From patchwork Mon Feb 6 18:52:24 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: 13130503 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 EEC29C636D3 for ; Mon, 6 Feb 2023 18:55:58 +0000 (UTC) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pP6c8-0005E9-8I; Mon, 06 Feb 2023 13:53:16 -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 1pP6c2-00053U-IG for qemu-devel@nongnu.org; Mon, 06 Feb 2023 13:53:10 -0500 Received: from mailout02.t-online.de ([194.25.134.17]) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pP6c0-0005W0-Pu for qemu-devel@nongnu.org; Mon, 06 Feb 2023 13:53:10 -0500 Received: from fwd88.dcpf.telekom.de (fwd88.aul.t-online.de [10.223.144.114]) by mailout02.t-online.de (Postfix) with SMTP id C3DA425E0C; Mon, 6 Feb 2023 19:52:46 +0100 (CET) Received: from linpower.localnet ([79.208.25.151]) by fwd88.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1pP6bd-4LvybJ0; Mon, 6 Feb 2023 19:52:45 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id 7959A20062C; Mon, 6 Feb 2023 19:52:37 +0100 (CET) From: =?utf-8?q?Volker_R=C3=BCmelin?= To: Gerd Hoffmann , =?utf-8?q?Marc-Andr=C3=A9_Lureau?= Cc: qemu-devel@nongnu.org, Christian Schoenebeck , Mark Cave-Ayland Subject: [PATCH v2 04/17] audio: replace the resampling loop in audio_pcm_sw_write() Date: Mon, 6 Feb 2023 19:52:24 +0100 Message-Id: <20230206185237.8358-4-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::1675709565-70EA07D8-9CF8A3F1/0/0 CLEAN NORMAL X-TOI-MSGID: f3601ead-7c50-4406-9ba7-a522ccdb7adc Received-SPF: none client-ip=194.25.134.17; envelope-from=volker.ruemelin@t-online.de; helo=mailout02.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 Signed-off-by: Volker Rümelin Acked-by: Marc-André Lureau --- 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;