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Message-ID: <s5h7ehgdiqq.wl-tiwai@suse.de>
Date: Tue, 13 Nov 2018 17:50:37 +0100
From: Takashi Iwai <tiwai@...e.de>
To: Mike Brady <mikebrady@...com.net>
Cc: Stefan Wahren <stefan.wahren@...e.com>, devel@...verdev.osuosl.org,
alsa-devel@...a-project.org, f.fainelli@...il.com,
Eric Anholt <eric@...olt.net>,
Greg Kroah-Hartman <gregkh@...uxfoundation.org>,
linux-kernel@...r.kernel.org, julia.lawall@...6.fr,
linux-rpi-kernel@...ts.infradead.org,
nishka.dasgupta_ug18@...oka.edu.in,
Kirill Marinushkin <k.marinushkin@...il.com>,
linux-arm-kernel@...ts.infradead.org
Subject: Re: [PATCH v2] ARM: staging: bcm2835-audio: interpolate audio delay
On Sun, 11 Nov 2018 19:21:29 +0100,
Mike Brady wrote:
>
> /* hardware definition */
> static const struct snd_pcm_hardware snd_bcm2835_playback_hw = {
> .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
> SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
> - SNDRV_PCM_INFO_DRAIN_TRIGGER | SNDRV_PCM_INFO_SYNC_APPLPTR),
> + SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_DRAIN_TRIGGER |
> + SNDRV_PCM_INFO_SYNC_APPLPTR),
As already mentioned, the addition of SNDRV_PCM_INFO_BATCH should be
irrelevant with this change. Please create another patch to add this
and clarify it in the changelog.
> diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
> index 4e6110d778bd..574df7d7a1fa 100644
> --- a/sound/core/pcm_lib.c
> +++ b/sound/core/pcm_lib.c
> @@ -229,19 +229,38 @@ static void update_audio_tstamp(struct snd_pcm_substream *substream,
> (runtime->audio_tstamp_report.actual_type ==
> SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT)) {
>
> - /*
> - * provide audio timestamp derived from pointer position
> - * add delay only if requested
> - */
> + // provide audio timestamp derived from pointer position
>
> audio_frames = runtime->hw_ptr_wrap + runtime->status->hw_ptr;
>
> - if (runtime->audio_tstamp_config.report_delay) {
> + /*
> + * If the runtime->delay is greater than zero, it's a
> + * genuine delay, e.g. a delay due to a hardware fifo.
> + *
> + * But if the runtime->delay is less than zero, it's an
> + * interpolated estimate of the number of frames output
> + * since the hardware pointer was last updated.
> + *
> + * It would be calculated in the pointer callback.
> + * For example, for the bcm_2835 driver, it is calculated in
> + * snd_bcm2835_pcm_pointer().
> + */
> +
> + if (runtime->delay < 0) {
> + // The delay is an interpolated estimate...
> if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
> - audio_frames -= runtime->delay;
> - else
> - audio_frames += runtime->delay;
> + audio_frames += runtime->delay;
> + } else {
> + // The delay is a real delay. Add it if requested.
> + if (runtime->audio_tstamp_config.report_delay) {
> + if (substream->stream ==
> + SNDRV_PCM_STREAM_PLAYBACK)
> + audio_frames -= runtime->delay;
> + else
> + audio_frames += runtime->delay;
> + }
> }
Well, honestly speaking, I'm really not fond of changing the PCM core
just for this.
Can we make bcm audio driver providing the finer pointer update
instead? If we have a module option to disable that behavior, it's an
enough excuse in case anyone really cares about the accuracy.
thanks,
Takashi
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