From mboxrd@z Thu Jan 1 00:00:00 1970 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org Received: from lists.gnu.org (lists.gnu.org [209.51.188.17]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by smtp.lore.kernel.org (Postfix) with ESMTPS id 435FEC07E9D for ; Fri, 23 Sep 2022 18:48:53 +0000 (UTC) Received: from localhost ([::1]:47236 helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1obnjH-0000FU-3Y for qemu-devel@archiver.kernel.org; Fri, 23 Sep 2022 14:48:51 -0400 Received: from eggs.gnu.org ([2001:470:142:3::10]:40158) by lists.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1obnXw-0002mq-OD for qemu-devel@nongnu.org; Fri, 23 Sep 2022 14:37:15 -0400 Received: from mailout11.t-online.de ([194.25.134.85]:47566) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1obnXv-0005Gn-0j for qemu-devel@nongnu.org; Fri, 23 Sep 2022 14:37:08 -0400 Received: from fwd76.dcpf.telekom.de (fwd76.aul.t-online.de [10.223.144.102]) by mailout11.t-online.de (Postfix) with SMTP id 047CA127EA; Fri, 23 Sep 2022 20:37:05 +0200 (CEST) Received: from linpower.localnet ([84.175.228.229]) by fwd76.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1obnXs-2r31Kj0; Fri, 23 Sep 2022 20:37:04 +0200 Received: by linpower.localnet (Postfix, from userid 1000) id B0392200620; Fri, 23 Sep 2022 20:36:40 +0200 (CEST) From: =?UTF-8?q?Volker=20R=C3=BCmelin?= To: Gerd Hoffmann Cc: qemu-devel@nongnu.org Subject: [PATCH 11/12] audio: fix sw->buf size for audio recording Date: Fri, 23 Sep 2022 20:36:39 +0200 Message-Id: <20220923183640.8314-11-vr_qemu@t-online.de> X-Mailer: git-send-email 2.35.3 In-Reply-To: <38d7417a-fc26-be39-6a63-99c79ee15fc9@t-online.de> References: <38d7417a-fc26-be39-6a63-99c79ee15fc9@t-online.de> MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit X-TOI-EXPURGATEID: 150726::1663958224-EEFF2A26-51EF2029/0/0 CLEAN NORMAL X-TOI-MSGID: cb28168f-647c-4334-a556-853adb77d7d4 Received-SPF: none client-ip=194.25.134.85; envelope-from=volker.ruemelin@t-online.de; helo=mailout11.t-online.de X-Spam_score_int: -18 X-Spam_score: -1.9 X-Spam_bar: - X-Spam_report: (-1.9 / 5.0 requ) BAYES_00=-1.9, FREEMAIL_FROM=0.001, RCVD_IN_DNSWL_NONE=-0.0001, RCVD_IN_MSPIKE_H2=-0.001, SPF_HELO_NONE=0.001, SPF_NONE=0.001 autolearn=ham autolearn_force=no X-Spam_action: no action X-BeenThere: qemu-devel@nongnu.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: qemu-devel-bounces+qemu-devel=archiver.kernel.org@nongnu.org Sender: "Qemu-devel" The calculation of the buffer size needed to store audio samples after resampling is wrong for audio recording. For audio recording sw->ratio is calculated as sw->ratio = frontend sample rate / backend sample rate. >From this follows frontend samples = frontend sample rate / backend sample rate * backend samples frontend samples = sw->ratio * backend samples In 2 of 3 places in the audio recording code where sw->ratio is used in a calculation to get the number of frontend frames, the calculation is wrong. Fix this. The 3rd formula in audio_pcm_sw_read() is correct. Resolves: https://gitlab.com/qemu-project/qemu/-/issues/71 Signed-off-by: Volker Rümelin --- audio/audio.c | 2 +- audio/audio_template.h | 4 ++++ 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/audio/audio.c b/audio/audio.c index ba0c62b120..60c7472d37 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -995,7 +995,7 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) */ static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in) { - return ((int64_t)frames_in << 32) / sw->ratio; + return (int64_t)frames_in * sw->ratio >> 32; } static size_t audio_get_avail (SWVoiceIn *sw) diff --git a/audio/audio_template.h b/audio/audio_template.h index 7192b19e73..6a0337ac6b 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -112,7 +112,11 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) return 0; } +#ifdef DAC samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio; +#else + samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; +#endif sw->buf = audio_calloc(__func__, samples, sizeof(struct st_sample)); if (!sw->buf) { -- 2.35.3