From patchwork Fri Feb 24 19:05:41 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: 1747662 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 4PNfWn1mGsz240f for ; Sat, 25 Feb 2023 06:06:45 +1100 (AEDT) Received: from localhost ([::1] helo=lists1p.gnu.org) by lists.gnu.org with esmtp (Exim 4.90_1) (envelope-from ) id 1pVdOO-0001ls-9Z; Fri, 24 Feb 2023 14:06:04 -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 1pVdON-0001lc-6x for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:03 -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 1pVdOK-0003lK-MU for qemu-devel@nongnu.org; Fri, 24 Feb 2023 14:06:02 -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 51E4B16CE5; Fri, 24 Feb 2023 20:05:56 +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 1pVdOF-3ZdQ010; Fri, 24 Feb 2023 20:05:55 +0100 Received: by linpower.localnet (Postfix, from userid 1000) id 8CB13200626; 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 01/15] audio: change type of mix_buf and conv_buf Date: Fri, 24 Feb 2023 20:05:41 +0100 Message-Id: <20230224190555.7409-1-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::1677265556-CFFFF046-A0C207CB/0/0 CLEAN NORMAL X-TOI-MSGID: d36e48dc-d081-4edd-a552-c8a2d8dd1ffc 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+incoming=patchwork.ozlabs.org@nongnu.org Sender: qemu-devel-bounces+incoming=patchwork.ozlabs.org@nongnu.org Change the type of mix_buf in struct HWVoiceOut and conv_buf in struct HWVoiceIn from STSampleBuffer * to STSampleBuffer. However, a buffer pointer is still needed. For this reason in struct STSampleBuffer samples[] is changed to *buffer. This is a preparation for the next patch. The next patch will add this line, which is not possible with the current struct STSampleBuffer definition. + sw->resample_buf.buffer = hw->mix_buf.buffer + rpos2; There are no functional changes. Acked-by: Mark Cave-Ayland Reviewed-by: Marc-André Lureau Signed-off-by: Volker Rümelin --- audio/audio.c | 106 ++++++++++++++++++++--------------------- audio/audio_int.h | 6 +-- audio/audio_template.h | 19 ++++---- 3 files changed, 67 insertions(+), 64 deletions(-) diff --git a/audio/audio.c b/audio/audio.c index 772c3cc320..a0b54e4a2e 100644 --- a/audio/audio.c +++ b/audio/audio.c @@ -523,8 +523,8 @@ static size_t audio_pcm_hw_find_min_in (HWVoiceIn *hw) static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) { size_t live = hw->total_samples_captured - audio_pcm_hw_find_min_in (hw); - if (audio_bug(__func__, live > hw->conv_buf->size)) { - dolog("live=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); + if (audio_bug(__func__, live > hw->conv_buf.size)) { + dolog("live=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); return 0; } return live; @@ -533,13 +533,13 @@ static size_t audio_pcm_hw_get_live_in(HWVoiceIn *hw) static size_t audio_pcm_hw_conv_in(HWVoiceIn *hw, void *pcm_buf, size_t samples) { size_t conv = 0; - STSampleBuffer *conv_buf = hw->conv_buf; + STSampleBuffer *conv_buf = &hw->conv_buf; while (samples) { uint8_t *src = advance(pcm_buf, conv * hw->info.bytes_per_frame); size_t proc = MIN(samples, conv_buf->size - conv_buf->pos); - hw->conv(conv_buf->samples + conv_buf->pos, src, proc); + hw->conv(conv_buf->buffer + conv_buf->pos, src, proc); conv_buf->pos = (conv_buf->pos + proc) % conv_buf->size; samples -= proc; conv += proc; @@ -561,12 +561,12 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) if (!live) { return 0; } - if (audio_bug(__func__, live > hw->conv_buf->size)) { - dolog("live_in=%zu hw->conv_buf->size=%zu\n", live, hw->conv_buf->size); + if (audio_bug(__func__, live > hw->conv_buf.size)) { + dolog("live_in=%zu hw->conv_buf.size=%zu\n", live, hw->conv_buf.size); return 0; } - rpos = audio_ring_posb(hw->conv_buf->pos, live, hw->conv_buf->size); + rpos = audio_ring_posb(hw->conv_buf.pos, live, hw->conv_buf.size); samples = size / sw->info.bytes_per_frame; @@ -574,11 +574,11 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) swlim = MIN (swlim, samples); while (swlim) { - src = hw->conv_buf->samples + rpos; - if (hw->conv_buf->pos > rpos) { - isamp = hw->conv_buf->pos - rpos; + src = hw->conv_buf.buffer + rpos; + if (hw->conv_buf.pos > rpos) { + isamp = hw->conv_buf.pos - rpos; } else { - isamp = hw->conv_buf->size - rpos; + isamp = hw->conv_buf.size - rpos; } if (!isamp) { @@ -588,7 +588,7 @@ static size_t audio_pcm_sw_read(SWVoiceIn *sw, void *buf, size_t size) st_rate_flow (sw->rate, src, dst, &isamp, &osamp); swlim -= osamp; - rpos = (rpos + isamp) % hw->conv_buf->size; + rpos = (rpos + isamp) % hw->conv_buf.size; dst += osamp; ret += osamp; total += isamp; @@ -636,8 +636,8 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) if (nb_live1) { size_t live = smin; - if (audio_bug(__func__, live > hw->mix_buf->size)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); + if (audio_bug(__func__, live > hw->mix_buf.size)) { + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); return 0; } return live; @@ -654,17 +654,17 @@ static size_t audio_pcm_hw_get_free(HWVoiceOut *hw) static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) { size_t clipped = 0; - size_t pos = hw->mix_buf->pos; + size_t pos = hw->mix_buf.pos; while (len) { - st_sample *src = hw->mix_buf->samples + pos; + st_sample *src = hw->mix_buf.buffer + pos; uint8_t *dst = advance(pcm_buf, clipped * hw->info.bytes_per_frame); - size_t samples_till_end_of_buf = hw->mix_buf->size - pos; + size_t samples_till_end_of_buf = hw->mix_buf.size - pos; size_t samples_to_clip = MIN(len, samples_till_end_of_buf); hw->clip(dst, src, samples_to_clip); - pos = (pos + samples_to_clip) % hw->mix_buf->size; + pos = (pos + samples_to_clip) % hw->mix_buf.size; len -= samples_to_clip; clipped += samples_to_clip; } @@ -683,11 +683,11 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) return size; } - hwsamples = sw->hw->mix_buf->size; + hwsamples = sw->hw->mix_buf.size; live = sw->total_hw_samples_mixed; if (audio_bug(__func__, live > hwsamples)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hwsamples); + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hwsamples); return 0; } @@ -698,7 +698,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) return 0; } - wpos = (sw->hw->mix_buf->pos + live) % hwsamples; + wpos = (sw->hw->mix_buf.pos + live) % hwsamples; dead = hwsamples - live; hw_free = audio_pcm_hw_get_free(sw->hw); @@ -725,7 +725,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) st_rate_flow_mix ( sw->rate, sw->buf + pos, - sw->hw->mix_buf->samples + wpos, + sw->hw->mix_buf.buffer + wpos, &isamp, &osamp ); @@ -989,9 +989,9 @@ static size_t audio_get_avail (SWVoiceIn *sw) } live = sw->hw->total_samples_captured - sw->total_hw_samples_acquired; - if (audio_bug(__func__, live > sw->hw->conv_buf->size)) { - dolog("live=%zu sw->hw->conv_buf->size=%zu\n", live, - sw->hw->conv_buf->size); + if (audio_bug(__func__, live > sw->hw->conv_buf.size)) { + dolog("live=%zu sw->hw->conv_buf.size=%zu\n", live, + sw->hw->conv_buf.size); return 0; } @@ -1026,13 +1026,13 @@ static size_t audio_get_free(SWVoiceOut *sw) live = sw->total_hw_samples_mixed; - if (audio_bug(__func__, live > sw->hw->mix_buf->size)) { - dolog("live=%zu sw->hw->mix_buf->size=%zu\n", live, - sw->hw->mix_buf->size); + if (audio_bug(__func__, live > sw->hw->mix_buf.size)) { + dolog("live=%zu sw->hw->mix_buf.size=%zu\n", live, + sw->hw->mix_buf.size); return 0; } - dead = sw->hw->mix_buf->size - live; + dead = sw->hw->mix_buf.size - live; #ifdef DEBUG_OUT dolog("%s: get_free live %zu dead %zu frontend frames %zu\n", @@ -1056,12 +1056,12 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, n = samples; while (n) { - size_t till_end_of_hw = hw->mix_buf->size - rpos2; + size_t till_end_of_hw = hw->mix_buf.size - rpos2; size_t to_write = MIN(till_end_of_hw, n); size_t bytes = to_write * hw->info.bytes_per_frame; size_t written; - sw->buf = hw->mix_buf->samples + rpos2; + sw->buf = hw->mix_buf.buffer + rpos2; written = audio_pcm_sw_write (sw, NULL, bytes); if (written - bytes) { dolog("Could not mix %zu bytes into a capture " @@ -1070,14 +1070,14 @@ static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, break; } n -= to_write; - rpos2 = (rpos2 + to_write) % hw->mix_buf->size; + rpos2 = (rpos2 + to_write) % hw->mix_buf.size; } } } - n = MIN(samples, hw->mix_buf->size - rpos); - mixeng_clear(hw->mix_buf->samples + rpos, n); - mixeng_clear(hw->mix_buf->samples, samples - n); + n = MIN(samples, hw->mix_buf.size - rpos); + mixeng_clear(hw->mix_buf.buffer + rpos, n); + mixeng_clear(hw->mix_buf.buffer, samples - n); } static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) @@ -1103,7 +1103,7 @@ static size_t audio_pcm_hw_run_out(HWVoiceOut *hw, size_t live) live -= proc; clipped += proc; - hw->mix_buf->pos = (hw->mix_buf->pos + proc) % hw->mix_buf->size; + hw->mix_buf.pos = (hw->mix_buf.pos + proc) % hw->mix_buf.size; if (proc == 0 || proc < decr) { break; @@ -1174,8 +1174,8 @@ static void audio_run_out (AudioState *s) live = 0; } - if (audio_bug(__func__, live > hw->mix_buf->size)) { - dolog("live=%zu hw->mix_buf->size=%zu\n", live, hw->mix_buf->size); + if (audio_bug(__func__, live > hw->mix_buf.size)) { + dolog("live=%zu hw->mix_buf.size=%zu\n", live, hw->mix_buf.size); continue; } @@ -1203,13 +1203,13 @@ static void audio_run_out (AudioState *s) continue; } - prev_rpos = hw->mix_buf->pos; + prev_rpos = hw->mix_buf.pos; played = audio_pcm_hw_run_out(hw, live); replay_audio_out(&played); - if (audio_bug(__func__, hw->mix_buf->pos >= hw->mix_buf->size)) { - dolog("hw->mix_buf->pos=%zu hw->mix_buf->size=%zu played=%zu\n", - hw->mix_buf->pos, hw->mix_buf->size, played); - hw->mix_buf->pos = 0; + if (audio_bug(__func__, hw->mix_buf.pos >= hw->mix_buf.size)) { + dolog("hw->mix_buf.pos=%zu hw->mix_buf.size=%zu played=%zu\n", + hw->mix_buf.pos, hw->mix_buf.size, played); + hw->mix_buf.pos = 0; } #ifdef DEBUG_OUT @@ -1290,10 +1290,10 @@ static void audio_run_in (AudioState *s) if (replay_mode != REPLAY_MODE_PLAY) { captured = audio_pcm_hw_run_in( - hw, hw->conv_buf->size - audio_pcm_hw_get_live_in(hw)); + hw, hw->conv_buf.size - audio_pcm_hw_get_live_in(hw)); } - replay_audio_in(&captured, hw->conv_buf->samples, &hw->conv_buf->pos, - hw->conv_buf->size); + replay_audio_in(&captured, hw->conv_buf.buffer, &hw->conv_buf.pos, + hw->conv_buf.size); min = audio_pcm_hw_find_min_in (hw); hw->total_samples_captured += captured - min; @@ -1326,14 +1326,14 @@ static void audio_run_capture (AudioState *s) SWVoiceOut *sw; captured = live = audio_pcm_hw_get_live_out (hw, NULL); - rpos = hw->mix_buf->pos; + rpos = hw->mix_buf.pos; while (live) { - size_t left = hw->mix_buf->size - rpos; + size_t left = hw->mix_buf.size - rpos; size_t to_capture = MIN(live, left); struct st_sample *src; struct capture_callback *cb; - src = hw->mix_buf->samples + rpos; + src = hw->mix_buf.buffer + rpos; hw->clip (cap->buf, src, to_capture); mixeng_clear (src, to_capture); @@ -1341,10 +1341,10 @@ static void audio_run_capture (AudioState *s) cb->ops.capture (cb->opaque, cap->buf, to_capture * hw->info.bytes_per_frame); } - rpos = (rpos + to_capture) % hw->mix_buf->size; + rpos = (rpos + to_capture) % hw->mix_buf.size; live -= to_capture; } - hw->mix_buf->pos = rpos; + hw->mix_buf.pos = rpos; for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { if (!sw->active && sw->empty) { @@ -1903,7 +1903,7 @@ CaptureVoiceOut *AUD_add_capture( audio_pcm_init_info (&hw->info, as); - cap->buf = g_malloc0_n(hw->mix_buf->size, hw->info.bytes_per_frame); + cap->buf = g_malloc0_n(hw->mix_buf.size, hw->info.bytes_per_frame); if (hw->info.is_float) { hw->clip = mixeng_clip_float[hw->info.nchannels == 2]; @@ -1955,7 +1955,7 @@ void AUD_del_capture (CaptureVoiceOut *cap, void *cb_opaque) sw = sw1; } QLIST_REMOVE (cap, entries); - g_free (cap->hw.mix_buf); + g_free(cap->hw.mix_buf.buffer); g_free (cap->buf); g_free (cap); } diff --git a/audio/audio_int.h b/audio/audio_int.h index 5028f2354a..061845dcc2 100644 --- a/audio/audio_int.h +++ b/audio/audio_int.h @@ -58,7 +58,7 @@ typedef struct SWVoiceCap SWVoiceCap; typedef struct STSampleBuffer { size_t pos, size; - st_sample samples[]; + st_sample *buffer; } STSampleBuffer; typedef struct HWVoiceOut { @@ -71,7 +71,7 @@ typedef struct HWVoiceOut { f_sample *clip; uint64_t ts_helper; - STSampleBuffer *mix_buf; + STSampleBuffer mix_buf; void *buf_emul; size_t pos_emul, pending_emul, size_emul; @@ -93,7 +93,7 @@ typedef struct HWVoiceIn { size_t total_samples_captured; uint64_t ts_helper; - STSampleBuffer *conv_buf; + STSampleBuffer conv_buf; void *buf_emul; size_t pos_emul, pending_emul, size_emul; diff --git a/audio/audio_template.h b/audio/audio_template.h index 980e1f4bd0..dd87170cbd 100644 --- a/audio/audio_template.h +++ b/audio/audio_template.h @@ -71,8 +71,9 @@ static void glue(audio_init_nb_voices_, TYPE)(AudioState *s, static void glue (audio_pcm_hw_free_resources_, TYPE) (HW *hw) { g_free(hw->buf_emul); - g_free (HWBUF); - HWBUF = NULL; + g_free(HWBUF.buffer); + HWBUF.buffer = NULL; + HWBUF.size = 0; } static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) @@ -83,10 +84,12 @@ static void glue(audio_pcm_hw_alloc_resources_, TYPE)(HW *hw) dolog("Attempted to allocate empty buffer\n"); } - HWBUF = g_malloc0(sizeof(STSampleBuffer) + sizeof(st_sample) * samples); - HWBUF->size = samples; + HWBUF.buffer = g_new0(st_sample, samples); + HWBUF.size = samples; + HWBUF.pos = 0; } else { - HWBUF = NULL; + HWBUF.buffer = NULL; + HWBUF.size = 0; } } @@ -111,9 +114,9 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) } #ifdef DAC - samples = ((int64_t) sw->HWBUF->size << 32) / sw->ratio; + samples = ((int64_t)sw->HWBUF.size << 32) / sw->ratio; #else - samples = (int64_t)sw->HWBUF->size * sw->ratio >> 32; + samples = (int64_t)sw->HWBUF.size * sw->ratio >> 32; #endif if (audio_bug(__func__, samples < 0)) { dolog("Can not allocate buffer for `%s' (%d samples)\n", @@ -126,7 +129,7 @@ static int glue (audio_pcm_sw_alloc_resources_, TYPE) (SW *sw) size_t f_fe_min; /* f_fe_min = ceil(1 [frames] * f_be [Hz] / size_be [frames]) */ - f_fe_min = (hw->info.freq + HWBUF->size - 1) / HWBUF->size; + f_fe_min = (hw->info.freq + HWBUF.size - 1) / HWBUF.size; qemu_log_mask(LOG_UNIMP, AUDIO_CAP ": The guest selected a " NAME " sample rate" " of %d Hz for %s. Only sample rates >= %zu Hz are"