- Patch - mailman.alsa-project.org

test 12/24
by Jaroslav Kysela 11 Dec '24

11 Dec '24

test 12/24, please, ignore -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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test2
by Jaroslav Kysela 11 May '23

11 May '23

test2, please, ignore -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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test
by Jaroslav Kysela 11 May '23

11 May '23

ignore please -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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Re: [ALSA patch] [PATCH v2] ASoC: Intel: catpt: select WANT_DEV_COREDUMP
by Rojewski, Cezary 17 Nov '20

17 Nov '20

On 2020-11-17 3:52 PM, Gustaw Lewandowski wrote: > From: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> > > Select WANT_DEV_COREDUMP for catpt driver. > > Signed-off-by: Piotr Maziarz <piotrx.maziarz(a)linux.intel.com> > Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski(a)linux.intel.com> > Thanks for this update, Gustaw. Acked-by: Cezary Rojewski <cezary.rojewski(a)intel.com> > -- > Changes in v2: > - change should be added to catpt only > --- > sound/soc/intel/Kconfig | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig > index a5b446d5af19..52ae7c981f56 100644 > --- a/sound/soc/intel/Kconfig > +++ b/sound/soc/intel/Kconfig > @@ -24,6 +24,7 @@ config SND_SOC_INTEL_CATPT > depends on DMADEVICES && SND_DMA_SGBUF > select DW_DMAC_CORE > select SND_SOC_ACPI_INTEL_MATCH > + select WANT_DEV_COREDUMP > help > Enable support for Intel(R) Haswell and Broadwell platforms > with I2S codec present. This is a recommended option. >

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Re: [ALSA patch] [PATCH v3 1/1] ALSA: hda: Refactor calculating SDnFMT according to specification
by Takashi Iwai 22 Sep '20

22 Sep '20

On Tue, 22 Sep 2020 17:33:32 +0200, Pawel Harlozinski wrote: > > Fix setting SDnFMT based on :wq > High Definition Audio Specification Rev. 1.0a page 48. > > Bits per Sample (BITS): > 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. > 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. > 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 101-111 = Reserved > > Set SDnFMT depending on which format was given. > Henceforth split cases for formats 20, 24, 32 bits, > but leave constraints to maxbps. > > Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> As I repeatedly wrote, the bits 20 and 24 cases never hit practically, hence this "refactoring" doesn't give any change in reality. Your commit misses that point. That said, this can be taken as a complementary for the theoretical case, but the situation has to be clarified in the commit log, and the subject should be adjusted as well. thanks, Takashi > > --- > > v3: drop gerrit Change-Id > v2: leave constraints to maxbps > > sound/hda/hdac_device.c | 7 +++++++ > 1 file changed, 7 insertions(+) > > diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c > index 3e9e9ac804f6..ccc47a10ba63 100644 > --- a/sound/hda/hdac_device.c > +++ b/sound/hda/hdac_device.c > @@ -764,7 +764,14 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, > val |= AC_FMT_BITS_16; > break; > case 20: > + val |= AC_FMT_BITS_20; > + break; > case 24: > + if (maxbps >= 24) > + val |= AC_FMT_BITS_24; > + else > + val |= AC_FMT_BITS_20; > + break; > case 32: > if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) > val |= AC_FMT_BITS_32; > -- > 2.17.1 >

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Re: [ALSA patch] [PATCH v2 1/1] ALSA: hda: Refactor calculating SDnFMT according to specification
by Takashi Iwai 21 Sep '20

21 Sep '20

On Mon, 21 Sep 2020 11:42:00 +0200, Pawel Harlozinski wrote: > > Fix setting SDnFMT based on High Definition Audio Specification Rev. 1.0a page 48. > > Bits per Sample (BITS): > 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. > 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. > 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 101-111 = Reserved > > Set SDnFMT depending on which format was given. > Henceforth split cases for formats 20, 24, 32 bits, > but leave constraints to maxbps. As mentioned in the previous thread, practically seen, case 20 and 24 are never met, since HD-audio doesn't accept the SNDRV_PCM_FORMAT_20* or _24* due to the 32bit container. So this change is superfluous. I can take the change, but it has to be clarified that it won't affect the behavior on the real hardware but pure theoretical. > Change-Id: I97771b16da14e85b7f35372f5dfc87bb13bb5ce0 Please drop change-id. It's not useful for others. thanks, Takashi > Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> > --- > sound/hda/hdac_device.c | 7 +++++++ > 1 file changed, 7 insertions(+) > > diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c > index 3e9e9ac804f6..ccc47a10ba63 100644 > --- a/sound/hda/hdac_device.c > +++ b/sound/hda/hdac_device.c > @@ -764,7 +764,14 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, > val |= AC_FMT_BITS_16; > break; > case 20: > + val |= AC_FMT_BITS_20; > + break; > case 24: > + if (maxbps >= 24) > + val |= AC_FMT_BITS_24; > + else > + val |= AC_FMT_BITS_20; > + break; > case 32: > if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) > val |= AC_FMT_BITS_32; > -- > 2.17.1 >

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Re: [ALSA patch] [PATCH v2 1/1] ALSA: hda: Refactor calculating SDnFMT according to specification
by Rojewski, Cezary 21 Sep '20

21 Sep '20

On 2020-09-21 11:42 AM, Pawel Harlozinski wrote: > Fix setting SDnFMT based on High Definition Audio Specification Rev. 1.0a page 48. > > Bits per Sample (BITS): > 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. > 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. > 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. > 101-111 = Reserved > > Set SDnFMT depending on which format was given. > Henceforth split cases for formats 20, 24, 32 bits, > but leave constraints to maxbps. > > Change-Id: I97771b16da14e85b7f35372f5dfc87bb13bb5ce0 Hello, checkpatch script should have notified you about need for removing Change-Id. Please remove for the next version. Has there been any explanation for why v2 is sent and what changes have been made between v1 and v2? Czarek > Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> > --- > sound/hda/hdac_device.c | 7 +++++++ > 1 file changed, 7 insertions(+) > > diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c > index 3e9e9ac804f6..ccc47a10ba63 100644 > --- a/sound/hda/hdac_device.c > +++ b/sound/hda/hdac_device.c > @@ -764,7 +764,14 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, > val |= AC_FMT_BITS_16; > break; > case 20: > + val |= AC_FMT_BITS_20; > + break; > case 24: > + if (maxbps >= 24) > + val |= AC_FMT_BITS_24; > + else > + val |= AC_FMT_BITS_20; > + break; > case 32: > if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) > val |= AC_FMT_BITS_32; >

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Re: [ALSA patch] [PATCH] ALSA: hda: Refactor calculating SDnFMT according to specification
by Takashi Iwai 02 Sep '20

02 Sep '20

On Mon, 24 Aug 2020 14:16:26 +0200, Kai Vehmanen wrote: > > Hey, > > On Mon, 24 Aug 2020, Pawel Harlozinski wrote: > > > Set SDnFMT depending on which format was given, as maxbps only describes container size. > > hmm, I'm not entirely sure that is correct. Usage may be a bit varied, but > most places in existing code, "maxbps" is treated as number of significant > bits, not the container size. E.g. in hdac_hda.c: > > » if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > » » maxbps = dai->driver->playback.sig_bits; > » else > » » maxbps = dai->driver->capture.sig_bits; > > It would seem "maxbps" is a bit superfluous given the same information can > be relayed in "format" as well. But currently it's still used. E.g. if you > look at snd_hdac_query_supported_pcm(), if codec reports 24bit support, > format is always set to SNDRV_PCM_FMTBIT_S32_LE, even if only 24bit are > valid. So snd_pcm_format_width() will not return the expected significant > bits info, but you have to use "maxbps". So original code seems correct > (or at least you'd need to update both places). Hm, we need to check the call pattern, then. The maxbps passed to this function was supposed to be the value obtained from snd_hdac_query_supported_pcm(), i.e. the codec capability. But, basically this patch wouldn't change any practical behavior. In the current code, snd_pcm_format_width() can be never 20 or 24, because the 24 and 24bit supports are also with SNDRV_PCM_FMT_S32_LE. That is, the cases 20 and 24 there are superfluous from the beginning (although the checks of maxbps are still needed). Instead, what we could improve is: - Set up the proper msbits hw_constraint to reflect the maxbps value - Choose the right AC_FMT_BITS_* depending on the hw_params msbits We may change the query function not to return a single maxbps value but rather storing the raw PCM parameter value (AC_SUPPCM_*), and pass it at re-encoding the format value, too, if we want to make things perfect. thanks, Takashi

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[PATCH] ALSA: hda: Refactor calculating SDnFMT according to specification
by Pawel Harlozinski 24 Aug '20

24 Aug '20

Fix setting SDnFMT based on High Definition Audio Specification Rev. 1.0a page 48. Bits per Sample (BITS): 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 101-111 = Reserved Set SDnFMT depending on which format was given, as maxbps only describes container size. Henceforth split cases for formats 20, 24, 32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so as previously it will end up with mask for 32 bits. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8af..2ccf79866f99 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -762,14 +762,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

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[PATCH] ALSA: usb-audio: ignore broken processing/extension unit
by tom.ty89＠gmail.com 17 Aug '20

17 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). Signed-off-by: Tom Yan <tom.ty89(a)gmail.com> --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c3..e0b7174c1043 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.28.0

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[PATCH] ALSA: usb-audio: ignore broken processing/extension unit
by tom.ty89＠gmail.com 16 Aug '20

16 Aug '20

From: Tom Yan <tom.ty89(a)gmail.com> Some devices have broken extension unit where getting current value doesn't work. Attempt that once when creating mixer control for it. If it fails, just ignore it, so that it won't cripple the device entirely (and/or make the error floods). --- sound/usb/mixer.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index eab0fd4fd7c3..e0b7174c1043 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2367,7 +2367,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2470,6 +2470,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); -- 2.28.0

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Re: [ALSA patch] [PATCH] pcm: dmix: fix access to sum-buffer in non-interleaved mixing mode
by Takashi Iwai 13 Aug '20

13 Aug '20

On Wed, 12 Aug 2020 08:18:25 +0200, Vijay Palaniswamy (RBEI/ECF3) wrote: > > Hi All, > > When dmix uses non-interleaved mixing mode the offset and step width to > sum_buffer was calculated by using the dmix channels instead of the slave > channels. This leads to audio distortions due to frame corruption. > > Issue happens while doing channel wise mixing of src and destination buffers. > > Attached the patch to fix the offset and step width calculation based on slave > channels for sum buffer. So mixed data will be saved in respective channels of > sum buffer. > > Steps to reproduce the issue: > > With below configuratio, Do aplay on both device in parallel for audio > distortion > > pcm.dmix_2_channels { > > type dmix > > ipc_key 5678293 > > ipc_perm 0660 > > ipc_gid audio > > bindings [0 1] > > slave { > > pcm "hardware" > > channels 2 > > periods 4 > > period_time 40000 > > } > > } > > pcm.dmix_1_channels { > > type dmix > > ipc_key 5678293 > > ipc_perm 0660 > > ipc_gid audio > > bindings [0] > > slave { > > pcm "hardware" > > channels 1 > > periods 4 > > period_time 40000 > > } > > } > > pcm.hardware { > > type hw > > card 0 > > channels 2 > > rate 48000 > > format S16_LE > > } > > Best regards > > Palaniswamy Vijay Applied now. Thanks. Takashi

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Re: [ALSA patch] [PATCH 1/1] ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Xtreme
by Takashi Iwai 18 May '20

18 May '20

On Mon, 18 May 2020 07:38:44 +0200, Christian Lachner wrote: > > The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec > requires a similar workaround for Clevo laptops to enforce the > DAC/mixer connection path. Set up a quirk entry for that. > > BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275 > Signed-off-by: Christian Lachner <gladiac(a)gmail.com> Applied now. thanks, Takashi

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Re: [ALSA patch] [PATCH alsa-lib] conf: USB-Audio: Fix S/PDIF output of ASUS Xonar AE
by Jaroslav Kysela 17 Apr '20

17 Apr '20

Dne 26. 03. 20 v 20:46 Ivan Mironov napsal(a): > ASUS Xonar AE is a PCI-Express card containing USB controller: > > USB controller [0c03]: ASMedia Technology Inc. ASM1042A USB 3.0 Host Controller [1b21:1142] > > and the actual USB sound card: > > ID 0b05:180f ASUSTek Computer, Inc. XONAR SOUND CARD > > As other Xonar USB sound cards, it uses second device for digital > output. > > Signed-off-by: Ivan Mironov <mironov.ivan(a)gmail.com> > > diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf > index 89811086db1d..3f4c421b7e2c 100644 > --- a/src/conf/cards/USB-Audio.conf > +++ b/src/conf/cards/USB-Audio.conf > @@ -40,6 +40,7 @@ USB-Audio.pcm.iec958_device { > "USB Sound Blaster HD" 1 > "Xonar U7" 1 > "ASUS XONAR U5" 1 > + "XONAR SOUND CARD" 1 > > # The below don't have digital in/out, so prevent them from being opened. > "Andrea PureAudio USB-SA Headset" 999 > Merged. Thank you. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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Re: [ALSA patch] [alsa-devel] [PATCH] do not set close-on-exec flag on descriptor if it was already set
by Jaroslav Kysela 11 Feb '20

11 Feb '20

Dne 11. 02. 20 v 11:22 Rolf Eike Beer napsal(a): > There is no need to set this again if O_CLOEXEC is supported. Thanks. Applied. Jroslav > > Signed-off-by: Rolf Eike Beer <eb(a)emlix.com> > --- > include/local.h | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/include/local.h b/include/local.h > index ea0ec32d..ed6ba936 100644 > --- a/include/local.h > +++ b/include/local.h > @@ -320,8 +320,10 @@ static inline int snd_open_device(const char *filename, int fmode) > fd = rsm_open_device(filename, fmode); > } > #endif > +#ifndef O_CLOEXEC > if (fd >= 0) > fcntl(fd, F_SETFD, FD_CLOEXEC); > +#endif > return fd; > } > > -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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Re: [ALSA patch] [alsa-devel] [PATCH - 1/1] pcm: direct: Round down of slave_hw_ptr when buffer size is two period size
by Takashi Iwai 06 Feb '20

06 Feb '20

On Thu, 06 Feb 2020 09:14:42 +0100, Channaiah Vanitha (RBEI/ECF3) wrote: > > Hello Takashi-san, > > This patch is regarding the fix for rounding down/up of slave pointers. > For buffer_size >= 2*period_size, round down of slave pointers and > for buffer_size < 2*period_size, round up of slave pointers will avoid xruns. > which otherwise causes snd_pcm_wait() to block for more than expected snd_pcm_period_elapsed() which leads to xruns. > > We had similar discussion for same issue in below link: > https://mailman.alsa-project.org/pipermail/alsa-devel/2019-June/151169.html Yes, and I still don't get why this must be required. Doesn't this imply that you drop the samples instead? Takashi > > Regards, > Vanitha > > > -----Original Message----- > From: Channaiah Vanitha (RBEI/ECF3) <Vanitha.Channaiah(a)in.bosch.com> > Sent: Thursday, February 6, 2020 1:43 PM > To: Channaiah Vanitha (RBEI/ECF3) <Vanitha.Channaiah(a)in.bosch.com>; patch(a)alsa-project.org > Cc: alsa-devel(a)alsa-project.org; Wischer Timo (ADITG/ESS) <twischer(a)de.adit-jv.com> > Subject: [PATCH - 1/1] pcm: direct: Round down of slave_hw_ptr when buffer size is two period size > > From: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> > > For buffer size equal to two period size, the start position of slave_hw_ptr is rounded down in order to avoid xruns > > For e.g.: > Considering below parameters and its values: > Period size = 96 (0x60) > Buffer size = 192 (0xC0) > Timer ticks = 1 > avail_min = 0x60 > slave_hw_ptr = unaligned value during dmix_start() > > Issue: > - Initial, app_ptr = hw_ptr = 0 > - Application fills buffer size of data. so app_ptr = 0xC0, hw_ptr = 0 > - During dmix_start(), current slave_hw_ptr is not rounded down. current slave_hw_ptr would be 0x66 > - slave_hw_ptr is keep on updating at the hardware 0x67, 0x68, 0x69 > - The diff calculation between old_slave_hw_ptr(0x66) and new_slave_hw_ptr(0x69) > results in avail = 0x6 > - After 1 snd_pcm_period_elapsed(), slave_hw_ptr = 0xC0 > - The result of avail = 0x5A which is less than avail_min(0x60) > - Here, xruns will be observed > > Fix: > - Initial, app_ptr = hw_ptr = 0 > - Application fills buffer size of data. so app_ptr = 0xC0, hw_ptr = 0 > - Round down of slave_hw_ptr during dmix_start() leads to below calculation: > - During dmix_start(), slave_hw_ptr rounded to 0x60 (old slave_hw_ptr) > - The diff calculation between old_slave_hw_ptr(0x60) and new_slave_hw_ptr(0x69) > results in avail = 0x9 > - After 1 snd_pcm_period_elapsed(), slave_hw_ptr = 0xC0 > - The result of avail = 0x60 which is avail_min(0x60) > - Here, xruns can be avoided > > Signed-off-by: Vanitha Channaiah <vanitha.channaiah(a)in.bosch.com> > > diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 54d9900..a201fa3 100644 > --- a/src/pcm/pcm_direct.c > +++ b/src/pcm/pcm_direct.c > @@ -2043,10 +2043,14 @@ int snd_pcm_direct_parse_open_conf(snd_config_t *root, snd_config_t *conf, > > void snd_pcm_direct_reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { > + /* > + * For buffer size equal to two period size, slave_hw_ptr is rounded down > + * to avoid xruns > + */ > > if (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_ROUNDUP || > (dmix->hw_ptr_alignment == SND_PCM_HW_PTR_ALIGNMENT_AUTO && > - pcm->buffer_size <= pcm->period_size * 2)) > + pcm->buffer_size < pcm->period_size * 2)) > dmix->slave_appl_ptr = > ((dmix->slave_appl_ptr + dmix->slave_period_size - 1) / > dmix->slave_period_size) * dmix->slave_period_size; > -- > 2.7.4 > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > https://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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Re: [ALSA patch] [alsa-devel] [PATCH] ASoC: rt274: Disable jack report IRQ with disabling jack
by Harlozinski, Pawel 13 Nov '19

13 Nov '19

On 11/12/2019 6:10 PM, Mark Brown wrote: > On Tue, Nov 12, 2019 at 02:02:37PM +0100, Pawel Harlozinski wrote: > >> /* Disable jack detection */ >> regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL, >> RT274_IRQ_EN, RT274_IRQ_DIS); >> - >> + disable_irq(rt274->i2c->irq); >> return 0; > Shouldn't the register update above be suppressing interrupts? For rt274 disable_irq is also needed, otherwise we're getting flood of irq's in case of not loaded machine board. > disable_irq() is a bit of a hammer and interferes with things like > possible share use. This irq should be handled in codec code anyway - control of jack detect events from non-codec code is done with set_jack. Similar solutions for jack report irq enable/disable flow are implemented in rt5640 and rt5651. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > https://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[PATCH] ASoC: Jack: Fix NULL pointer dereference in snd_soc_jack_report
by Pawel Harlozinski 12 Nov '19

12 Nov '19

Check for existance of jack before tracing. NULL pointer dereference has been reported by KASAN while unloading machine driver (snd_soc_cnl_rt274). Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/soc-jack.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index a71d2340eb05..b5748dcd490f 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -82,10 +82,9 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) unsigned int sync = 0; int enable; - trace_snd_soc_jack_report(jack, mask, status); - if (!jack) return; + trace_snd_soc_jack_report(jack, mask, status); dapm = &jack->card->dapm; -- 2.17.1

1 1

[PATCH v3] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 12 Sep '19

12 Sep '19

When DSP boots with timeout error try to reload firmware to keep best user expeirience. Change-Id: Ic7377da687e7441a487c443f9de1ebae5967afa2 Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 92a82e6b5fe6..fdf4c198445e 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -446,14 +446,19 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); + /* FIXME this FW recovery applies only for BXT */ if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x. Reloading FW\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + + ret = bxt_sst_init_fw(skl->dev, skl); + if (ret < 0) { + dev_err(ctx->dev, "Reload fw failed: %d. Failed to set core0 to D0 state\n", + ret); + goto err; + } } } @@ -478,7 +483,6 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) if (core_id == SKL_DSP_CORE0_ID) core_mask |= SKL_DSP_CORE_MASK(1); skl_dsp_disable_core(ctx, core_mask); - return ret; } -- 2.17.1

1 0

[RFC PATCH 1/3] ALSA:hda: Simplify and clear calculating SDxFMT
by Pawel Harlozinski 06 Sep '19

06 Sep '19

Set SDxFMT based only on given format, due to maxbps not always being set. Split cases for formats 20,24,32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so it will end up with same mask. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..add758e11b85 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -759,14 +759,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

3 7

Re: [ALSA patch] [alsa-devel] [RFC PATCH 2/3] ALSA:core: Add rate 24kHz
by Takashi Iwai 05 Sep '19

05 Sep '19

On Thu, 05 Sep 2019 16:00:51 +0200, Pierre-Louis Bossart wrote: > > > >>>> Adds SNDRV_PCM_RATE_24000 at the bottom to keep backward compability > >>>> with alsa library. > >>>> > >>>> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> > >>> > >>> No. Such a fancy rate has to be handled inside the driver locally > >>> instead of adding to the common rate. > >> > >> It's not that crazy, this is supported in the HDaudio spec: > >> > >> Sample Base Rate Divisor (DIV): > >> 000 = Divide by 1 (48 kHz, 44.1 kHz) > >> 001 = Divide by 2 (24 kHz, 22.05 kHz) > >> > >> I am not sure why 22.05 made the cut and not 24 kHz, they are both > >> derived from common clocks with the same divider... Same for 11.025 > >> and 12... > > > > I'm not against supporting it. It'd be fine if it were the changes > > that are applied only to HD-audio driver side. What I'm against is to > > change the ALSA PCM core. It's not necessarily done there at all. > > Humm, out of curiosity what is the issue here? Would this addition > break anything? I don't personally care too much but I've never quite > understood why the ALSA core only defined a subset of 'common' rates. It's simply a policy that we don't add a thing just because one driver wants for some reason possibly no one would actually use. In general such a core stuff is changed only when it has to be and inevitably necessary, and/or it'd be benefit for all the rest and majority users. thanks, Takashi

1 0

Re: [ALSA patch] [RFC PATCH 2/3] ALSA:core: Add rate 24kHz
by Takashi Iwai 05 Sep '19

05 Sep '19

On Thu, 05 Sep 2019 15:11:43 +0200, Pierre-Louis Bossart wrote: > > On 9/5/19 12:48 AM, Takashi Iwai wrote: > > On Thu, 05 Sep 2019 07:33:00 +0200, > > Pawel Harlozinski wrote: > >> > >> Adds SNDRV_PCM_RATE_24000 at the bottom to keep backward compability > >> with alsa library. > >> > >> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> > > > > No. Such a fancy rate has to be handled inside the driver locally > > instead of adding to the common rate. > > It's not that crazy, this is supported in the HDaudio spec: > > Sample Base Rate Divisor (DIV): > 000 = Divide by 1 (48 kHz, 44.1 kHz) > 001 = Divide by 2 (24 kHz, 22.05 kHz) > > I am not sure why 22.05 made the cut and not 24 kHz, they are both > derived from common clocks with the same divider... Same for 11.025 > and 12... I'm not against supporting it. It'd be fine if it were the changes that are applied only to HD-audio driver side. What I'm against is to change the ALSA PCM core. It's not necessarily done there at all. thanks, Takashi

1 0

[PATCH] ALSA: pcm: fix lost wakeup event scenarios in snd_pcm_drain
by Adam Miartus 29 Jul '19

29 Jul '19

From: Yuki Tsunashima <ytsunashima(a)jp.adit-jv.com> lost wakeup can occur after enabling irq, therefore put task into interruptible before enabling interrupts, without this change, task can be put to sleep and snd_pcm_drain will delay Signed-off-by: Yuki Tsunashima <ytsunashima(a)jp.adit-jv.com> Signed-off-by: Suresh Udipi <sudipi(a)jp.adit-jv.com> [ported from 4.9] Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 860543a..fe333c8 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1874,6 +1874,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (!to_check) break; /* all drained */ init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); add_wait_queue(&to_check->sleep, &wait); snd_pcm_stream_unlock_irq(substream); if (runtime->no_period_wakeup) @@ -1886,7 +1887,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } tout = msecs_to_jiffies(tout * 1000); } - tout = schedule_timeout_interruptible(tout); + tout = schedule_timeout(tout); snd_pcm_stream_lock_irq(substream); group = snd_pcm_stream_group_ref(substream); -- 2.7.4

1 0

[PATCH 0/2 v2] Open Alsa dshare without channel binding
by Adam Miartus 24 Jul '19

24 Jul '19

The general idea of this patch is to be able to open the device without defined bindings. As a result, it allows starting the audio clock of underlying device without blocking a channel. This might be useful for creating a dummy dsnoop pcm for hardware that requires running clock early at system start. This clock is then present even without application streaming audio data. Patch n.1 is intended to check for null pointer of pcm callbacks, it is not intended to alter any current functionality, properly defined plugins should not suffer from this change Patch n.2 is not intended to alter any current functionality, only extend it V2 of the patch, changes from V1: - instead of creating new "pcm_unsupported.c" with ops and fast_ops callbacks returning -ENOSYS, callback == NULL is checked directly in core implementation and error is returned when pcm has no callback defined Adam Miartus (2): alsa: pcm: return -ENOSYS when ops or fast_ops callback is NULL alsa: dshare: allow missing bindings src/pcm/pcm.c | 134 +++++++++++++++++++++++++++++++++++++++++---------- src/pcm/pcm_dshare.c | 29 ++++++----- src/pcm/pcm_local.h | 20 ++++++++ src/pcm/pcm_mmap.c | 15 ++++-- src/pcm/pcm_params.c | 10 +++- 5 files changed, 166 insertions(+), 42 deletions(-) -- 2.7.4

1 2

[PATCH v2] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 08 Jul '19

08 Jul '19

When DSP boots with timeout error try to reload firmware to keep best user expeirience. Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 440bca7afbf1..dd0260557530 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -455,13 +455,19 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + dev_warn(ctx->dev,"Reloading FW\n"); + + ret = bxt_sst_init_fw(skl->dev, skl); + if (ret < 0) { + dev_err(ctx->dev, "Reload fw failed: %d\n", ret); + dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); + ret = -EIO; + goto err; + } } } @@ -486,7 +492,6 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) if (core_id == SKL_DSP_CORE0_ID) core_mask |= SKL_DSP_CORE_MASK(1); skl_dsp_disable_core(ctx, core_mask); - return ret; } -- 2.17.1

1 0

[PATCH 0/2] expand dshare to allow audio clock when not streaming data
by Adam Miartus 08 Jul '19

08 Jul '19

The general idea of these changes is to be able to open the device without defined bindings. This would start the audio clock without blocking a channel. This might be useful for hardware that requires running clock early at system start and to be available even without application streaming audio data. Andreas Pape (2): alsa: pcm: add unsupported OPS alsa: dshare: allow missing bindings src/pcm/Makefile.am | 4 +- src/pcm/pcm_dshare.c | 72 +++++++++++-- src/pcm/pcm_unsupported.c | 265 ++++++++++++++++++++++++++++++++++++++++++++++ src/pcm/pcm_unsupported.h | 167 +++++++++++++++++++++++++++++ 4 files changed, 495 insertions(+), 13 deletions(-) create mode 100644 src/pcm/pcm_unsupported.c create mode 100644 src/pcm/pcm_unsupported.h -- 2.7.4

1 2

[PATCH 2/3 v2] pcm_file: improve error checking in write_wav_header function
by Adam Miartus 05 Jul '19

05 Jul '19

previously errno would be returned even for cases where it may have not been populated, for example one of the write functions failing, or writing only partial buffer, now progress through write operations separately and report errno when appropriate Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> --- src/pcm/pcm_file.c | 38 ++++++++++++++++++++++++++++++-------- 1 file changed, 30 insertions(+), 8 deletions(-) diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 03c1640..da7f087 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -327,6 +327,8 @@ static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) static int write_wav_header(snd_pcm_t *pcm) { snd_pcm_file_t *file = pcm->private_data; + ssize_t res; + static const char header[] = { 'R', 'I', 'F', 'F', 0x24, 0, 0, 0, @@ -341,15 +343,35 @@ static int write_wav_header(snd_pcm_t *pcm) setup_wav_header(pcm, &file->wav_header); - if (write(file->fd, header, sizeof(header)) != sizeof(header) || - write(file->fd, &file->wav_header, sizeof(file->wav_header)) != - sizeof(file->wav_header) || - write(file->fd, header2, sizeof(header2)) != sizeof(header2)) { - int err = errno; - SYSERR("%s write header failed, file data may be corrupt", file->fname); - return -err; - } + res = write(file->fd, header, sizeof(header)); + if (res != sizeof(header)) + goto write_error; + + res = write(file->fd, &file->wav_header, sizeof(file->wav_header)); + if (res != sizeof(file->wav_header)) + goto write_error; + + res = write(file->fd, header2, sizeof(header2)); + if (res != sizeof(header2)) + goto write_error; + return 0; + +write_error: + /* + * print real errno if available and return EIO, reason for this is + * to block possible EPIPE in case file->fd is a pipe. EPIPE from + * file->fd conflicts with EPIPE from playback stream which should + * be used to signal XRUN on playback device + */ + if (res < 0) + SYSERR("%s write header failed, file data may be corrupt", file->fname); + else + SNDERR("%s write header incomplete, file data may be corrupt", file->fname); + + memset(&file->wav_header, 0, sizeof(struct wav_fmt)); + + return -EIO; } /* fix up the length fields in WAV header */ -- 2.7.4

1 0

[PATCH 0/3] pcm_file report EIO when fail to write to debug file
by Adam Miartus 01 Jul '19

01 Jul '19

Previously, I introduced a patch that may return EPIPE error when pcm_file plugin fails to write to output file for some reason. The purpose of reporting the error was to evade assert in code to allow API user error handling and recovery, or more clarity in debugging. Failing to write to a file usually means bad file descriptor, running out of free memory, insufficient file permissions or in case file is a pipe there may be other reasons preventing the write operation. All these are not easily recoverable by restaring playback as EPIPE error which is used to signal XRUN might suggest. Therefore, use EIO to signal a different kind of error. write_wav_header function is refactored a bit for more detailed error handling in case of failed write operation. Adam Miartus (3): pcm_file: use EIO instead of EPIPE when failing to write output file pcm_file: improve error checking in write_wav_header function pcm_file: in case of failed write clear file buffer variables src/pcm/pcm_file.c | 54 +++++++++++++++++++++++++++++++++++++++++------------- 1 file changed, 41 insertions(+), 13 deletions(-) -- 2.7.4

1 3

GIT: Regarding the issue we are facing in the commit 37728639ae05de702825d96bd1d42e24ae772248
by vanitha.channaiah＠in.bosch.com 01 Jul '19

01 Jul '19

This mail is regarding the fix at below: commit 37728639ae05de702825d96bd1d42e24ae772248 Author: Jaroslav Kysela <perex(a)perex.cz> Date: Sat Feb 7 15:01:31 2004 +00 The commit has following changes : ******************************* code ***************************************** static int snd_pcm_rate_hw_refine_cchange(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_hw_params_t *sparams) { + buffer_size = snd_pcm_hw_param_get_interval(params, SND_PCM_HW_PARAM_BUFFER_SIZE); + /* + * this condition probably needs more work: + * in case when the buffer_size is known and we are looking + * for best period_size, we should prefer situation when + * (buffer_size / period_size) * period_size == buffer_size + */ + if (snd_interval_single(buffer_size) && buffer_size->integer) { + snd_interval_t *period_size; + period_size = (snd_interval_t *)snd_pcm_hw_param_get_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE); + if (!snd_interval_checkempty(period_size) && + period_size->openmin && period_size->openmax && + period_size->min + 1 == period_size->max) { + if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { + snd_interval_set_value(period_size, period_size->min); + } else if ((buffer_size->max / period_size->max) * period_size->max == buffer_size->max) { + snd_interval_set_value(period_size, period_size->max); + } + } + } } ******************************* code **************************************** Update rate depending on period_size and period_time. Issue found in Hardware : a. RCAR salvator-xs which supports 2 channel. b. IMX which supports 8 channel. 1. Usecase which is PASSED without the mentioned commit and FAILED with the commit. For the usecase with below command : rate = 11025 channel = 6 $aplay -Dentertainment_main -r11025 -c6 -fS16_LE /dev/urandom a. With the commit: In the calculation of RATE at Rule 7: RATE=11025 dependent parameters are: PERIOD_SIZE=88 PERIOD_TIME=8000 parameters min max open_min open_max interval PERIOD_SIZE 88 88 0 0 0 PERIO_TIME 8000 8000 0 0 0 RATE 11025 11000 0 0 1 RATE values are calculated in snd_interval_refine() Return value: INVALID as rate_min > rate_max This is because, dependent parameter “period_size” is rounded to 88 in “rate plugin” in snd_pcm_rate_hw_refine_cchange() Since, the below condition gets satisfied(buffer_size = 352 aligned to period_size = 88), period_size gets rounded to 88. if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { This commit changes causing the issue to get rate_min > rate_max. The flow of code execution is as follows: - snd_pcm_hw_refine_slave() - Enters snd_pcm_rate_hw_refine_cchange(), rounding of period_size to 88. - snd_pcm_rate_hw_refine_cchange() exit. - snd_pcm_hw_refine_soft() is called, here exists params->rmask to evaluate for SAMPLE_BITS, FRAME_BITS, PERIOD_BYTES, BUFFER_BYTES - execution of RULES calculation. At RULE 7 rate calculation INVALID error is observed. b. Without the commit: In the case if period_size doesn’t get rounded off to 88, the usecase would get PASSED. Rate calculation goes fine with period_size open interval (88, 89) parameters min max open_min open_max interval PERIOD_SIZE 88 89 0 0 0 PERIOD_TIME 8000 8000 0 0 0 RATE 11025 11025 0 0 1 Here, there is no issue of rate_min > rate_max. **************************************************************************************************************************************************************** Now, I just have one scenario where with/without commit, rate calculation goes fine. 2. Usecase which is PASSED with and without above commit. For the usecase with below command : rate =11025 channel=2 $aplay -Dentertainment_main -r11025 -c2 -fS16_LE /dev/urandom a. Including the above commit code: Below is the calculation that goes: The flow of code execution is as follows: - snd_pcm_hw_refine_slave() - Enters snd_pcm_rate_hw_refine_cchange() , rounding of period_size to 88. - snd_pcm_rate_hw_refine_cchange() exit. - snd_pcm_hw_refine_soft() is called, here params->rmask is 0. - RULES calculation doesnt occur. b. Without the commit: In the case if period_size doesn’t get rounded off to 88. Rate calculation goes fine with period_size open interval (88, 89) Is there any dependency for the commit ? If yes, can you please suggest the corner case which fails without the commit ? Can we revert the changes ?

1 0

[PATCH - use build platform compiler while cross compilation 1/1] use build compilers in cross compilation setup
by Ding Xiang Fei 29 Jun '19

29 Jun '19

This patch is to make the build script always use the build platform compilers, especially during the cross compilation process, since the Makefile will pick up the cross compilers instead of the compilers on the build machine. Signed-off-by: Ding Xiang Fei <dingxiangfei2009(a)gmail.com> diff --git a/compile b/compile new file mode 120000 index 0000000..349ac66 --- /dev/null +++ b/compile @@ -0,0 +1 @@ +/nix/store/hwcmvzv53jzws5vnaga16l5dmqvslwkz-automake-1.16.1/share/automake-1.16/compile \ No newline at end of file diff --git a/configure.ac b/configure.ac index 368854e..ee0e242 100644 --- a/configure.ac +++ b/configure.ac @@ -1,6 +1,8 @@ AC_PREREQ(2.59) AC_INIT(alsa-firmware, 1.0.29) +AC_CONFIG_MACRO_DIR([m4]) AC_PROG_CC +AC_PROG_CC_FOR_BUILD AC_PROG_INSTALL AC_PROG_LN_S AC_HEADER_STDC diff --git a/echoaudio/Makefile.am b/echoaudio/Makefile.am index 815742f..5ef6086 100644 --- a/echoaudio/Makefile.am +++ b/echoaudio/Makefile.am @@ -74,33 +74,42 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer DSP/LoaderDSP.c loader_dsp.fw - ./fw_writer DSP/Darla20DSP.c darla20_dsp.fw - ./fw_writer DSP/Gina20DSP.c gina20_dsp.fw - ./fw_writer DSP/Layla20DSP.c layla20_dsp.fw - ./fw_writer ASIC/LaylaASIC.c layla20_asic.fw - ./fw_writer DSP/Darla24DSP.c darla24_dsp.fw - ./fw_writer DSP/Gina24DSP.c gina24_301_dsp.fw - ./fw_writer ASIC/Gina24ASIC.c gina24_301_asic.fw - ./fw_writer DSP/Gina24_361DSP.c gina24_361_dsp.fw - ./fw_writer ASIC/Gina24ASIC_361.c gina24_361_asic.fw - ./fw_writer DSP/Layla24DSP.c layla24_dsp.fw - ./fw_writer ASIC/Layla24_1ASIC.c layla24_1_asic.fw - ./fw_writer ASIC/Layla24_2A_ASIC.c layla24_2A_asic.fw - ./fw_writer ASIC/Layla24_2S_ASIC.c layla24_2S_asic.fw - ./fw_writer DSP/MonaDSP.c mona_301_dsp.fw - ./fw_writer ASIC/Mona1ASIC48.c mona_301_1_asic_48.fw - ./fw_writer ASIC/Mona1ASIC96.c mona_301_1_asic_96.fw - ./fw_writer DSP/Mona361DSP.c mona_361_dsp.fw - ./fw_writer ASIC/Mona1ASIC48_361.c mona_361_1_asic_48.fw - ./fw_writer ASIC/Mona1ASIC96_361.c mona_361_1_asic_96.fw - ./fw_writer ASIC/Mona2ASIC.c mona_2_asic.fw - ./fw_writer DSP/MiaDSP.c mia_dsp.fw - ./fw_writer DSP/Echo3gDSP.c echo3g_dsp.fw - ./fw_writer ASIC/3G_ASIC.c 3g_asic.fw - ./fw_writer DSP/IndigoDSP.c indigo_dsp.fw - ./fw_writer DSP/IndigoIODSP.c indigo_io_dsp.fw - ./fw_writer DSP/IndigoIOxDSP.c indigo_iox_dsp.fw - ./fw_writer DSP/IndigoDJDSP.c indigo_dj_dsp.fw - ./fw_writer DSP/IndigoDJxDSP.c indigo_djx_dsp.fw +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) DSP/LoaderDSP.c loader_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Darla20DSP.c darla20_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Gina20DSP.c gina20_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Layla20DSP.c layla20_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/LaylaASIC.c layla20_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Darla24DSP.c darla24_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Gina24DSP.c gina24_301_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Gina24ASIC.c gina24_301_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Gina24_361DSP.c gina24_361_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Gina24ASIC_361.c gina24_361_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Layla24DSP.c layla24_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Layla24_1ASIC.c layla24_1_asic.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Layla24_2A_ASIC.c layla24_2A_asic.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Layla24_2S_ASIC.c layla24_2S_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/MonaDSP.c mona_301_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC48.c mona_301_1_asic_48.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC96.c mona_301_1_asic_96.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Mona361DSP.c mona_361_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC48_361.c mona_361_1_asic_48.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona1ASIC96_361.c mona_361_1_asic_96.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/Mona2ASIC.c mona_2_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/MiaDSP.c mia_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/Echo3gDSP.c echo3g_dsp.fw + ./fw_writer$(BUILD_EXEEXT) ASIC/3G_ASIC.c 3g_asic.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoDSP.c indigo_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoIODSP.c indigo_io_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoIOxDSP.c indigo_iox_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoDJDSP.c indigo_dj_dsp.fw + ./fw_writer$(BUILD_EXEEXT) DSP/IndigoDJxDSP.c indigo_djx_dsp.fw diff --git a/emu/Makefile.am b/emu/Makefile.am index 3a633c0..e5a3017 100644 --- a/emu/Makefile.am +++ b/emu/Makefile.am @@ -22,5 +22,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/hdsploader/Makefile.am b/hdsploader/Makefile.am index 48802a6..b13245b 100644 --- a/hdsploader/Makefile.am +++ b/hdsploader/Makefile.am @@ -32,5 +32,14 @@ EXTRA_DIST = $(dsp_hex_files:%.bin=%.dat) \ tobin.c CLEANFILES = $(dsp_hex_files) -$(dsp_hex_files): tobin - ./tobin +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(tobin_OBJECTS) : CC=$(CC_FOR_BUILD) +$(tobin_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(tobin_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +tobin$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(dsp_hex_files): tobin$(BUILD_EXEEXT) + ./$< diff --git a/m4/ax_prog_cc_for_build.m4 b/m4/ax_prog_cc_for_build.m4 new file mode 100644 index 0000000..12cb005 --- /dev/null +++ b/m4/ax_prog_cc_for_build.m4 @@ -0,0 +1,125 @@ +# =========================================================================== +# https://www.gnu.org/software/autoconf-archive/ax_prog_cc_for_build.html +# =========================================================================== +# +# SYNOPSIS +# +# AX_PROG_CC_FOR_BUILD +# +# DESCRIPTION +# +# This macro searches for a C compiler that generates native executables, +# that is a C compiler that surely is not a cross-compiler. This can be +# useful if you have to generate source code at compile-time like for +# example GCC does. +# +# The macro sets the CC_FOR_BUILD and CPP_FOR_BUILD macros to anything +# needed to compile or link (CC_FOR_BUILD) and preprocess (CPP_FOR_BUILD). +# The value of these variables can be overridden by the user by specifying +# a compiler with an environment variable (like you do for standard CC). +# +# It also sets BUILD_EXEEXT and BUILD_OBJEXT to the executable and object +# file extensions for the build platform, and GCC_FOR_BUILD to `yes' if +# the compiler we found is GCC. All these variables but GCC_FOR_BUILD are +# substituted in the Makefile. +# +# LICENSE +# +# Copyright (c) 2008 Paolo Bonzini <bonzini(a)gnu.org> +# +# Copying and distribution of this file, with or without modification, are +# permitted in any medium without royalty provided the copyright notice +# and this notice are preserved. This file is offered as-is, without any +# warranty. + +#serial 9 + +AU_ALIAS([AC_PROG_CC_FOR_BUILD], [AX_PROG_CC_FOR_BUILD]) +AC_DEFUN([AX_PROG_CC_FOR_BUILD], [dnl +AC_REQUIRE([AC_PROG_CC])dnl +AC_REQUIRE([AC_PROG_CPP])dnl +AC_REQUIRE([AC_EXEEXT])dnl +AC_REQUIRE([AC_CANONICAL_HOST])dnl + +dnl Use the standard macros, but make them use other variable names +dnl +pushdef([ac_cv_prog_CPP], ac_cv_build_prog_CPP)dnl +pushdef([ac_cv_prog_gcc], ac_cv_build_prog_gcc)dnl +pushdef([ac_cv_prog_cc_works], ac_cv_build_prog_cc_works)dnl +pushdef([ac_cv_prog_cc_cross], ac_cv_build_prog_cc_cross)dnl +pushdef([ac_cv_prog_cc_g], ac_cv_build_prog_cc_g)dnl +pushdef([ac_cv_exeext], ac_cv_build_exeext)dnl +pushdef([ac_cv_objext], ac_cv_build_objext)dnl +pushdef([ac_exeext], ac_build_exeext)dnl +pushdef([ac_objext], ac_build_objext)dnl +pushdef([CC], CC_FOR_BUILD)dnl +pushdef([CPP], CPP_FOR_BUILD)dnl +pushdef([CFLAGS], CFLAGS_FOR_BUILD)dnl +pushdef([CPPFLAGS], CPPFLAGS_FOR_BUILD)dnl +pushdef([LDFLAGS], LDFLAGS_FOR_BUILD)dnl +pushdef([host], build)dnl +pushdef([host_alias], build_alias)dnl +pushdef([host_cpu], build_cpu)dnl +pushdef([host_vendor], build_vendor)dnl +pushdef([host_os], build_os)dnl +pushdef([ac_cv_host], ac_cv_build)dnl +pushdef([ac_cv_host_alias], ac_cv_build_alias)dnl +pushdef([ac_cv_host_cpu], ac_cv_build_cpu)dnl +pushdef([ac_cv_host_vendor], ac_cv_build_vendor)dnl +pushdef([ac_cv_host_os], ac_cv_build_os)dnl +pushdef([ac_cpp], ac_build_cpp)dnl +pushdef([ac_compile], ac_build_compile)dnl +pushdef([ac_link], ac_build_link)dnl + +save_cross_compiling=$cross_compiling +save_ac_tool_prefix=$ac_tool_prefix +cross_compiling=no +ac_tool_prefix= + +AC_PROG_CC +AC_PROG_CPP +AC_EXEEXT + +ac_tool_prefix=$save_ac_tool_prefix +cross_compiling=$save_cross_compiling + +dnl Restore the old definitions +dnl +popdef([ac_link])dnl +popdef([ac_compile])dnl +popdef([ac_cpp])dnl +popdef([ac_cv_host_os])dnl +popdef([ac_cv_host_vendor])dnl +popdef([ac_cv_host_cpu])dnl +popdef([ac_cv_host_alias])dnl +popdef([ac_cv_host])dnl +popdef([host_os])dnl +popdef([host_vendor])dnl +popdef([host_cpu])dnl +popdef([host_alias])dnl +popdef([host])dnl +popdef([LDFLAGS])dnl +popdef([CPPFLAGS])dnl +popdef([CFLAGS])dnl +popdef([CPP])dnl +popdef([CC])dnl +popdef([ac_objext])dnl +popdef([ac_exeext])dnl +popdef([ac_cv_objext])dnl +popdef([ac_cv_exeext])dnl +popdef([ac_cv_prog_cc_g])dnl +popdef([ac_cv_prog_cc_cross])dnl +popdef([ac_cv_prog_cc_works])dnl +popdef([ac_cv_prog_gcc])dnl +popdef([ac_cv_prog_CPP])dnl + +dnl Finally, set Makefile variables +dnl +BUILD_EXEEXT=$ac_build_exeext +BUILD_OBJEXT=$ac_build_objext +AC_SUBST(BUILD_EXEEXT)dnl +AC_SUBST(BUILD_OBJEXT)dnl +AC_SUBST([CFLAGS_FOR_BUILD])dnl +AC_SUBST([CPPFLAGS_FOR_BUILD])dnl +AC_SUBST([LDFLAGS_FOR_BUILD])dnl +]) diff --git a/maestro3/Makefile.am b/maestro3/Makefile.am index 7544f12..0826570 100644 --- a/maestro3/Makefile.am +++ b/maestro3/Makefile.am @@ -17,5 +17,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/sb16/Makefile.am b/sb16/Makefile.am index 17eead9..58d0032 100644 --- a/sb16/Makefile.am +++ b/sb16/Makefile.am @@ -18,5 +18,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/vxloader/Makefile.am b/vxloader/Makefile.am index ca21494..adcc204 100644 --- a/vxloader/Makefile.am +++ b/vxloader/Makefile.am @@ -43,5 +43,14 @@ hotplugfwdir = hotplugfw_DATA = endif -%.xlx: %.rbt toxlx - ./toxlx < $< > $@ +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(toxlx_OBJECTS) : CC=$(CC_FOR_BUILD) +$(toxlx_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(toxlx_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +toxlx$(BUILD_EXEEXT): $(toxlx_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +%.xlx: %.rbt toxlx$(BUILD_EXEEXT) + ./toxlx$(BUILD_EXEEXT) < $< > $@ diff --git a/wavefront/Makefile.am b/wavefront/Makefile.am index e8c9fe1..8f67b59 100644 --- a/wavefront/Makefile.am +++ b/wavefront/Makefile.am @@ -17,5 +17,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) diff --git a/ymfpci/Makefile.am b/ymfpci/Makefile.am index 8795fbc..645b471 100644 --- a/ymfpci/Makefile.am +++ b/ymfpci/Makefile.am @@ -17,5 +17,14 @@ hotplugfwdir = hotplugfw_DATA = endif -$(firmware_files): fw_writer - ./fw_writer +LINK_FOR_BUILD.c = $(CC_FOR_BUILD) $(CFLAGS_FOR_BUILD) $(CPPFLAGS_FOR_BUILD) $(LDFLAGS_FOR_BUILD) $(TARGET_ARCH_FOR_BUILD) + +$(fw_writer_OBJECTS) : CC=$(CC_FOR_BUILD) +$(fw_writer_OBJECTS) : CFLAGS=$(CFLAGS_FOR_BUILD) +$(fw_writer_OBJECTS) : CPPFLAGS=$(CPPFLAGS_FOR_BUILD) + +fw_writer$(BUILD_EXEEXT): $(tobin_OBJECTS) + $(LINK_FOR_BUILD.c) $^ $(LOADLIBES_FOR_BUILD) $(LDLIBS_FOR_BUILD) -o $@ + +$(firmware_files): fw_writer$(BUILD_EXEEXT) + ./fw_writer$(BUILD_EXEEXT) -- 2.22.0

1 0

[PATCH] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 25 Jun '19

25 Jun '19

When DSP boots with timeout error, reinitialize, transfer and boot firmware to recover audio. Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 440bca7afbf1..a2844bdca1b8 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -455,13 +455,18 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + + ret = bxt_sst_init_fw(skl->dev, skl); + dev_warn(ctx->dev, "Reload fw status: %d\n", ret); + if (ret < 0) { + dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); + ret = -EIO; + goto err; + } } } -- 2.17.1

1 0

[PATCH] pcm_file: return errno in case of a failed write operation
by Adam Miartus 24 Jun '19

24 Jun '19

allows correct error code propagation for snd_pcm_file_write_bytes caller Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> --- src/pcm/pcm_file.c | 1 + 1 file changed, 1 insertion(+) diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 11d0953..ab47da8 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -402,6 +402,7 @@ static int snd_pcm_file_write_bytes(snd_pcm_t *pcm, size_t bytes) n = cont; err = write(file->fd, file->wbuf + file->file_ptr_bytes, n); if (err < 0) { + err = -errno; SYSERR("%s write failed, file data may be corrupt", file->fname); return err; } -- 2.7.4

1 0

[PATCH 1/2] pcm_file: remove double line
by Adam Miartus 24 Jun '19

24 Jun '19

snd_pcm_file_add_frames called two times by mistake, introduced in 2a800c0c4f259043c9217a4aa6a57afaf9f77477 Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> --- src/pcm/pcm_file.c | 1 - 1 file changed, 1 deletion(-) diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index ca8e0c8..11d0953 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -613,7 +613,6 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc snd_pcm_areas_from_buf(pcm, areas, buffer); snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); __snd_pcm_lock(pcm); - snd_pcm_file_add_frames(pcm, areas, 0, frames); if (snd_pcm_file_add_frames(pcm, areas, 0, frames) < 0) { __snd_pcm_unlock(pcm); return -EPIPE; -- 2.7.4

1 1

[PATCH 0/2] v2 pcm_file propagate write error instead of an assert
by Adam Miartus 12 Jun '19

12 Jun '19

patch possible case whenre file write causes assert in pcm_file plugin, by reporting error to user and printing a message user has a chance to recover error state, or capture correct logs before assert happens, and may have an idea how to recover Adam Miartus (2): pcm_file: do not disrupt playback on output file write fail pcm_file: report write output file error to api user src/pcm/pcm_file.c | 61 ++++++++++++++++++++++++++++++++++++++---------------- 1 file changed, 43 insertions(+), 18 deletions(-) -- 2.7.4

1 2

[PATCH 0/6] alsa-lib: Various portability fixes
by Michael Forney 12 Jun '19

12 Jun '19

Even though alsa-lib doesn't aim for ISO C conformance, hopefully these changes are uncontroversial enough to be considered anyway. Michael Forney (6): Use __func__ instead of __FUNCTION__ List cases in range explicitly Don't return in a void function Make sure parameter qualifiers match between declaration and definition Remove unused empty struct Avoid pointer arithmetic on `void *` aserver/aserver.c | 4 ++-- include/error.h | 8 ++++---- include/local.h | 4 ++-- src/conf.c | 22 ++++++++++++---------- src/pcm/pcm.c | 16 ++++++++-------- src/pcm/pcm_direct.h | 2 -- src/pcm/pcm_hw.c | 2 +- src/pcm/pcm_rate.c | 8 ++++---- src/pcm/pcm_share.c | 4 ++-- 9 files changed, 35 insertions(+), 35 deletions(-) -- 2.20.1

1 6

[PATCH 0/2] pcm_file write file assert case handling
by Adam Miartus 11 Jun '19

11 Jun '19

patched possible case where failed file write can cause assert in pcm_file plugin, by reporting error EPIPE and printing debug message. reporting an error with debug hint message is preferred to assert, user may have a way to recover in upper API level. providing a debug message makes it easier to debug the issue from logs. previously, reaching assert did not provide useful message to API level user. Adam Miartus (2): pcm_file: do not disrupt playback on output file write fail pcm_file: report write output file error to api user src/pcm/pcm_file.c | 63 +++++++++++++++++++++++++++++++++++++++--------------- 1 file changed, 46 insertions(+), 17 deletions(-) -- 2.7.4

3 4

[PATCH] ALSA: usb-audio: Enable .product_name override for Emagic, Unitor 8.
by Stefan Sauer 10 Jun '19

10 Jun '19

>From a41303935608e9a98653d4dff1e10baf1ce07e05 Mon Sep 17 00:00:00 2001 From: Stefan Sauer <ensonic(a)google.com> Date: Mon, 10 Jun 2019 11:26:53 +0200 Subject: [PATCH] ALSA: usb-audio: Enable .product_name override for Emagic Unitor 8. The Emagic Unitor 8 does not provide iManufacturer and iProduct fields in its device descriptor. These fields are used by alsa to make build the device name. Thus uncomment the .product-name in the quirks-table. Without this change the device shows up as 'USB Device 0x86a:0x01'. Output of lsusb and amidi: https://gist.github.com/ensonic/7820a102e91f31575be355da2b6b33bc Signed-off-by: Stefan Sauer <ensonic(a)google.com> --- sound/usb/quirks-table.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9e049f60e80e..e918ce346027 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2408,7 +2408,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE(0x086a, 0x0001), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Emagic", - /* .product_name = "Unitor8", */ + .product_name = "Unitor8", .ifnum = 2, .type = QUIRK_MIDI_EMAGIC, .data = & (const struct snd_usb_midi_endpoint_info) { -- 2.21.0

1 0

[PATCH 0/2] add pcm_file support in MMAP read access mode
by Adam Miartus 23 May '19

23 May '19

adds support for overwritting input buffer in MMAP access mode for in pcm_file plugin new api pcm_fast_ops callback mmap_begin is introduced to ease the operation needed to overwrite mmaped buffer. this allows to modify only the next continuous part of data in ring buffer. Adam Miartus (2): pcm: add mmap_begin callback to snd_pcm_fast_ops_t api pcm_file: add infile read support for mmap mode src/pcm/pcm.c | 6 ++++++ src/pcm/pcm_file.c | 31 +++++++++++++++++++++++++++++++ src/pcm/pcm_local.h | 1 + 3 files changed, 38 insertions(+) -- 2.7.4

1 2

[PATCH - alsa-lib 0/1] pcm_file add missing unlock
by Adam Miartus 23 May '19

23 May '19

previous patches: pcm: file: add support for infile reading in non interleaved mode pcm: file: use snd_pcm_file_areas_read_infile for readi introduced an issue where pcm would not be unlocked on early return Adam Miartus (1): pcm_file: add missing unlock on early return src/pcm/pcm_file.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) -- 2.7.4

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[PATCH - alsa-lib 4/4] pcm_file: add infile read support for mmap mode
by Adam Miartus 21 May '19

21 May '19

mmap_begin callback is used to copy data from input file to mmaped buffer guard for corner use of api (multiple mmap_begin calls by user) is introduced to check if next continuous buffer was already overwritten buffer is overwritten with input file data only in case of stream capture Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 10d227b..7d3439a 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -88,6 +88,7 @@ typedef struct { size_t buffer_bytes; struct wav_fmt wav_header; size_t filelen; + char ifmmap_overwritten; } snd_pcm_file_t; #if __BYTE_ORDER == __LITTLE_ENDIAN @@ -626,6 +627,8 @@ static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, const snd_pcm_channel_area_t *areas; snd_pcm_sframes_t result; + file->ifmmap_overwritten = 0; + result = snd_pcm_mmap_begin(file->gen.slave, &areas, &ofs, &siz); if (result >= 0) { assert(ofs == offset && siz == size); @@ -636,6 +639,32 @@ static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, return result; } +static int snd_pcm_file_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, + snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames) +{ + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + snd_pcm_uframes_t frames_if; + void *buffer = NULL; + int result; + + result = snd_pcm_mmap_begin(file->gen.slave, areas, offset, frames); + if (result < 0) + return result; + + if (pcm->stream != SND_PCM_STREAM_CAPTURE) + return result; + + /* user may run mmap_begin without mmap_commit multiple times in row */ + if (file->ifmmap_overwritten) + return result; + file->ifmmap_overwritten = 1; + + snd_pcm_file_areas_read_infile(pcm, *areas, *offset, *frames); + + return result; +} + static int snd_pcm_file_hw_free(snd_pcm_t *pcm) { snd_pcm_file_t *file = pcm->private_data; @@ -662,6 +691,7 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) file->wbuf_size = slave->buffer_size * 2; file->wbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->wbuf_size); file->wbuf_used_bytes = 0; + file->ifmmap_overwritten = 0; assert(!file->wbuf); file->wbuf = malloc(file->wbuf_size_bytes); if (file->wbuf == NULL) { @@ -773,7 +803,7 @@ static const snd_pcm_fast_ops_t snd_pcm_file_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .htimestamp = snd_pcm_generic_htimestamp, - .mmap_begin = NULL, + .mmap_begin = snd_pcm_file_mmap_begin, }; /** -- 2.7.4

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[PATCH - alsa-lib 3/4] pcm: add mmap_begin callback to snd_pcm_fast_ops_t api
by Adam Miartus 21 May '19

21 May '19

main motivation for adding the callback is to use it to enable operation on mmaped buffer before user access for pcm_file plugin support for MMAP read access with masking by data from input file is not implemented for pcm_file plugin, by adding this callback implementing such feature can be done by rewriting next continuous portion of buffer on each mmap_begin call plugins like softvol use pcm_plugin interface and overwrite the buffer by looping around it in avail_update callback, this patch hopes to simplify the task by adding new api callback, removing the need for rewriting pcm_file (to use pcm_plugin callbacks) and careful checking when looping around whole mmaped buffer Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 3a71d79..323926e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7129,7 +7129,13 @@ int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t f; snd_pcm_uframes_t avail; const snd_pcm_channel_area_t *xareas; + assert(pcm && areas && offset && frames); + + if (pcm->fast_ops->mmap_begin) + return pcm->fast_ops->mmap_begin(pcm->fast_op_arg, areas, offset, frames); + + /* fallback for plugins that do not specify new callback */ xareas = snd_pcm_mmap_areas(pcm); if (xareas == NULL) return -EBADFD; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 274726e..96049e0 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -990,6 +990,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index b75809c..328ef83 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -667,6 +667,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 58b1e53..5b9d9f5 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -544,6 +544,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index dcaa41d..10d227b 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -773,6 +773,7 @@ static const snd_pcm_fast_ops_t snd_pcm_file_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .htimestamp = snd_pcm_generic_htimestamp, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_generic.h b/src/pcm/pcm_generic.h index dfbe21a..683e60b 100644 --- a/src/pcm/pcm_generic.h +++ b/src/pcm/pcm_generic.h @@ -161,4 +161,3 @@ snd_pcm_chmap_query_t **snd_pcm_generic_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_generic_get_chmap(snd_pcm_t *pcm); int snd_pcm_generic_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); int snd_pcm_generic_may_wait_for_avail_min(snd_pcm_t *pcm, snd_pcm_uframes_t avail); - diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 4416d36..14396fa 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -200,6 +200,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hooks_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 91370a8..8ee1d69 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1481,6 +1481,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .poll_descriptors = NULL, .poll_descriptors_count = NULL, .poll_revents = NULL, + .mmap_begin = NULL, }; static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops_timer = { @@ -1512,6 +1513,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops_timer = { .poll_descriptors = snd_pcm_hw_poll_descriptors, .poll_descriptors_count = snd_pcm_hw_poll_descriptors_count, .poll_revents = snd_pcm_hw_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a437ca3..3e19499 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -940,6 +940,7 @@ static const snd_pcm_fast_ops_t snd_pcm_ioplug_fast_ops = { .poll_descriptors_count = snd_pcm_ioplug_poll_descriptors_count, .poll_descriptors = snd_pcm_ioplug_poll_descriptors, .poll_revents = snd_pcm_ioplug_poll_revents, + .mmap_begin = NULL, }; #endif /* !DOC_HIDDEN */ diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index d52229d..d5726eb 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -184,6 +184,7 @@ typedef struct { int (*poll_descriptors)(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space); /* locked */ int (*poll_revents)(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); /* locked */ int (*may_wait_for_avail_min)(snd_pcm_t *pcm, snd_pcm_uframes_t avail); + int (*mmap_begin)(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames); /* locked */ } snd_pcm_fast_ops_t; struct _snd_pcm { diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 20b4187..28bb578 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -548,6 +548,7 @@ static const snd_pcm_fast_ops_t snd_pcm_meter_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_mmap_emul.c b/src/pcm/pcm_mmap_emul.c index 009cebb..6fa5fe6 100644 --- a/src/pcm/pcm_mmap_emul.c +++ b/src/pcm/pcm_mmap_emul.c @@ -401,6 +401,7 @@ static const snd_pcm_fast_ops_t snd_pcm_mmap_emul_fast_ops = { .poll_descriptors_count = snd_pcm_generic_poll_descriptors_count, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; #ifndef DOC_HIDDEN diff --git a/src/pcm/pcm_multi.c b/src/pcm/pcm_multi.c index 53c414d..069d224 100644 --- a/src/pcm/pcm_multi.c +++ b/src/pcm/pcm_multi.c @@ -1042,6 +1042,7 @@ static const snd_pcm_fast_ops_t snd_pcm_multi_fast_ops = { .poll_descriptors = snd_pcm_multi_poll_descriptors, .poll_revents = snd_pcm_multi_poll_revents, .may_wait_for_avail_min = snd_pcm_multi_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index 1d81548..e6fa049 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -361,6 +361,7 @@ static const snd_pcm_fast_ops_t snd_pcm_null_fast_ops = { .avail_update = snd_pcm_null_avail_update, .mmap_commit = snd_pcm_null_mmap_commit, .htimestamp = snd_pcm_generic_real_htimestamp, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index ea60eb9..0714d2c 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -646,6 +646,7 @@ const snd_pcm_fast_ops_t snd_pcm_plugin_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_plugin_may_wait_for_avail_min, + .mmap_begin = NULL, }; #endif diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 5dc8a92..f29739b 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1217,6 +1217,7 @@ static const snd_pcm_fast_ops_t snd_pcm_rate_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_rate_poll_revents, .may_wait_for_avail_min = snd_pcm_plugin_may_wait_for_avail_min, + .mmap_begin = NULL, }; static const snd_pcm_ops_t snd_pcm_rate_ops = { diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c index 5a540c4..b803a82 100644 --- a/src/pcm/pcm_share.c +++ b/src/pcm/pcm_share.c @@ -1346,6 +1346,7 @@ static const snd_pcm_fast_ops_t snd_pcm_share_fast_ops = { .avail_update = snd_pcm_share_avail_update, .htimestamp = snd_pcm_share_htimestamp, .mmap_commit = snd_pcm_share_mmap_commit, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_shm.c b/src/pcm/pcm_shm.c index 26a27a5..a18720f 100644 --- a/src/pcm/pcm_shm.c +++ b/src/pcm/pcm_shm.c @@ -629,6 +629,7 @@ static const snd_pcm_fast_ops_t snd_pcm_shm_fast_ops = { .avail_update = snd_pcm_shm_avail_update, .mmap_commit = snd_pcm_shm_mmap_commit, .htimestamp = snd_pcm_shm_htimestamp, + .mmap_begin = NULL, }; static int make_local_socket(const char *filename) -- 2.7.4

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[PATCH - alsa-lib 2/4] pcm_file: use snd_pcm_file_areas_read_infile for readi
by Adam Miartus 21 May '19

21 May '19

use previously introduced helper function, this commit unifies behavior of readi and readn corner case behavior of readi is changed by this commit, previously, in case 0 bytes were red from file (EOF), frames = 0 was returned, signaling api user as if no data was red from slave, after the patch, amount of frames red from slave with data red from slave stored in buffer is returned when EOF is reached Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3c68265..dcaa41d 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -579,22 +579,21 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; + + __snd_pcm_lock(pcm); + + frames = _snd_pcm_readi(file->gen.slave, buffer, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readi(file->gen.slave, buffer, size); - if (n <= 0) - return n; - if (file->ifd >= 0) { - __snd_pcm_lock(pcm); - n = read(file->ifd, buffer, n * pcm->frame_bits / 8); - __snd_pcm_unlock(pcm); - if (n < 0) - return n; - n = n * 8 / pcm->frame_bits; - } snd_pcm_areas_from_buf(pcm, areas, buffer); - snd_pcm_file_add_frames(pcm, areas, 0, n); - return n; + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + + return frames; } /* locking */ -- 2.7.4

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[PATCH - alsa-lib 1/4] pcm_file: add support for infile reading in non interleaved mode
by Adam Miartus 21 May '19

21 May '19

add helper function to copy input file data to buffer mapped by areas, in case of an error, do not fill the areas, allowing device read buffer to be provided to api caller previously unused rbuf variable is reused for this purpose Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3a19cef..3c68265 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -77,6 +77,7 @@ typedef struct { snd_pcm_uframes_t appl_ptr; snd_pcm_uframes_t file_ptr_bytes; snd_pcm_uframes_t wbuf_size; + snd_pcm_uframes_t rbuf_size; size_t wbuf_size_bytes; size_t wbuf_used_bytes; char *wbuf; @@ -266,6 +267,39 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) return 0; } +/* fill areas with data from input file, return bytes red */ +static int snd_pcm_file_areas_read_infile(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t frames) +{ + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + ssize_t bytes; + + if (file->ifd < 0) + return -EBADF; + + if (file->rbuf == NULL) + return -ENOMEM; + + if (file->rbuf_size < frames) { + SYSERR("requested more frames than pcm buffer"); + return -ENOMEM; + } + + bytes = read(file->ifd, file->rbuf, snd_pcm_frames_to_bytes(pcm, frames)); + if (bytes < 0) { + SYSERR("read from file failed, error: %d", bytes); + return bytes; + } + + snd_pcm_areas_from_buf(pcm, areas_if, file->rbuf); + snd_pcm_areas_copy(areas, offset, areas_if, 0, pcm->channels, snd_pcm_bytes_to_frames(pcm, bytes), pcm->format); + + return bytes; +} + static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) { fmt->fmt = TO_LE16(0x01); @@ -568,19 +602,19 @@ static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; - if (file->ifd >= 0) { - SNDERR("DEBUG: Noninterleaved read not yet implemented.\n"); - return 0; /* TODO: Noninterleaved read */ - } + __snd_pcm_lock(pcm); + frames = _snd_pcm_readn(file->gen.slave, bufs, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readn(file->gen.slave, bufs, size); - if (n > 0) { - snd_pcm_areas_from_bufs(pcm, areas, bufs); - snd_pcm_file_add_frames(pcm, areas, 0, n); - } - return n; + snd_pcm_areas_from_bufs(pcm, areas, bufs); + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + return frames; } static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, @@ -609,9 +643,11 @@ static int snd_pcm_file_hw_free(snd_pcm_t *pcm) free(file->wbuf); free(file->wbuf_areas); free(file->final_fname); + free(file->rbuf); file->wbuf = NULL; file->wbuf_areas = NULL; file->final_fname = NULL; + file->rbuf = NULL; return snd_pcm_hw_free(file->gen.slave); } @@ -638,6 +674,15 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_file_hw_free(pcm); return -ENOMEM; } + assert(!file->rbuf); + file->rbuf_size = slave->buffer_size; + file->rbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->rbuf_size); + file->rbuf_used_bytes = 0; + file->rbuf = malloc(file->rbuf_size_bytes); + if (file->rbuf == NULL) { + snd_pcm_file_hw_free(pcm); + return -ENOMEM; + } file->appl_ptr = file->file_ptr_bytes = 0; for (channel = 0; channel < slave->channels; ++channel) { snd_pcm_channel_area_t *a = &file->wbuf_areas[channel]; -- 2.7.4

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[PATCH - alsa-lib 0/4] Add pcm_file input file support for MMAP
by Adam Miartus 21 May '19

21 May '19

implement overwriting read buffer with input file in case of RW_NONINTERLEAVED and MMAP access modes for alsa file plugin first two patches introduce a helper function for copying data from input file to readi and readn buffers for user. to facilitate this snd_pcm_areas_from_buf and snd_pcm_areas_from_bufs with snd_pcm_areas_copy is used. Previous readi behaviour in corner case of input file EOF is changed (more in patch comment). Patch 3,4 introduce new api callback mmap_begin to ease overwritting mmaped buffer in case of MMAP access. Adam Miartus (4): pcm_file: add support for infile reading in non interleaved mode pcm_file: use snd_pcm_file_areas_read_infile for readi pcm: add mmap_begin callback to snd_pcm_fast_ops_t api pcm_file: add infile read support for mmap mode src/pcm/pcm.c | 6 +++ src/pcm/pcm_dmix.c | 1 + src/pcm/pcm_dshare.c | 1 + src/pcm/pcm_dsnoop.c | 1 + src/pcm/pcm_file.c | 125 ++++++++++++++++++++++++++++++++++++++---------- src/pcm/pcm_generic.h | 1 - src/pcm/pcm_hooks.c | 1 + src/pcm/pcm_hw.c | 2 + src/pcm/pcm_ioplug.c | 1 + src/pcm/pcm_local.h | 1 + src/pcm/pcm_meter.c | 1 + src/pcm/pcm_mmap_emul.c | 1 + src/pcm/pcm_multi.c | 1 + src/pcm/pcm_null.c | 1 + src/pcm/pcm_plugin.c | 1 + src/pcm/pcm_rate.c | 1 + src/pcm/pcm_share.c | 1 + src/pcm/pcm_shm.c | 1 + 18 files changed, 122 insertions(+), 26 deletions(-) -- 2.7.4

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[PATCH - alsa-lib 1/4] pcm_file: add support for infile reading in non interleaved mode
by Adam Miartus 21 May '19

21 May '19

add helper function to copy input file data to buffer mapped by areas, in case of an error, do not fill the areas, allowing device read buffer to be provided to api caller previously unused rbuf variable is reused for this purpose Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3a19cef..3c68265 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -77,6 +77,7 @@ typedef struct { snd_pcm_uframes_t appl_ptr; snd_pcm_uframes_t file_ptr_bytes; snd_pcm_uframes_t wbuf_size; + snd_pcm_uframes_t rbuf_size; size_t wbuf_size_bytes; size_t wbuf_used_bytes; char *wbuf; @@ -266,6 +267,39 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) return 0; } +/* fill areas with data from input file, return bytes red */ +static int snd_pcm_file_areas_read_infile(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t frames) +{ + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + ssize_t bytes; + + if (file->ifd < 0) + return -EBADF; + + if (file->rbuf == NULL) + return -ENOMEM; + + if (file->rbuf_size < frames) { + SYSERR("requested more frames than pcm buffer"); + return -ENOMEM; + } + + bytes = read(file->ifd, file->rbuf, snd_pcm_frames_to_bytes(pcm, frames)); + if (bytes < 0) { + SYSERR("read from file failed, error: %d", bytes); + return bytes; + } + + snd_pcm_areas_from_buf(pcm, areas_if, file->rbuf); + snd_pcm_areas_copy(areas, offset, areas_if, 0, pcm->channels, snd_pcm_bytes_to_frames(pcm, bytes), pcm->format); + + return bytes; +} + static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) { fmt->fmt = TO_LE16(0x01); @@ -568,19 +602,19 @@ static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; - if (file->ifd >= 0) { - SNDERR("DEBUG: Noninterleaved read not yet implemented.\n"); - return 0; /* TODO: Noninterleaved read */ - } + __snd_pcm_lock(pcm); + frames = _snd_pcm_readn(file->gen.slave, bufs, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readn(file->gen.slave, bufs, size); - if (n > 0) { - snd_pcm_areas_from_bufs(pcm, areas, bufs); - snd_pcm_file_add_frames(pcm, areas, 0, n); - } - return n; + snd_pcm_areas_from_bufs(pcm, areas, bufs); + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + return frames; } static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, @@ -609,9 +643,11 @@ static int snd_pcm_file_hw_free(snd_pcm_t *pcm) free(file->wbuf); free(file->wbuf_areas); free(file->final_fname); + free(file->rbuf); file->wbuf = NULL; file->wbuf_areas = NULL; file->final_fname = NULL; + file->rbuf = NULL; return snd_pcm_hw_free(file->gen.slave); } @@ -638,6 +674,15 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_file_hw_free(pcm); return -ENOMEM; } + assert(!file->rbuf); + file->rbuf_size = slave->buffer_size; + file->rbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->rbuf_size); + file->rbuf_used_bytes = 0; + file->rbuf = malloc(file->rbuf_size_bytes); + if (file->rbuf == NULL) { + snd_pcm_file_hw_free(pcm); + return -ENOMEM; + } file->appl_ptr = file->file_ptr_bytes = 0; for (channel = 0; channel < slave->channels; ++channel) { snd_pcm_channel_area_t *a = &file->wbuf_areas[channel]; -- 2.7.4

1 0

[PATCH - alsa-lib 4/4] pcm_file: add infile read support for mmap mode
by Miartus, Adam (Arion Recruitment; ADITG/ESM) 20 May '19

20 May '19

From: Adam Miartus <amiartus(a)de.adit-jv.com> mmap_begin callback is used to copy data from input file to mmaped buffer guard for corner use of api (multiple mmap_begin calls by user) is introduced to check if next continuous buffer was already overwritten buffer is overwritten with input file data only in case of stream capture Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index c8f4c84..d27586c 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -88,6 +88,7 @@ typedef struct { size_t buffer_bytes; struct wav_fmt wav_header; size_t filelen; + char ifmmap_overwritten; } snd_pcm_file_t; #if __BYTE_ORDER == __LITTLE_ENDIAN @@ -624,6 +625,8 @@ static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, const snd_pcm_channel_area_t *areas; snd_pcm_sframes_t result; + file->ifmmap_overwritten = 0; + result = snd_pcm_mmap_begin(file->gen.slave, &areas, &ofs, &siz); if (result >= 0) { assert(ofs == offset && siz == size); @@ -634,6 +637,32 @@ static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, return result; } +static int snd_pcm_file_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, + snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames) { + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + snd_pcm_uframes_t frames_if; + void *buffer = NULL; + int result; + + result = snd_pcm_mmap_begin(file->gen.slave, areas, offset, frames); + if (result < 0) + return result; + + if (pcm->stream != SND_PCM_STREAM_CAPTURE) + return result; + + /* user may run mmap_begin without mmap_commit multiple times in row */ + if (file->ifmmap_overwritten) + return result; + file->ifmmap_overwritten = 1; + + snd_pcm_file_areas_read_infile(pcm, *areas, *offset, *frames); + + return result; +} + static int snd_pcm_file_hw_free(snd_pcm_t *pcm) { snd_pcm_file_t *file = pcm->private_data; @@ -660,6 +689,7 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) file->wbuf_size = slave->buffer_size * 2; file->wbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->wbuf_size); file->wbuf_used_bytes = 0; + file->ifmmap_overwritten = 0; assert(!file->wbuf); file->wbuf = malloc(file->wbuf_size_bytes); if (file->wbuf == NULL) { @@ -771,7 +801,7 @@ static const snd_pcm_fast_ops_t snd_pcm_file_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .htimestamp = snd_pcm_generic_htimestamp, - .mmap_begin = NULL, + .mmap_begin = snd_pcm_file_mmap_begin, }; /** -- 2.7.4

1 0

[PATCH - alsa-lib 3/4] pcm: add mmap_begin callback to snd_pcm_fast_ops_t api
by Miartus, Adam (Arion Recruitment; ADITG/ESM) 20 May '19

20 May '19

From: Adam Miartus <amiartus(a)de.adit-jv.com> main motivation for adding the callback is to use it to enable operation on mmaped buffer before user access for pcm_file plugin support for MMAP read access with masking by data from input file is not implemented for pcm_file plugin, by adding this callback implementing such feature can be done by rewriting next continuous portion of buffer on each mmap_begin call plugins like softvol use pcm_plugin interface and overwrite the buffer by looping around it in avail_update callback, this patch hopes to simplify the task by adding new api callback, removing the need for rewriting pcm_file (to use pcm_plugin callbacks) and careful checking when looping around whole mmaped buffer Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 3a71d79..323926e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7129,7 +7129,13 @@ int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t f; snd_pcm_uframes_t avail; const snd_pcm_channel_area_t *xareas; + assert(pcm && areas && offset && frames); + + if (pcm->fast_ops->mmap_begin) + return pcm->fast_ops->mmap_begin(pcm->fast_op_arg, areas, offset, +frames); + + /* fallback for plugins that do not specify new callback */ xareas = snd_pcm_mmap_areas(pcm); if (xareas == NULL) return -EBADFD; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 274726e..96049e0 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -990,6 +990,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index b75809c..328ef83 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -667,6 +667,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 58b1e53..5b9d9f5 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -544,6 +544,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 29ab2fb..c8f4c84 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -771,6 +771,7 @@ static const snd_pcm_fast_ops_t snd_pcm_file_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .htimestamp = snd_pcm_generic_htimestamp, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_generic.h b/src/pcm/pcm_generic.h index dfbe21a..683e60b 100644 --- a/src/pcm/pcm_generic.h +++ b/src/pcm/pcm_generic.h @@ -161,4 +161,3 @@ snd_pcm_chmap_query_t **snd_pcm_generic_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_generic_get_chmap(snd_pcm_t *pcm); int snd_pcm_generic_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); int snd_pcm_generic_may_wait_for_avail_min(snd_pcm_t *pcm, snd_pcm_uframes_t avail); - diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 4416d36..14396fa 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -200,6 +200,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hooks_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 91370a8..8ee1d69 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1481,6 +1481,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .poll_descriptors = NULL, .poll_descriptors_count = NULL, .poll_revents = NULL, + .mmap_begin = NULL, }; static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops_timer = { @@ -1512,6 +1513,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops_timer = { .poll_descriptors = snd_pcm_hw_poll_descriptors, .poll_descriptors_count = snd_pcm_hw_poll_descriptors_count, .poll_revents = snd_pcm_hw_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a437ca3..3e19499 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -940,6 +940,7 @@ static const snd_pcm_fast_ops_t snd_pcm_ioplug_fast_ops = { .poll_descriptors_count = snd_pcm_ioplug_poll_descriptors_count, .poll_descriptors = snd_pcm_ioplug_poll_descriptors, .poll_revents = snd_pcm_ioplug_poll_revents, + .mmap_begin = NULL, }; #endif /* !DOC_HIDDEN */ diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index d52229d..d5726eb 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -184,6 +184,7 @@ typedef struct { int (*poll_descriptors)(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space); /* locked */ int (*poll_revents)(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); /* locked */ int (*may_wait_for_avail_min)(snd_pcm_t *pcm, snd_pcm_uframes_t avail); + int (*mmap_begin)(snd_pcm_t *pcm, const snd_pcm_channel_area_t +**areas, snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames); /* +locked */ } snd_pcm_fast_ops_t; struct _snd_pcm { diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 20b4187..28bb578 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -548,6 +548,7 @@ static const snd_pcm_fast_ops_t snd_pcm_meter_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_mmap_emul.c b/src/pcm/pcm_mmap_emul.c index 009cebb..6fa5fe6 100644 --- a/src/pcm/pcm_mmap_emul.c +++ b/src/pcm/pcm_mmap_emul.c @@ -401,6 +401,7 @@ static const snd_pcm_fast_ops_t snd_pcm_mmap_emul_fast_ops = { .poll_descriptors_count = snd_pcm_generic_poll_descriptors_count, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; #ifndef DOC_HIDDEN diff --git a/src/pcm/pcm_multi.c b/src/pcm/pcm_multi.c index 53c414d..069d224 100644 --- a/src/pcm/pcm_multi.c +++ b/src/pcm/pcm_multi.c @@ -1042,6 +1042,7 @@ static const snd_pcm_fast_ops_t snd_pcm_multi_fast_ops = { .poll_descriptors = snd_pcm_multi_poll_descriptors, .poll_revents = snd_pcm_multi_poll_revents, .may_wait_for_avail_min = snd_pcm_multi_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index 1d81548..e6fa049 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -361,6 +361,7 @@ static const snd_pcm_fast_ops_t snd_pcm_null_fast_ops = { .avail_update = snd_pcm_null_avail_update, .mmap_commit = snd_pcm_null_mmap_commit, .htimestamp = snd_pcm_generic_real_htimestamp, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index ea60eb9..0714d2c 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -646,6 +646,7 @@ const snd_pcm_fast_ops_t snd_pcm_plugin_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_plugin_may_wait_for_avail_min, + .mmap_begin = NULL, }; #endif diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 5dc8a92..f29739b 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1217,6 +1217,7 @@ static const snd_pcm_fast_ops_t snd_pcm_rate_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_rate_poll_revents, .may_wait_for_avail_min = snd_pcm_plugin_may_wait_for_avail_min, + .mmap_begin = NULL, }; static const snd_pcm_ops_t snd_pcm_rate_ops = { diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c index 5a540c4..b803a82 100644 --- a/src/pcm/pcm_share.c +++ b/src/pcm/pcm_share.c @@ -1346,6 +1346,7 @@ static const snd_pcm_fast_ops_t snd_pcm_share_fast_ops = { .avail_update = snd_pcm_share_avail_update, .htimestamp = snd_pcm_share_htimestamp, .mmap_commit = snd_pcm_share_mmap_commit, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_shm.c b/src/pcm/pcm_shm.c index 26a27a5..a18720f 100644 --- a/src/pcm/pcm_shm.c +++ b/src/pcm/pcm_shm.c @@ -629,6 +629,7 @@ static const snd_pcm_fast_ops_t snd_pcm_shm_fast_ops = { .avail_update = snd_pcm_shm_avail_update, .mmap_commit = snd_pcm_shm_mmap_commit, .htimestamp = snd_pcm_shm_htimestamp, + .mmap_begin = NULL, }; static int make_local_socket(const char *filename) -- 2.7.4

1 0

[PATCH - alsa-lib 2/4] pcm_file: use snd_pcm_file_areas_read_infile for readi
by Miartus, Adam (Arion Recruitment; ADITG/ESM) 20 May '19

20 May '19

From: Adam Miartus <amiartus(a)de.adit-jv.com> use previously introduced helper function, this commit unifies behavior of readi and readn corner case behavior of readi is changed by this commit, previously, in case 0 bytes were red from file (EOF), frames = 0 was returned, signaling api user as if no data was red from slave, after the patch, amount of frames red from slave with data red from slave stored in buffer is returned when EOF is reached Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 7998b64..29ab2fb 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -577,22 +577,21 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; + + __snd_pcm_lock(pcm); + + frames = _snd_pcm_readi(file->gen.slave, buffer, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readi(file->gen.slave, buffer, size); - if (n <= 0) - return n; - if (file->ifd >= 0) { - __snd_pcm_lock(pcm); - n = read(file->ifd, buffer, n * pcm->frame_bits / 8); - __snd_pcm_unlock(pcm); - if (n < 0) - return n; - n = n * 8 / pcm->frame_bits; - } snd_pcm_areas_from_buf(pcm, areas, buffer); - snd_pcm_file_add_frames(pcm, areas, 0, n); - return n; + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + + return frames; } /* locking */ -- 2.7.4

1 0

FW: [PATCH - alsa-lib 1/4] pcm_file: add support for infile reading in non interleaved mode
by Miartus, Adam (Arion Recruitment; ADITG/ESM) 20 May '19

20 May '19

From: Adam Miartus <amiartus(a)de.adit-jv.com> add helper function to copy input file data to buffer mapped by areas, in case of an error, do not fill the areas, allowing device read buffer to be provided to api caller previously unused rbuf variable is reused for this purpose Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3a19cef..7998b64 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -77,6 +77,7 @@ typedef struct { snd_pcm_uframes_t appl_ptr; snd_pcm_uframes_t file_ptr_bytes; snd_pcm_uframes_t wbuf_size; + snd_pcm_uframes_t rbuf_size; size_t wbuf_size_bytes; size_t wbuf_used_bytes; char *wbuf; @@ -266,6 +267,37 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) return 0; } +/* fill areas with data from input file, return bytes red */ static int +snd_pcm_file_areas_read_infile(snd_pcm_t *pcm, const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, snd_pcm_uframes_t frames) { + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + ssize_t bytes; + + if (file->ifd < 0) + return -EBADF; + + if (file->rbuf == NULL) + return -ENOMEM; + + if (file->rbuf_size < frames) { + SYSERR("requested more frames than pcm buffer"); + return -ENOMEM; + } + + bytes = read(file->ifd, file->rbuf, snd_pcm_frames_to_bytes(pcm, frames)); + if (bytes < 0) { + SYSERR("read from file failed, error: %d", bytes); + return bytes; + } + + snd_pcm_areas_from_buf(pcm, areas_if, file->rbuf); + snd_pcm_areas_copy(areas, offset, areas_if, 0, pcm->channels, +snd_pcm_bytes_to_frames(pcm, bytes), pcm->format); + + return bytes; +} + static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) { fmt->fmt = TO_LE16(0x01); @@ -568,19 +600,19 @@ static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; - if (file->ifd >= 0) { - SNDERR("DEBUG: Noninterleaved read not yet implemented.\n"); - return 0; /* TODO: Noninterleaved read */ - } + __snd_pcm_lock(pcm); + frames = _snd_pcm_readn(file->gen.slave, bufs, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readn(file->gen.slave, bufs, size); - if (n > 0) { - snd_pcm_areas_from_bufs(pcm, areas, bufs); - snd_pcm_file_add_frames(pcm, areas, 0, n); - } - return n; + snd_pcm_areas_from_bufs(pcm, areas, bufs); + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + return frames; } static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, @@ -609,9 +641,11 @@ static int snd_pcm_file_hw_free(snd_pcm_t *pcm) free(file->wbuf); free(file->wbuf_areas); free(file->final_fname); + free(file->rbuf); file->wbuf = NULL; file->wbuf_areas = NULL; file->final_fname = NULL; + file->rbuf = NULL; return snd_pcm_hw_free(file->gen.slave); } @@ -638,6 +672,15 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_file_hw_free(pcm); return -ENOMEM; } + assert(!file->rbuf); + file->rbuf_size = slave->buffer_size; + file->rbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->rbuf_size); + file->rbuf_used_bytes = 0; + file->rbuf = malloc(file->rbuf_size_bytes); + if (file->rbuf == NULL) { + snd_pcm_file_hw_free(pcm); + return -ENOMEM; + } file->appl_ptr = file->file_ptr_bytes = 0; for (channel = 0; channel < slave->channels; ++channel) { snd_pcm_channel_area_t *a = &file->wbuf_areas[channel]; -- 2.7.4

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[PATCH v2 0/6]
by vanitha.channaiah＠in.bosch.com 15 May '19

15 May '19

- Updated the patches by incorporating review comments from Takashi Iwai-san [PATCH 1/5] pcm: direct: Add generic hw_ptr_alignment function for dmix, dshare and dsnoop [Takashi Iwai:] > The patch description needs rephrasing. What actually this does is to move the code from pcm_dmix.c to pcm_direct.c > and its header so that the helper code can be re-used by other direct-PCM plugins. - Commit message is rephrased as suggested. - New commit: [PATCH v2 1/6] pcm: direct: Add generic hw_ptr_alignment function for [PATCH 2/5] pcm: dshare: Added "hw_ptr_alignment" option in configuration for alignment of slave pointers [Takashi Iwai:] > Again, this patch description is too ambiguous. > And, if it enables the hw_ptr_alignment option, update the documentation as well. - Commit message is explained in detail for the changes done. - Documentation updated. - New commit: [PATCH v2 2/6] pcm: dshare: Added "hw_ptr_alignment" option in [PATCH 3/5] pcm: dsnoop: Added "hw_ptr_alignment" option in configuration for slave pointer alignment [Takashi Iwai:] > Ditto as patch 2, the description is too ambiguous, and the update of documentation is missing. > It's not good to change the helper function semantics out of sudden, even without any description. - Commit message is explained in detail for the changes done. - Documentation updated. - Divided the patch with commit ("pcm: dsnoop: Add hw_ptr_alignment option in configuration") into additional patch commit ("pcm: direct: Round up of slave_app_ptr pointer if buffer") - Usecase scenario is described for the changes done in helper function. - New commit: [PATCH v2 3/6] pcm: dsnoop: Added "hw_ptr_alignment" option in [PATCH v2 4/6] pcm: direct: Round up of slave_app_ptr pointer if buffer [PATCH 4/5] pcm: restructuring sw params function [Takashi Iwai:] > I see no reason to do that. Please describe. - Commit message is explained in detail why reformating was done. - New commit: [PATCH v2 5/6] pcm: restructuring sw params function [PATCH 5/5] pcm: Update pcm->avail_min with needed_slave_avail_min, after reading unaligned frames [Takashi Iwai:] > This kind of changes in the core code should be avoided as much as possible, especially if it's only relevant with the specific plugins. > Sorry, this isn't convincing enough. If this is a MUST, please clarify better. - Commit message is explained in detail with the generic usecase and specific use case we came across. - New commit: [PATCH v2 6/6] pcm: Update pcm->avail_min with needed_slave_avail_min,

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