From patchwork Fri Feb 24 19:05:53 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: 13151712 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 E5C36C7EE23 for ; Fri, 24 Feb 2023 19:06:53 +0000 (UTC) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pVdP5-0003dC-Rv; Fri, 24 Feb 2023 14:06:48 -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 1pVdP3-0003Ou-Sf for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:46 -0500 Received: from mailout09.t-online.de ([194.25.134.84]) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pVdP0-0004j1-UZ for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:45 -0500 Received: from fwd75.dcpf.telekom.de (fwd75.aul.t-online.de [10.223.144.101]) by mailout09.t-online.de (Postfix) with SMTP id A4DA12709F; Fri, 24 Feb 2023 20:06:29 +0100 (CET) Received: from linpower.localnet ([84.175.228.75]) by fwd75.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1pVdOi-2kLnU10; Fri, 24 Feb 2023 20:06:24 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id AC8B8335531; 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 13/15] audio: handle leftover audio frame from upsampling Date: Fri, 24 Feb 2023 20:05:53 +0100 Message-Id: <20230224190555.7409-13-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::1677265584-DB61A2C4-2A984DD5/2/51138760147 SUSPECT URL-COUNT X-TOI-MSGID: b1fb9612-6db0-4773-99f3-03f526ddc73d Received-SPF: none client-ip=194.25.134.84; envelope-from=volker.ruemelin@t-online.de; helo=mailout09.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 Upsampling may leave one remaining audio frame in the input buffer. The emulated audio playback devices are currently resposible to write this audio frame again in the next write cycle. Push that task down to audio_pcm_sw_write. This is another step towards an audio callback interface that guarantees that when audio frontends are told they can write n audio frames, they can actually do so. Acked-by: Mark Cave-Ayland Acked-by: Marc-André Lureau Signed-off-by: Volker Rümelin --- audio/audio.c | 34 ++++++++++++++++++++++++++++------ audio/audio_template.h | 6 ++++++ 2 files changed, 34 insertions(+), 6 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index dad17e59b8..4836ab8ca8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -731,16 +731,21 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) hw_free = hw_free > live ? hw_free - live : 0; 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); + fe_max = MIN(buf_len / sw->info.bytes_per_frame + sw->resample_buf.pos, + sw->resample_buf.size); frames_in_max = MIN(sw_max, fe_max); if (!frames_in_max) { return 0; } - 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); + if (frames_in_max > sw->resample_buf.pos) { + sw->conv(sw->resample_buf.buffer + sw->resample_buf.pos, + buf, frames_in_max - sw->resample_buf.pos); + if (!sw->hw->pcm_ops->volume_out) { + mixeng_volume(sw->resample_buf.buffer + sw->resample_buf.pos, + frames_in_max - sw->resample_buf.pos, &sw->vol); + } } audio_pcm_sw_resample_out(sw, frames_in_max, frames_out_max, @@ -749,6 +754,22 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t buf_len) sw->total_hw_samples_mixed += total_out; sw->empty = sw->total_hw_samples_mixed == 0; + /* + * Upsampling may leave one audio frame in the resample buffer. Decrement + * total_in by one if there was a leftover frame from the previous resample + * pass in the resample buffer. Increment total_in by one if the current + * resample pass left one frame in the resample buffer. + */ + if (frames_in_max - total_in == 1) { + /* copy one leftover audio frame to the beginning of the buffer */ + *sw->resample_buf.buffer = *(sw->resample_buf.buffer + total_in); + total_in += 1 - sw->resample_buf.pos; + sw->resample_buf.pos = 1; + } else if (total_in >= sw->resample_buf.pos) { + total_in -= sw->resample_buf.pos; + sw->resample_buf.pos = 0; + } + #ifdef DEBUG_OUT dolog ( "%s: write size %zu written %zu total mixed %zu\n", @@ -1155,8 +1176,9 @@ static void audio_run_out (AudioState *s) } else { free = 0; } - if (free > 0) { - free = MIN(free, sw->resample_buf.size); + if (free > sw->resample_buf.pos) { + free = MIN(free, sw->resample_buf.size) + - sw->resample_buf.pos; sw->callback.fn(sw->callback.opaque, free * sw->info.bytes_per_frame); } diff --git a/audio/audio_template.h b/audio/audio_template.h index a0b653f52c..0d8aab6fad 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -138,6 +138,12 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) return -1; } + /* + * 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. + */ + samples = samples < INT_MAX ? samples + 1 : INT_MAX; sw->resample_buf.buffer = g_new0(st_sample, samples); sw->resample_buf.size = samples; sw->resample_buf.pos = 0;