From patchwork Mon Feb 6 18:52:30 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: 1738327 Return-Path: X-Original-To: incoming@patchwork.ozlabs.org Delivered-To: patchwork-incoming@legolas.ozlabs.org Authentication-Results: legolas.ozlabs.org; spf=pass (sender SPF authorized) smtp.mailfrom=nongnu.org (client-ip=209.51.188.17; helo=lists.gnu.org; envelope-from=qemu-devel-bounces+incoming=patchwork.ozlabs.org@nongnu.org; receiver=) Received: from lists.gnu.org (lists.gnu.org [209.51.188.17]) (using TLSv1.2 with cipher ECDHE-ECDSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by legolas.ozlabs.org (Postfix) with ESMTPS id 4P9b9c2fhPz23r8 for ; Tue, 7 Feb 2023 05:57:40 +1100 (AEDT) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pP6bx-00051K-Mx; Mon, 06 Feb 2023 13:53:05 -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 1pP6bw-00050w-3I for qemu-devel@nongnu.org; Mon, 06 Feb 2023 13:53:04 -0500 Received: from mailout04.t-online.de ([194.25.134.18]) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pP6bu-0005VQ-Bn for qemu-devel@nongnu.org; Mon, 06 Feb 2023 13:53:03 -0500 Received: from fwd89.dcpf.telekom.de (fwd89.aul.t-online.de [10.223.144.115]) by mailout04.t-online.de (Postfix) with SMTP id AD4CF520B; Mon, 6 Feb 2023 19:52:59 +0100 (CET) Received: from linpower.localnet ([79.208.25.151]) by fwd89.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1pP6br-3nW74z0; Mon, 6 Feb 2023 19:52:59 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id 899E22006C1; 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 10/17] audio: wire up st_rate_frames_in() Date: Mon, 6 Feb 2023 19:52:30 +0100 Message-Id: <20230206185237.8358-10-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::1675709579-D571F046-7875955D/0/0 CLEAN NORMAL X-TOI-MSGID: c0a60082-f42e-429f-a42b-4e3a7b45a1b9 Received-SPF: none client-ip=194.25.134.18; envelope-from=volker.ruemelin@t-online.de; helo=mailout04.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+incoming=patchwork.ozlabs.org@nongnu.org Sender: qemu-devel-bounces+incoming=patchwork.ozlabs.org@nongnu.org Wire up the st_rate_frames_in() function and replace audio_frontend_frames_out() to make audio packet length calculation exact. When upsampling, it's still possible that the audio frontends can't write the last audio frame. This will be fixed later. Acked-by: Mark Cave-Ayland Signed-off-by: Volker Rümelin --- audio/audio.c | 43 ++++++++++++++++++------------------------- 1 file changed, 18 insertions(+), 25 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 556696b095..e18b5e98c5 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -701,8 +701,8 @@ static void audio_pcm_sw_resample_out(SWVoiceOut *sw, static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) { HWVoiceOut *hw = sw->hw; - size_t live, dead, hw_free; - size_t frames_in_max, total_in, total_out; + size_t live, dead, hw_free, sw_max, fe_max; + size_t frames_in_max, frames_out_max, total_in, total_out; live = sw->total_hw_samples_mixed; if (audio_bug(__func__, live > hw->mix_buf.size)) { @@ -720,17 +720,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) dead = hw->mix_buf.size - live; hw_free = audio_pcm_hw_get_free(hw); hw_free = hw_free > live ? hw_free - live : 0; - frames_in_max = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio; - frames_in_max = MIN(frames_in_max, buf_len / sw->info.bytes_per_frame); - if (frames_in_max) { - sw->conv(sw->resample_buf.buffer, buf, frames_in_max); + frames_out_max = MIN(dead, hw_free); + sw_max = st_rate_frames_in(sw->rate, frames_out_max); + fe_max = MIN(buf_len / sw->info.bytes_per_frame, sw->resample_buf.size); + frames_in_max = MIN(sw_max, fe_max); - if (!sw->hw->pcm_ops->volume_out) { - mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol); - } + if (!frames_in_max) { + return 0; } - audio_pcm_sw_resample_out(sw, frames_in_max, MIN(dead, hw_free), + sw->conv(sw->resample_buf.buffer, buf, frames_in_max); + if (!sw->hw->pcm_ops->volume_out) { + mixeng_volume(sw->resample_buf.buffer, frames_in_max, &sw->vol); + } + + audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max, &total_in, &total_out); sw->total_hw_samples_mixed += total_out; @@ -1000,18 +1004,6 @@ static size_t audio_get_avail (SWVoiceIn *sw) return live; } -/** - * audio_frontend_frames_out() - returns the number of frames needed to - * get frames_out frames after resampling - * - * @sw: audio playback frontend - * @frames_out: number of frames - */ -static size_t audio_frontend_frames_out(SWVoiceOut *sw, size_t frames_out) -{ - return ((int64_t)frames_out << 32) / sw->ratio; -} - static size_t audio_get_free(SWVoiceOut *sw) { size_t live, dead; @@ -1031,8 +1023,8 @@ static size_t audio_get_free(SWVoiceOut *sw) dead = sw->hw->mix_buf.size - live; #ifdef DEBUG_OUT - dolog("%s: get_free live %zu dead %zu frontend frames %zu\n", - SW_NAME(sw), live, dead, audio_frontend_frames_out(sw, dead)); + dolog("%s: get_free live %zu dead %zu frontend frames %u\n", + SW_NAME(sw), live, dead, st_rate_frames_in(sw->rate, dead)); #endif return dead; @@ -1161,12 +1153,13 @@ static void audio_run_out (AudioState *s) size_t free; if (hw_free > sw->total_hw_samples_mixed) { - free = audio_frontend_frames_out(sw, + free = st_rate_frames_in(sw->rate, MIN(sw_free, hw_free - sw->total_hw_samples_mixed)); } else { free = 0; } if (free > 0) { + free = MIN(free, sw->resample_buf.size); sw->callback.fn(sw->callback.opaque, free * sw->info.bytes_per_frame); }