From patchwork Fri Feb 24 19:05:50 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: 13151717 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 7464FC7EE23 for ; Fri, 24 Feb 2023 19:07:36 +0000 (UTC) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pVdOg-000264-Ms; Fri, 24 Feb 2023 14:06:22 -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 1pVdOf-00025p-No for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:21 -0500 Received: from mailout03.t-online.de ([194.25.134.81]) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pVdOe-0004QE-3z for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:21 -0500 Received: from fwd83.dcpf.telekom.de (fwd83.aul.t-online.de [10.223.144.109]) by mailout03.t-online.de (Postfix) with SMTP id 9D751227E9; Fri, 24 Feb 2023 20:06:17 +0100 (CET) Received: from linpower.localnet ([84.175.228.75]) by fwd83.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1pVdOb-0OvfuL0; Fri, 24 Feb 2023 20:06:17 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id A428733552E; 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 10/15] audio: replace the resampling loop in audio_pcm_sw_read() Date: Fri, 24 Feb 2023 20:05:50 +0100 Message-Id: <20230224190555.7409-10-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::1677265577-422C1378-AF44DAA5/0/0 CLEAN NORMAL X-TOI-MSGID: a505ce50-5bdb-415e-9774-938744978cf7 Received-SPF: none client-ip=194.25.134.81; envelope-from=volker.ruemelin@t-online.de; helo=mailout03.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_H3=0.001, RCVD_IN_MSPIKE_WL=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 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 --- 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);