From patchwork Fri Feb 24 19:05:44 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: 13151722 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 F3783C7EE2D for ; Fri, 24 Feb 2023 19:07:48 +0000 (UTC) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pVdOT-0001u7-DU; Fri, 24 Feb 2023 14:06:09 -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 1pVdOS-0001tt-Ks for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:08 -0500 Received: from mailout08.t-online.de ([194.25.134.20]) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pVdOQ-0003mq-Tn for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:08 -0500 Received: from fwd74.dcpf.telekom.de (fwd74.aul.t-online.de [10.223.144.100]) by mailout08.t-online.de (Postfix) with SMTP id 3B4464E32; Fri, 24 Feb 2023 20:06:04 +0100 (CET) Received: from linpower.localnet ([84.175.228.75]) by fwd74.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1pVdON-4FqQS10; Fri, 24 Feb 2023 20:06:03 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id 93B51335527; 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 04/15] audio: replace the resampling loop in audio_pcm_sw_write() Date: Fri, 24 Feb 2023 20:05:44 +0100 Message-Id: <20230224190555.7409-4-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::1677265563-B202B007-2802F2FA/0/0 CLEAN NORMAL X-TOI-MSGID: a9b84f8d-2b49-40f1-bbae-6c6953a434c2 Received-SPF: none client-ip=194.25.134.20; envelope-from=volker.ruemelin@t-online.de; helo=mailout08.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_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 --- 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;