From patchwork Fri Feb 24 19:05:52 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: 13151716 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 BBA7FC6FA8E for ; Fri, 24 Feb 2023 19:07:27 +0000 (UTC) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pVdPC-0003xU-DD; Fri, 24 Feb 2023 14:06:54 -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 1pVdP6-0003eN-As for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:48 -0500 Received: from mailout01.t-online.de ([194.25.134.80]) by eggs.gnu.org with esmtps (TLS1.2:ECDHE_RSA_AES_256_GCM_SHA384:256) (Exim 4.90_1) (envelope-from ) id 1pVdP4-0004jW-GA for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:48 -0500 Received: from fwd89.dcpf.telekom.de (fwd89.aul.t-online.de [10.223.144.115]) by mailout01.t-online.de (Postfix) with SMTP id 76D8ED4AB; Fri, 24 Feb 2023 20:06:22 +0100 (CET) Received: from linpower.localnet ([84.175.228.75]) by fwd89.t-online.de with (TLSv1.3:TLS_AES_256_GCM_SHA384 encrypted) esmtp id 1pVdOg-1bTqzZ0; Fri, 24 Feb 2023 20:06:22 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id A9A77335530; 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 12/15] audio: make recording packet length calculation exact Date: Fri, 24 Feb 2023 20:05:52 +0100 Message-Id: <20230224190555.7409-12-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::1677265582-D5F1C046-69EB792A/0/0 CLEAN NORMAL X-TOI-MSGID: ae0f8922-1d0e-4199-b814-dc122626b5f6 Received-SPF: none client-ip=194.25.134.80; envelope-from=volker.ruemelin@t-online.de; helo=mailout01.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_H2=-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 Introduce the new function st_rate_frames_out() to calculate the exact number of audio output frames the resampling code can generate from a given number of audio input frames. When upsampling, this function returns the maximum number of output frames. This new function replaces the audio_frontend_frames_in() function, which calculated the average number of output frames rounded down to the nearest integer. The audio_frontend_frames_in() function was additionally used to limit the number of output frames to the resample buffer size. In audio_pcm_sw_read() the variable resample_buf.size replaces the open coded audio_frontend_frames_in() function. In audio_run_in() an additional MIN() function is necessary. After this patch the audio packet length calculation for audio recording is exact. Acked-by: Mark Cave-Ayland Signed-off-by: Volker Rümelin --- audio/audio.c | 29 ++++++++--------------------- audio/mixeng.c | 41 +++++++++++++++++++++++++++++++++++++++++ audio/mixeng.h | 1 + 3 files changed, 50 insertions(+), 21 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 22c36d6660..dad17e59b8 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -579,7 +579,7 @@ static void audio_pcm_sw_resample_in(SWVoiceIn *sw, static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len) { HWVoiceIn *hw = sw->hw; - size_t live, frames_out_max, swlim, total_in, total_out; + size_t live, frames_out_max, total_in, total_out; live = hw->total_samples_captured - sw->total_hw_samples_acquired; if (!live) { @@ -590,12 +590,10 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t buf_len) return 0; } - frames_out_max = buf_len / sw->info.bytes_per_frame; + frames_out_max = MIN(buf_len / sw->info.bytes_per_frame, + sw->resample_buf.size); - swlim = (live * sw->ratio) >> 32; - swlim = MIN(swlim, frames_out_max); - - audio_pcm_sw_resample_in(sw, live, swlim, &total_in, &total_out); + audio_pcm_sw_resample_in(sw, live, frames_out_max, &total_in, &total_out); if (!hw->pcm_ops->volume_in) { mixeng_volume(sw->resample_buf.buffer, total_out, &sw->vol); @@ -979,18 +977,6 @@ void AUD_set_active_in (SWVoiceIn *sw, int on) } } -/** - * audio_frontend_frames_in() - returns the number of frames the resampling - * code generates from frames_in frames - * - * @sw: audio recording frontend - * @frames_in: number of frames - */ -static size_t audio_frontend_frames_in(SWVoiceIn *sw, size_t frames_in) -{ - return (int64_t)frames_in * sw->ratio >> 32; -} - static size_t audio_get_avail (SWVoiceIn *sw) { size_t live; @@ -1007,9 +993,9 @@ static size_t audio_get_avail (SWVoiceIn *sw) } ldebug ( - "%s: get_avail live %zu frontend frames %zu\n", + "%s: get_avail live %zu frontend frames %u\n", SW_NAME (sw), - live, audio_frontend_frames_in(sw, live) + live, st_rate_frames_out(sw->rate, live) ); return live; @@ -1314,8 +1300,9 @@ static void audio_run_in (AudioState *s) size_t sw_avail = audio_get_avail(sw); size_t avail; - avail = audio_frontend_frames_in(sw, sw_avail); + avail = st_rate_frames_out(sw->rate, sw_avail); if (avail > 0) { + avail = MIN(avail, sw->resample_buf.size); sw->callback.fn(sw->callback.opaque, avail * sw->info.bytes_per_frame); } diff --git a/audio/mixeng.c b/audio/mixeng.c index a24c8c45a7..69f6549224 100644 --- a/audio/mixeng.c +++ b/audio/mixeng.c @@ -440,6 +440,47 @@ void st_rate_stop (void *opaque) g_free (opaque); } +/** + * st_rate_frames_out() - returns the number of frames the resampling code + * generates from frames_in frames + * + * @opaque: pointer to struct rate + * @frames_in: number of frames + * + * When upsampling, there may be more than one correct result. In this case, + * the function returns the maximum number of output frames the resampling + * code can generate. + */ +uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in) +{ + struct rate *rate = opaque; + uint64_t opos_end, opos_delta; + uint32_t ipos_end; + uint32_t frames_out; + + if (rate->opos_inc == 1ULL << 32) { + return frames_in; + } + + /* no output frame without at least one input frame */ + if (!frames_in) { + return 0; + } + + /* last frame read was at rate->ipos - 1 */ + ipos_end = rate->ipos - 1 + frames_in; + opos_end = (uint64_t)ipos_end << 32; + + /* last frame written was at rate->opos - rate->opos_inc */ + if (opos_end + rate->opos_inc <= rate->opos) { + return 0; + } + opos_delta = opos_end - rate->opos + rate->opos_inc; + frames_out = opos_delta / rate->opos_inc; + + return opos_delta % rate->opos_inc ? frames_out : frames_out - 1; +} + /** * st_rate_frames_in() - returns the number of frames needed to * get frames_out frames after resampling diff --git a/audio/mixeng.h b/audio/mixeng.h index 64c1e231cc..f9de7cffeb 100644 --- a/audio/mixeng.h +++ b/audio/mixeng.h @@ -52,6 +52,7 @@ void st_rate_flow(void *opaque, st_sample *ibuf, st_sample *obuf, void st_rate_flow_mix(void *opaque, st_sample *ibuf, st_sample *obuf, size_t *isamp, size_t *osamp); void st_rate_stop (void *opaque); +uint32_t st_rate_frames_out(void *opaque, uint32_t frames_in); uint32_t st_rate_frames_in(void *opaque, uint32_t frames_out); void mixeng_clear (struct st_sample *buf, int len); void mixeng_volume (struct st_sample *buf, int len, struct mixeng_volume *vol);