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[153.243.13.130]) by smtp.gmail.com with ESMTPSA id w5-20020a17090aaf8500b001bd4c825deesm3039427pjq.43.2022.03.01.16.13.20 (version=TLS1_3 cipher=TLS_AES_128_GCM_SHA256 bits=128/128); Tue, 01 Mar 2022 16:13:22 -0800 (PST) Message-ID: Date: Wed, 2 Mar 2022 09:13:19 +0900 MIME-Version: 1.0 User-Agent: Mozilla/5.0 (X11; Linux aarch64; rv:91.0) Gecko/20100101 Thunderbird/91.5.0 Subject: Re: [PATCH v3 10/15] audio: restore mixing-engine playback buffer size Content-Language: en-US To: =?UTF-8?Q?Volker_R=c3=bcmelin?= , Gerd Hoffmann References: <3d0bd2ac-e5b9-9cf6-c98f-c047390a3ec5@t-online.de> <20220301191311.26695-10-vr_qemu@t-online.de> From: Akihiko Odaki In-Reply-To: <20220301191311.26695-10-vr_qemu@t-online.de> Content-Type: text/plain; charset=UTF-8; format=flowed Content-Transfer-Encoding: 8bit X-Host-Lookup-Failed: Reverse DNS lookup failed for 2607:f8b0:4864:20::42f (failed) Received-SPF: pass client-ip=2607:f8b0:4864:20::42f; envelope-from=akihiko.odaki@gmail.com; helo=mail-pf1-x42f.google.com X-Spam_score_int: -6 X-Spam_score: -0.7 X-Spam_bar: / X-Spam_report: (-0.7 / 5.0 requ) BAYES_00=-1.9, DKIM_SIGNED=0.1, DKIM_VALID=-0.1, DKIM_VALID_AU=-0.1, DKIM_VALID_EF=-0.1, FREEMAIL_FROM=0.001, NICE_REPLY_A=-0.001, PDS_HP_HELO_NORDNS=0.659, RCVD_IN_DNSWL_NONE=-0.0001, RDNS_NONE=0.793, SPF_HELO_NONE=0.001, SPF_PASS=-0.001, T_SCC_BODY_TEXT_LINE=-0.01 autolearn=no 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: , Cc: Thomas Huth , Christian Schoenebeck , qemu-devel@nongnu.org Errors-To: qemu-devel-bounces+qemu-devel=archiver.kernel.org@nongnu.org Sender: "Qemu-devel" Reviewed-by: Akihiko Odaki On 2022/03/02 4:13, Volker Rümelin wrote: > Commit ff095e5231 "audio: api for mixeng code free backends" > introduced another FIFO for the audio subsystem with exactly the > same size as the mixing-engine FIFO. Most audio backends use > this generic FIFO. The generic FIFO used together with the > mixing-engine FIFO doubles the audio FIFO size, because that's > just two independent FIFOs connected together in series. > > For audio playback this nearly doubles the playback latency. > > This patch restores the effective mixing-engine playback buffer > size to a pre v4.2.0 size by only accepting the amount of > samples for the mixing-engine queue which the downstream queue > accepts. > > Signed-off-by: Volker Rümelin > --- > audio/alsaaudio.c | 1 + > audio/audio.c | 69 +++++++++++++++++++++++++++++++++++------------ > audio/audio_int.h | 7 ++++- > audio/coreaudio.c | 3 +++ > audio/jackaudio.c | 1 + > audio/noaudio.c | 1 + > audio/ossaudio.c | 12 +++++++++ > audio/sdlaudio.c | 3 +++ > audio/wavaudio.c | 1 + > 9 files changed, 80 insertions(+), 18 deletions(-) > > diff --git a/audio/alsaaudio.c b/audio/alsaaudio.c > index 2b9789e647..b04716a6cc 100644 > --- a/audio/alsaaudio.c > +++ b/audio/alsaaudio.c > @@ -916,6 +916,7 @@ static struct audio_pcm_ops alsa_pcm_ops = { > .init_out = alsa_init_out, > .fini_out = alsa_fini_out, > .write = alsa_write, > + .buffer_get_free = audio_generic_buffer_get_free, > .run_buffer_out = audio_generic_run_buffer_out, > .enable_out = alsa_enable_out, > > diff --git a/audio/audio.c b/audio/audio.c > index c420a8bd1c..a88572e713 100644 > --- a/audio/audio.c > +++ b/audio/audio.c > @@ -663,6 +663,12 @@ static size_t audio_pcm_hw_get_live_out (HWVoiceOut *hw, int *nb_live) > return 0; > } > > +static size_t audio_pcm_hw_get_free(HWVoiceOut *hw) > +{ > + return (hw->pcm_ops->buffer_get_free ? hw->pcm_ops->buffer_get_free(hw) : > + INT_MAX) / hw->info.bytes_per_frame; > +} > + > static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) > { > size_t clipped = 0; > @@ -687,7 +693,8 @@ static void audio_pcm_hw_clip_out(HWVoiceOut *hw, void *pcm_buf, size_t len) > */ > static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) > { > - size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, swlim, blck; > + size_t hwsamples, samples, isamp, osamp, wpos, live, dead, left, blck; > + size_t hw_free; > size_t ret = 0, pos = 0, total = 0; > > if (!sw) { > @@ -710,27 +717,28 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) > } > > wpos = (sw->hw->mix_buf->pos + live) % hwsamples; > - samples = size / sw->info.bytes_per_frame; > > dead = hwsamples - live; > - swlim = ((int64_t) dead << 32) / sw->ratio; > - swlim = MIN (swlim, samples); > - if (swlim) { > - sw->conv (sw->buf, buf, swlim); > + hw_free = audio_pcm_hw_get_free(sw->hw); > + hw_free = hw_free > live ? hw_free - live : 0; > + samples = ((int64_t)MIN(dead, hw_free) << 32) / sw->ratio; > + samples = MIN(samples, size / sw->info.bytes_per_frame); > + if (samples) { > + sw->conv(sw->buf, buf, samples); > > if (!sw->hw->pcm_ops->volume_out) { > - mixeng_volume (sw->buf, swlim, &sw->vol); > + mixeng_volume(sw->buf, samples, &sw->vol); > } > } > > - while (swlim) { > + while (samples) { > dead = hwsamples - live; > left = hwsamples - wpos; > blck = MIN (dead, left); > if (!blck) { > break; > } > - isamp = swlim; > + isamp = samples; > osamp = blck; > st_rate_flow_mix ( > sw->rate, > @@ -740,7 +748,7 @@ static size_t audio_pcm_sw_write(SWVoiceOut *sw, void *buf, size_t size) > &osamp > ); > ret += isamp; > - swlim -= isamp; > + samples -= isamp; > pos += isamp; > live += osamp; > wpos = (wpos + osamp) % hwsamples; > @@ -1002,6 +1010,11 @@ static size_t audio_get_avail (SWVoiceIn *sw) > return (((int64_t) live << 32) / sw->ratio) * sw->info.bytes_per_frame; > } > > +static size_t audio_sw_bytes_free(SWVoiceOut *sw, size_t free) > +{ > + return (((int64_t)free << 32) / sw->ratio) * sw->info.bytes_per_frame; > +} > + > static size_t audio_get_free(SWVoiceOut *sw) > { > size_t live, dead; > @@ -1021,13 +1034,11 @@ 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 ret %" PRId64 "\n", > - SW_NAME (sw), > - live, dead, (((int64_t) dead << 32) / sw->ratio) * > - sw->info.bytes_per_frame); > + dolog("%s: get_free live %zu dead %zu sw_bytes %zu\n", > + SW_NAME(sw), live, dead, audio_sw_bytes_free(sw, dead)); > #endif > > - return (((int64_t) dead << 32) / sw->ratio) * sw->info.bytes_per_frame; > + return dead; > } > > static void audio_capture_mix_and_clear(HWVoiceOut *hw, size_t rpos, > @@ -1131,12 +1142,21 @@ static void audio_run_out (AudioState *s) > } > > while ((hw = audio_pcm_hw_find_any_enabled_out(s, hw))) { > - size_t played, live, prev_rpos, free; > + size_t played, live, prev_rpos; > + size_t hw_free = audio_pcm_hw_get_free(hw); > int nb_live; > > for (sw = hw->sw_head.lh_first; sw; sw = sw->entries.le_next) { > if (sw->active) { > - free = audio_get_free(sw); > + size_t sw_free = audio_get_free(sw); > + size_t free; > + > + if (hw_free > sw->total_hw_samples_mixed) { > + free = audio_sw_bytes_free(sw, > + MIN(sw_free, hw_free - sw->total_hw_samples_mixed)); > + } else { > + free = 0; > + } > if (free > 0) { > sw->callback.fn(sw->callback.opaque, free); > } > @@ -1398,6 +1418,15 @@ void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size) > hw->pending_emul -= size; > } > > +size_t audio_generic_buffer_get_free(HWVoiceOut *hw) > +{ > + if (hw->buf_emul) { > + return hw->size_emul - hw->pending_emul; > + } else { > + return hw->samples * hw->info.bytes_per_frame; > + } > +} > + > void audio_generic_run_buffer_out(HWVoiceOut *hw) > { > while (hw->pending_emul) { > @@ -1445,6 +1474,12 @@ size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size) > { > size_t total = 0; > > + if (hw->pcm_ops->buffer_get_free) { > + size_t free = hw->pcm_ops->buffer_get_free(hw); > + > + size = MIN(size, free); > + } > + > while (total < size) { > size_t dst_size = size - total; > size_t copy_size, proc; > diff --git a/audio/audio_int.h b/audio/audio_int.h > index 71be162271..2a6914d2aa 100644 > --- a/audio/audio_int.h > +++ b/audio/audio_int.h > @@ -161,10 +161,14 @@ struct audio_pcm_ops { > void (*fini_out)(HWVoiceOut *hw); > size_t (*write) (HWVoiceOut *hw, void *buf, size_t size); > void (*run_buffer_out)(HWVoiceOut *hw); > + /* > + * Get the free output buffer size. This is an upper limit. The size > + * returned by function get_buffer_out may be smaller. > + */ > + size_t (*buffer_get_free)(HWVoiceOut *hw); > /* > * get a buffer that after later can be passed to put_buffer_out; optional > * returns the buffer, and writes it's size to size (in bytes) > - * this is unrelated to the above buffer_size_out function > */ > void *(*get_buffer_out)(HWVoiceOut *hw, size_t *size); > /* > @@ -190,6 +194,7 @@ void audio_generic_run_buffer_in(HWVoiceIn *hw); > void *audio_generic_get_buffer_in(HWVoiceIn *hw, size_t *size); > void audio_generic_put_buffer_in(HWVoiceIn *hw, void *buf, size_t size); > void audio_generic_run_buffer_out(HWVoiceOut *hw); > +size_t audio_generic_buffer_get_free(HWVoiceOut *hw); > void *audio_generic_get_buffer_out(HWVoiceOut *hw, size_t *size); > size_t audio_generic_put_buffer_out(HWVoiceOut *hw, void *buf, size_t size); > size_t audio_generic_write(HWVoiceOut *hw, void *buf, size_t size); > diff --git a/audio/coreaudio.c b/audio/coreaudio.c > index 1fdd1d4b14..91ea6ae975 100644 > --- a/audio/coreaudio.c > +++ b/audio/coreaudio.c > @@ -283,6 +283,7 @@ static int coreaudio_buf_unlock (coreaudioVoiceOut *core, const char *fn_name) > coreaudio_buf_unlock(core, "coreaudio_" #name); \ > return ret; \ > } > +COREAUDIO_WRAPPER_FUNC(buffer_get_free, size_t, (HWVoiceOut *hw), (hw)) > COREAUDIO_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size), > (hw, size)) > COREAUDIO_WRAPPER_FUNC(put_buffer_out, size_t, > @@ -652,6 +653,8 @@ static struct audio_pcm_ops coreaudio_pcm_ops = { > .fini_out = coreaudio_fini_out, > /* wrapper for audio_generic_write */ > .write = coreaudio_write, > + /* wrapper for audio_generic_buffer_get_free */ > + .buffer_get_free = coreaudio_buffer_get_free, > /* wrapper for audio_generic_get_buffer_out */ > .get_buffer_out = coreaudio_get_buffer_out, > /* wrapper for audio_generic_put_buffer_out */ > diff --git a/audio/jackaudio.c b/audio/jackaudio.c > index 26246c3a8b..bf757250b5 100644 > --- a/audio/jackaudio.c > +++ b/audio/jackaudio.c > @@ -652,6 +652,7 @@ static struct audio_pcm_ops jack_pcm_ops = { > .init_out = qjack_init_out, > .fini_out = qjack_fini_out, > .write = qjack_write, > + .buffer_get_free = audio_generic_buffer_get_free, > .run_buffer_out = audio_generic_run_buffer_out, > .enable_out = qjack_enable_out, > > diff --git a/audio/noaudio.c b/audio/noaudio.c > index aac87dbc93..84a6bfbb1c 100644 > --- a/audio/noaudio.c > +++ b/audio/noaudio.c > @@ -118,6 +118,7 @@ static struct audio_pcm_ops no_pcm_ops = { > .init_out = no_init_out, > .fini_out = no_fini_out, > .write = no_write, > + .buffer_get_free = audio_generic_buffer_get_free, > .run_buffer_out = audio_generic_run_buffer_out, > .enable_out = no_enable_out, > > diff --git a/audio/ossaudio.c b/audio/ossaudio.c > index 60eff66424..1bd6800840 100644 > --- a/audio/ossaudio.c > +++ b/audio/ossaudio.c > @@ -389,6 +389,17 @@ static void oss_run_buffer_out(HWVoiceOut *hw) > } > } > > +static size_t oss_buffer_get_free(HWVoiceOut *hw) > +{ > + OSSVoiceOut *oss = (OSSVoiceOut *)hw; > + > + if (oss->mmapped) { > + return INT_MAX; > + } else { > + return audio_generic_buffer_get_free(hw); > + } > +} > + > static void *oss_get_buffer_out(HWVoiceOut *hw, size_t *size) > { > OSSVoiceOut *oss = (OSSVoiceOut *) hw; > @@ -750,6 +761,7 @@ static struct audio_pcm_ops oss_pcm_ops = { > .init_out = oss_init_out, > .fini_out = oss_fini_out, > .write = oss_write, > + .buffer_get_free = oss_buffer_get_free, > .run_buffer_out = oss_run_buffer_out, > .get_buffer_out = oss_get_buffer_out, > .put_buffer_out = oss_put_buffer_out, > diff --git a/audio/sdlaudio.c b/audio/sdlaudio.c > index d6f3aa1a9a..e605c787ba 100644 > --- a/audio/sdlaudio.c > +++ b/audio/sdlaudio.c > @@ -309,6 +309,7 @@ static void sdl_callback_in(void *opaque, Uint8 *buf, int len) > SDL_UnlockAudioDevice(sdl->devid); \ > } > > +SDL_WRAPPER_FUNC(buffer_get_free, size_t, (HWVoiceOut *hw), (hw), Out) > SDL_WRAPPER_FUNC(get_buffer_out, void *, (HWVoiceOut *hw, size_t *size), > (hw, size), Out) > SDL_WRAPPER_FUNC(put_buffer_out, size_t, > @@ -471,6 +472,8 @@ static struct audio_pcm_ops sdl_pcm_ops = { > .fini_out = sdl_fini_out, > /* wrapper for audio_generic_write */ > .write = sdl_write, > + /* wrapper for audio_generic_buffer_get_free */ > + .buffer_get_free = sdl_buffer_get_free, > /* wrapper for audio_generic_get_buffer_out */ > .get_buffer_out = sdl_get_buffer_out, > /* wrapper for audio_generic_put_buffer_out */ > diff --git a/audio/wavaudio.c b/audio/wavaudio.c > index 20e6853f85..ac666335c7 100644 > --- a/audio/wavaudio.c > +++ b/audio/wavaudio.c > @@ -197,6 +197,7 @@ static struct audio_pcm_ops wav_pcm_ops = { > .init_out = wav_init_out, > .fini_out = wav_fini_out, > .write = wav_write_out, > + .buffer_get_free = audio_generic_buffer_get_free, > .run_buffer_out = audio_generic_run_buffer_out, > .enable_out = wav_enable_out, > };