File 0001-spa-Fix-audioconvert-overflow-when-scaling... of Package pipewire.30041

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File 0001-spa-Fix-audioconvert-overflow-when-scaling.patch of Package pipewire.30041

From 1d9640af5a7906620f214aa0a39c63128c8506a6 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wtaymans@redhat.com>
Date: Mon, 16 Jan 2023 18:28:31 +0100
Subject: [PATCH] spa: Fix audioconvert overflow when scaling

Add SPA_SCALE32_UP that scales a uint32 without overflow.
Use this for scaling the threshold in ALSA.
Fix the scaling in audioconvert of the buffer size, the scaling was
wrong and it was also causing an overflow resulting in choppy sound in
some cases.

See #2680
---
 spa/include/spa/utils/defs.h            | 8 ++++++++
 spa/plugins/alsa/alsa-pcm.c             | 4 ++--
 spa/plugins/audioconvert/audioconvert.c | 2 +-
 3 files changed, 11 insertions(+), 3 deletions(-)

diff --git a/spa/include/spa/utils/defs.h b/spa/include/spa/utils/defs.h
index c602c9348..3b4862663 100644
--- a/spa/include/spa/utils/defs.h
+++ b/spa/include/spa/utils/defs.h
@@ -274,6 +274,14 @@ struct spa_fraction {
# #define SPA_ROUND_DOWN_N(num,align)	((num) & ~SPA_ROUND_MASK(num, align))
# #define SPA_ROUND_UP_N(num,align)	((((num)-1) | SPA_ROUND_MASK(num, align))+1)
# 
 #define SPA_ROUND_DOWN_N(num,align)	((num) & ~((align) - 1))
 #define SPA_ROUND_UP_N(num,align)	SPA_ROUND_DOWN_N((num) + ((align) - 1),align)
 
+#define SPA_SCALE32_UP(val,num,denom)				\
+({								\
+	uint64_t _val = (val);					\
+	uint64_t _denom = (denom);				\
+	(uint32_t)(((_val) * (num) + (_denom)-1) / (_denom));	\
+})
+
+
 #define SPA_PTR_ALIGNMENT(p,align)	((intptr_t)(p) & ((align)-1))
 #define SPA_IS_ALIGNED(p,align)		(SPA_PTR_ALIGNMENT(p,align) == 0)
 #define SPA_PTR_ALIGN(p,align,type)	(type*)SPA_ROUND_UP_N((intptr_t)(p), (intptr_t)(align))
# #define SPA_PTR_ALIGNMENT(p,align)	((intptr_t)(p) & ((align)-1))
# #define SPA_IS_ALIGNED(p,align)		(SPA_PTR_ALIGNMENT(p,align) == 0)
# #define SPA_PTR_ALIGN(p,align,type)	((type*)SPA_ROUND_UP_N((intptr_t)(p), (intptr_t)(align)))
diff --git a/spa/plugins/alsa/alsa-pcm.c b/spa/plugins/alsa/alsa-pcm.c
index 08b9ceddd..012b46031 100644
--- a/spa/plugins/alsa/alsa-pcm.c
+++ b/spa/plugins/alsa/alsa-pcm.c
@@ -1999,7 +1999,7 @@ static inline void check_position_config(struct state *state)
 	    (state->rate_denom != state->position->clock.rate.denom))) {
 		state->duration = state->position->clock.duration;
 		state->rate_denom = state->position->clock.rate.denom;
-		state->threshold = (state->duration * state->rate + state->rate_denom-1) / state->rate_denom;
+		state->threshold = SPA_SCALE32_UP(state->duration, state->rate, state->rate_denom);
 		state->resample = ((uint32_t)state->rate != state->rate_denom) || state->matching;
 		state->alsa_sync = true;
 	}
# 		state->max_error = SPA_MAX(256.0f, state->threshold / 2.0f);
# 		state->resample = ((uint32_t)state->rate != state->rate_denom) || state->matching;
# 		state->alsa_sync = true;
@@ -2569,7 +2569,7 @@ int spa_alsa_start(struct state *state)
# 	setup_matching(state);
# 
# 	spa_dll_init(&state->dll);
 	state->following = is_following(state);
 	setup_matching(state);
 
-	state->threshold = (state->duration * state->rate + state->rate_denom-1) / state->rate_denom;
+	state->threshold = SPA_SCALE32_UP(state->duration, state->rate, state->rate_denom);
 	state->last_threshold = state->threshold;
 
 	spa_dll_init(&state->dll);
# 	state->last_threshold = state->threshold;
# 	state->max_error = SPA_MAX(256.0f, state->threshold / 2.0f);
# 
#diff --git a/spa/plugins/audioconvert/audioconvert.c b/spa/plugins/audioconvert/audioconvert.c
#index 578f70ff9..783ab8174 100644
#--- a/spa/plugins/audioconvert/audioconvert.c
#+++ b/spa/plugins/audioconvert/audioconvert.c
#@@ -1755,7 +1755,7 @@ impl_node_port_enum_params(void *object, int seq,
# 			size = this->quantum_limit * 2;
# 			/*  scale the buffer size when we can. */
# 			if (irate != 0 && orate != 0)
#-				size = size * (irate + orate - 1) / orate;
#+				size = SPA_SCALE32_UP(size, irate, orate);
# 		}
# 
# 		param = spa_pod_builder_add_object(&b,
-- 
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