From patchwork Fri Feb 24 19:05:55 2023 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 8bit X-Patchwork-Submitter: =?utf-8?q?Volker_R=C3=BCmelin?= X-Patchwork-Id: 13151715 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 7B94CC7EE23 for ; Fri, 24 Feb 2023 19:07:27 +0000 (UTC) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pVdPF-0004NH-6S; Fri, 24 Feb 2023 14:06:57 -0500 Received: from eggs.gnu.org ([2001:470:142:3::10]) by lists.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pVdPC-00043f-T6 for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:54 -0500 Received: from mailout10.t-online.de ([194.25.134.21]) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pVdPB-0004kP-0m for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:54 -0500 Received: from fwd87.dcpf.telekom.de (fwd87.aul.t-online.de [10.223.144.113]) by mailout10.t-online.de (Postfix) with SMTP id A2C6C1583A; Fri, 24 Feb 2023 20:06:29 +0100 (CET) Received: from linpower.localnet ([84.175.228.75]) by fwd87.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1pVdOn-26bgDx0; Fri, 24 Feb 2023 20:06:29 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id B1D45335533; Fri, 24 Feb 2023 20:05:55 +0100 (CET) From: =?utf-8?q?Volker_R=C3=BCmelin?= To: Gerd Hoffmann , =?utf-8?q?Marc-Andr=C3=A9_Lureau?= Cc: Christian Schoenebeck , Mark Cave-Ayland , qemu-devel@nongnu.org Subject: [PATCH v3 15/15] audio: remove sw->ratio Date: Fri, 24 Feb 2023 20:05:55 +0100 Message-Id: <20230224190555.7409-15-vr_qemu@t-online.de> X-Mailer: git-send-email 2.35.3 In-Reply-To: References: MIME-Version: 1.0 X-TOI-EXPURGATEID: 150726::1677265589-9F29E0AF-2C711AB9/2/51138760147 SUSPECT URL-COUNT X-TOI-MSGID: 550620b0-94f6-474e-8fa6-052d93e5e5c2 Received-SPF: none client-ip=194.25.134.21; envelope-from=volker.ruemelin@t-online.de; helo=mailout10.t-online.de X-Spam_score_int: -25 X-Spam_score: -2.6 X-Spam_bar: -- X-Spam_report: (-2.6 / 5.0 requ) BAYES_00=-1.9, FREEMAIL_FROM=0.001, RCVD_IN_DNSWL_LOW=-0.7, 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-bounces+qemu-devel=archiver.kernel.org@nongnu.org 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 --- 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) {