[alsa-devel] [PATCH 0/2 v2] usb-audio misc fix
This patch set contains the following patches
Andreas Pape (1): ALSA: usb-audio: more tolerant packetsize
Daniel Girnus (1): ALSA: usb-audio: avoid setting of sample rate multiple times on bus
sound/usb/endpoint.c | 4 ++-- sound/usb/pcm.c | 21 +++++++++++---------- 2 files changed, 13 insertions(+), 12 deletions(-)
From: Andreas Pape apape@de.adit-jv.com
since commit 57e6dae1087b ("ALSA: usb-audio: do not trust too-big wMaxPacketSize values"), the expected packetsize is always limited to nominal + 25%. It was discovered, that some devices have a much higher jitter in used packetsizes than 25% which would result in BABBLE condition and dropping of packets. A better solution is so assume the jitter to be the nominal packetsize: -one nearly empty packet followed by a almost 150% sized one.
V2: changed to assume max frequency is +50 of nominal packetsize.
Signed-off-by: Andreas Pape apape@de.adit-jv.com Signed-off-by: Jiada Wang jiada_wang@mentor.com --- sound/usb/endpoint.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c470251..a2931f4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -632,8 +632,8 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0;
- /* assume max. frequency is 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); + /* assume max. frequency is 50% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 1); /* Round up freqmax to nearest integer in order to calculate maximum * packet size, which must represent a whole number of frames. * This is accomplished by adding 0x0.ffff before converting the
On Tue, 06 Dec 2016 06:46:14 +0100, Jiada Wang wrote:
From: Andreas Pape apape@de.adit-jv.com
since commit 57e6dae1087b ("ALSA: usb-audio: do not trust too-big wMaxPacketSize values"), the expected packetsize is always limited to nominal + 25%. It was discovered, that some devices have a much higher jitter in used packetsizes than 25% which would result in BABBLE condition and dropping of packets. A better solution is so assume the jitter to be the nominal packetsize: -one nearly empty packet followed by a almost 150% sized one.
V2: changed to assume max frequency is +50 of nominal packetsize.
Clemens, are you OK with this change?
Takashi
Signed-off-by: Andreas Pape apape@de.adit-jv.com Signed-off-by: Jiada Wang jiada_wang@mentor.com
sound/usb/endpoint.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c470251..a2931f4 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -632,8 +632,8 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0;
- /* assume max. frequency is 25% higher than nominal */
- ep->freqmax = ep->freqn + (ep->freqn >> 2);
- /* assume max. frequency is 50% higher than nominal */
- ep->freqmax = ep->freqn + (ep->freqn >> 1); /* Round up freqmax to nearest integer in order to calculate maximum
- packet size, which must represent a whole number of frames.
- This is accomplished by adding 0x0.ffff before converting the
-- 2.9.3
Jiada Wang wrote:
since commit 57e6dae1087b ("ALSA: usb-audio: do not trust too-big wMaxPacketSize values"), the expected packetsize is always limited to nominal + 25%. It was discovered, that some devices
Android audio accessory
have a much higher jitter in used packetsizes than 25% which would result in BABBLE condition and dropping of packets. A better solution is so assume the jitter to be the nominal packetsize: -one nearly empty packet followed by a almost 150% sized one.
V2: changed to assume max frequency is +50 of nominal packetsize.
Signed-off-by: Andreas Pape apape@de.adit-jv.com Signed-off-by: Jiada Wang jiada_wang@mentor.com
Acked-by: Clemens Ladisch clemens@ladisch.de
--- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -632,8 +632,8 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0;
- /* assume max. frequency is 25% higher than nominal */
- ep->freqmax = ep->freqn + (ep->freqn >> 2);
- /* assume max. frequency is 50% higher than nominal */
- ep->freqmax = ep->freqn + (ep->freqn >> 1); /* Round up freqmax to nearest integer in order to calculate maximum
- packet size, which must represent a whole number of frames.
- This is accomplished by adding 0x0.ffff before converting the
From: Daniel Girnus dgirnus@de.adit-jv.com
Some of userland applications call 'snd_pcm_hw_params()' and 'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()' is called twice and the second 'snd_pcm_hw_prepare()' is called in 'SNDRV_PCM_STATE_PREPARED' state.
Some devices are not able to manage this and they will stop playback if the sample rate will be configured several times over USB protocol.
V2: updated Changelog
Signed-off-by: Daniel Girnus dgirnus@de.adit-jv.com Signed-off-by: Jens Lorenz jlorenz@de.adit-jv.com Signed-off-by: Jiada Wang jiada_wang@mentor.com --- sound/usb/pcm.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-)
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 44d178e..a522c9a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -806,17 +806,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) if (ret < 0) goto unlock;
- iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface); - alts = &iface->altsetting[subs->cur_audiofmt->altset_idx]; - ret = snd_usb_init_sample_rate(subs->stream->chip, - subs->cur_audiofmt->iface, - alts, - subs->cur_audiofmt, - subs->cur_rate); - if (ret < 0) - goto unlock; - if (subs->need_setup_ep) { + + iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface); + alts = &iface->altsetting[subs->cur_audiofmt->altset_idx]; + ret = snd_usb_init_sample_rate(subs->stream->chip, + subs->cur_audiofmt->iface, + alts, + subs->cur_audiofmt, + subs->cur_rate); + if (ret < 0) + goto unlock; + ret = configure_endpoint(subs); if (ret < 0) goto unlock;
On Tue, 06 Dec 2016 06:46:13 +0100, Jiada Wang wrote:
This patch set contains the following patches
Andreas Pape (1): ALSA: usb-audio: more tolerant packetsize
Daniel Girnus (1): ALSA: usb-audio: avoid setting of sample rate multiple times on bus
Applied both patches, thanks.
Takashi
participants (3)
-
Clemens Ladisch -
Jiada Wang -
Takashi Iwai