 1; +} + /** * st_rate_frames_in() - returns the number of frames needed to * get frames_out frames after resampling diff --git a/audio/mixeng.h b/audio/mixeng.h index 64c1e231cc..f9de7cffeb 100644 --- a/audio/mixeng.h +++ b/audio/mixeng.h @@ -52,6 +52,7 @@ void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf, void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf, size_t *isamp, size_t *osamp); void st_rate_stop (void *opaque); +uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in); uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out); void mixeng_clear (struct st_sample *buf, int len); void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol); From e1e6a6fcc9921a76a32109af515146f85ccc5f84 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:53 +0100 Subject: [PATCH 25/27] audio: handle leftover audio frame from upsampling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Upsampling may leave one remaining audio frame in the input buffer. The emulated audio playback devices are currently resposible to write this audio frame again in the next write cycle. Push that task down to audio_pcm_sw_write. This is another step towards an audio callback interface that guarantees that when audio frontends are told they can write n audio frames, they can actually do so. Acked-by: Mark Cave-Ayland Acked-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-13-vr_qemu@t-online.de> --- audio/audio.c | 34 ++++++++++++++++++++++++++++------ audio/audio_template.h | 6 ++++++ 2 files changed, 34 insertions(+), 6 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index dad17e59b8..4836ab8ca8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -731,16 +731,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) hw_free = hw_free > live ? hw_free - live : 0; frames_out_max = MIN(dead, hw_free); sw_max = st_rate_frames_in(sw->rate, frames_out_max); - fe_max = MIN(buf_len / sw->info.bytes_per_frame, sw->resample_buf.size); + fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos, + sw->resample_buf.size); frames_in_max = MIN(sw_max, fe_max); if (!frames_in_max) { return 0; } - sw->conv(sw->resample_buf.buffer, buf, frames_in_max); - if (!sw->hw->pcm_ops->volume_out) { - mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol); + if (frames_in_max > sw->resample_buf.pos) { + sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos, + buf, frames_in_max - sw->resample_buf.pos); + if (!sw->hw->pcm_ops->volume_out) { + mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos, + frames_in_max - sw->resample_buf.pos, &sw->vol); + } } audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max, @@ -749,6 +754,22 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) sw->total_hw_samples_mixed += total_out; sw->empty = sw->total_hw_samples_mixed == 0; + /* + * Upsampling may leave one audio frame in the resample buffer. Decrement + * total_in by one if there was a leftover frame from the previous resample + * pass in the resample buffer. Increment total_in by one if the current + * resample pass left one frame in the resample buffer. + */ + if (frames_in_max - total_in == 1) { + /* copy one leftover audio frame to the beginning of the buffer */ + *sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in); + total_in += 1 - sw->resample_buf.pos; + sw->resample_buf.pos = 1; + } else if (total_in >= sw->resample_buf.pos) { + total_in -= sw->resample_buf.pos; + sw->resample_buf.pos = 0; + } + #ifdef DEBUG_OUT dolog ( "%s: write size %zu written %zu total mixed %zu\n", @@ -1155,8 +1176,9 @@ static void audio_run_out (AudioState *s) } else { free = 0; } - if (free > 0) { - free = MIN(free, sw->resample_buf.size); + if (free > sw->resample_buf.pos) { + free = MIN(free, sw->resample_buf.size) + - sw->resample_buf.pos; sw->callback.fn(sw->callback.opaque, free * sw->info.bytes_per_frame); } diff --git a/audio/audio_template.h b/audio/audio_template.h index a0b653f52c..0d8aab6fad 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -138,6 +138,12 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) return -1; } + /* + * Allocate one additional audio frame that is needed for upsampling + * if the resample buffer size is small. For large buffer sizes take + * care of overflows. + */ + samples = samples < INT_MAX ? samples + 1 : INT_MAX; sw->resample_buf.buffer = g_new0(st_sample, samples); sw->resample_buf.size = samples; sw->resample_buf.pos = 0; From 148392abef4ed3591675fc8b07cc3063a3369a7b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:54 +0100 Subject: [PATCH 26/27] audio/audio_template: substitute sw->hw with hw MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Substitute sw->hw with hw in the audio_pcm_sw_alloc_resources_* functions. Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-14-vr_qemu@t-online.de> --- audio/audio_template.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/audio/audio_template.h b/audio/audio_template.h index 0d8aab6fad..7e116426c7 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -107,6 +107,7 @@ static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw) static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) { + HW *hw = sw->hw; int samples; if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) { @@ -125,7 +126,6 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) } if (samples == 0) { - HW *hw = sw->hw; size_t f_fe_min; /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ @@ -149,9 +149,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) sw->resample_buf.pos = 0; #ifdef DAC - sw->rate = st_rate_start (sw->info.freq, sw->hw->info.freq); + sw->rate = st_rate_start(sw->info.freq, hw->info.freq); #else - sw->rate = st_rate_start (sw->hw->info.freq, sw->info.freq); + sw->rate = st_rate_start(hw->info.freq, sw->info.freq); #endif return 0; From 2f886a34bb7e6f6fcf39d64829f4499476f26dba Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Volker=20R=C3=BCmelin?= Date: Fri, 24 Feb 2023 20:05:55 +0100 Subject: [PATCH 27/27] audio: remove sw->ratio MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Simplify the resample buffer size calculation. For audio playback we have sw->ratio = ((int64_t)sw->hw->info.freq << 32) / sw->info.freq; samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; This can be simplified to samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); For audio recording we have sw->ratio = ((int64_t)sw->info.freq << 32) / sw->hw->info.freq; samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; This can be simplified to samples = muldiv64(sw->HWBUF.size, sw->info.freq, sw->hw->info.freq); With hw = sw->hw this becomes in both cases samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); Now that sw->ratio is no longer needed, remove sw->ratio. Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin Message-Id: <20230224190555.7409-15-vr_qemu@t-online.de> --- audio/audio.c | 1 - audio/audio_int.h | 2 -- audio/audio_template.h | 30 +++++++++--------------------- 3 files changed, 9 insertions(+), 24 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 4836ab8ca8..70b096713c 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -478,7 +478,6 @@ static int audio_attach_capture (HWVoiceOut *hw) sw->info = hw->info; sw->empty = 1; sw->active = hw->enabled; - sw->ratio = ((int64_t) hw_cap->info.freq << 32) / sw->info.freq; sw->vol = nominal_volume; sw->rate = st_rate_start (sw->info.freq, hw_cap->info.freq); QLIST_INSERT_HEAD (&hw_cap->sw_head, sw, entries); diff --git a/audio/audio_int.h b/audio/audio_int.h index 8b163e1759..d51d63f08d 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -108,7 +108,6 @@ struct SWVoiceOut { AudioState *s; struct audio_pcm_info info; t_sample *conv; - int64_t ratio; STSampleBuffer resample_buf; void *rate; size_t total_hw_samples_mixed; @@ -126,7 +125,6 @@ struct SWVoiceIn { AudioState *s; int active; struct audio_pcm_info info; - int64_t ratio; void *rate; size_t total_hw_samples_acquired; STSampleBuffer resample_buf; diff --git a/audio/audio_template.h b/audio/audio_template.h index 7e116426c7..e42326c20d 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -108,32 +108,23 @@ static void glue (audio_pcm_sw_free_resources_, TYPE) (SW *sw) static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) { HW *hw = sw->hw; - int samples; + uint64_t samples; if (!glue(audio_get_pdo_, TYPE)(sw->s->dev)->mixing_engine) { return 0; } -#ifdef DAC - samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; -#else - samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; -#endif - if (audio_bug(__func__, samples < 0)) { - dolog("Can not allocate buffer for `%s' (%d samples)\n", - SW_NAME(sw), samples); - return -1; - } - + samples = muldiv64(HWBUF.size, sw->info.freq, hw->info.freq); if (samples == 0) { - size_t f_fe_min; + uint64_t f_fe_min; + uint64_t f_be = (uint32_t)hw->info.freq; /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ - f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size; + f_fe_min = (f_be + HWBUF.size - 1) / HWBUF.size; qemu_log_mask(LOG_UNIMP, AUDIO_CAP ": The guest selected a " NAME " sample rate" - " of %d Hz for %s. Only sample rates >= %zu Hz are" - " supported.\n", + " of %d Hz for %s. Only sample rates >= %" PRIu64 " Hz" + " are supported.\n", sw->info.freq, sw->name, f_fe_min); return -1; } @@ -141,9 +132,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) /* * Allocate one additional audio frame that is needed for upsampling * if the resample buffer size is small. For large buffer sizes take - * care of overflows. + * care of overflows and truncation. */ - samples = samples < INT_MAX ? samples + 1 : INT_MAX; + samples = samples < SIZE_MAX ? samples + 1 : SIZE_MAX; sw->resample_buf.buffer = g_new0(st_sample, samples); sw->resample_buf.size = samples; sw->resample_buf.pos = 0; @@ -170,11 +161,8 @@ static int glue (audio_pcm_sw_init_, TYPE) ( sw->hw = hw; sw->active = 0; #ifdef DAC - sw->ratio = ((int64_t) sw->hw->info.freq << 32) / sw->info.freq; sw->total_hw_samples_mixed = 0; sw->empty = 1; -#else - sw->ratio = ((int64_t) sw->info.freq << 32) / sw->hw->info.freq; #endif if (sw->info.is_float) {