May 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RFC 00/11] ALSA: pcm: introduce copy_frame operation and SND_PCM_PROXY_DRIVER_SUPPORT
by Takashi Sakamoto 25 May '17

25 May '17

Hi, This RFC patchset is my concept work of another solution against a series of issues in ALSA PCM core, for which Iwai Takashi work in his hasty proposals[2][3]. Included changes are also available in my personal repository on github.com[4]. The aim of this work is to declare existent issues and to propose solutions as moderated as possible. Code refactoring is done to assist it. Patch 01-05: code refactoring phase, with an alias of function prototype. Patch 06-09: driver API changing phase. The .copy_frames is introduced. Patch 10-11: integration phase for proxy drivers, to get new configuration. I need to do this work with a little time (half a day), thus changes are as minimal as possible, especially two drivers are just modified even if drivers in below list should be changed: * drivers/media/pci/solo6x10/solo6x10-g723.c * sound/drivers/dummy.c * sound/isa/gus/gus_pcm.c * sound/isa/sb/emu8000_pcm.c * sound/pci/es1938.c * sound/pci/korg1212/korg1212.c * sound/pci/nm256/nm256.c * sound/pci/rme32.c * sound/pci/rme96.c * sound/pci/rme9652/hdsp.c * sound/pci/rme9652/rme9652.c * sound/sh/sh_dac_audio.c * sound/soc/blackfin/bf5xx-ac97-pcm.c * sound/soc/blackfin/bf5xx-i2s-pcm.c Furthermore, I apologize that this is untested. I wish you to check the concept. [1] [alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120542 [2] [alsa-devel] [PATCH 00/16] ALSA: Convert to new copy_silence PCM ops http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120895.html [3] [alsa-devel] [PATCH 0/4] ALSA: Extend PCM buffer-copy helpers http://mailman.alsa-project.org/pipermail/alsa-devel/2017-May/120957.html [4] takaswie/sound https://github.com/takaswie/sound/tree/topic/add-pcm-frame-ops Takashi Sakamoto (11): ALSA: pcm: introduce an alias of prototype to copy PCM frames ALSA: pcm: use new function alias instead of local one ALSA: pcm: refactoring silence operation ALSA: pcm: add alternative calls ALSA: core: remove duplicated codes to handle PCM frames ALSA: pcm: add new operation; copy_frames ALSA: rme9652: a sample for drivers which support interleaved buffer ALSA: gus: a sample for drivers which support non-interleaved buffer ALSA: pcm: remove copy and silence callbacks ALSA: pcm: add client_space parameter to runtime of PCM substream for PCM proxy drivers ALSA: pcm: add new configuration for PCM proxy drivers drivers/usb/gadget/Kconfig | 3 +- include/sound/pcm.h | 13 +- sound/core/Kconfig | 15 +- sound/core/pcm_lib.c | 448 +++++++++++++++++++++++++++++---------------- sound/core/pcm_native.c | 10 + sound/isa/gus/gus_pcm.c | 100 +++++----- sound/pci/rme9652/hdsp.c | 59 +++--- 7 files changed, 398 insertions(+), 250 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ALSA: sb: remove needless evaluation in implementation for copy callback
by Takashi Sakamoto 25 May '17

25 May '17

In design of ALSA pcm core, 'struct snd_pcm_ops.copy' is expected to copy PCM frames, according to frame alignment on intermediate buffer for userspace and dedicated buffer for data transmission. In this callback, value of 'channel' argument depends on the frame alignment, which drivers registers to runtime of PCM substream. When target devices can handle non-interleaved buffer, this value has positive value, otherwise negative. ALSA driver for PCM component of EMU8000 chip is programmed with local macro to switch the frame alignment. The 'copy' operation in non-interleaved side has evaluation of the 'channel' argument (actually it's 'voice' argument). This is useless. This commit remove the evaluation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/isa/sb/emu8000_pcm.c | 28 ++++------------------------ 1 file changed, 4 insertions(+), 24 deletions(-) diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index 32f234f494e5..bab796d75dbd 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -450,20 +450,8 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs, struct snd_emu8000 *emu = rec->emu; snd_emu8000_write_wait(emu, 1); - if (voice == -1) { - unsigned short *buf = src; - int i, err; - count /= rec->voices; - for (i = 0; i < rec->voices; i++) { - err = emu8k_transfer_block(emu, pos + rec->loop_start[i], buf, count); - if (err < 0) - return err; - buf += count; - } - return 0; - } else { - return emu8k_transfer_block(emu, pos + rec->loop_start[voice], src, count); - } + return emu8k_transfer_block(emu, pos + rec->loop_start[voice], src, + count); } /* make a channel block silence */ @@ -487,17 +475,9 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs, struct snd_emu8000 *emu = rec->emu; snd_emu8000_write_wait(emu, 1); - if (voice == -1 && rec->voices == 1) + if (rec->voices == 1) voice = 0; - if (voice == -1) { - int err; - err = emu8k_silence_block(emu, pos + rec->loop_start[0], count / 2); - if (err < 0) - return err; - return emu8k_silence_block(emu, pos + rec->loop_start[1], count / 2); - } else { - return emu8k_silence_block(emu, pos + rec->loop_start[voice], count); - } + return emu8k_silence_block(emu, pos + rec->loop_start[voice], count); } #else /* interleave */ -- 2.11.0

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[alsa-devel] [alsa-lib][PATCH] test: add a test for list operation to user-defined element sets
by Takashi Sakamoto 25 May '17

25 May '17

Current implementation of test for user-defined element doesn't perform list operation. This commit adds easy test for the operation. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/user-ctl-element-set.c | 80 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 80 insertions(+) diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c index 75083d21..8635c156 100644 --- a/test/user-ctl-element-set.c +++ b/test/user-ctl-element-set.c @@ -433,6 +433,78 @@ static int check_event(struct elem_set_trial *trial, unsigned int mask, return 0; } +static int check_elem_list(struct elem_set_trial *trial) +{ + snd_ctl_elem_list_t *list; + snd_ctl_elem_id_t *id; + int e; + unsigned int i; + int err; + + snd_ctl_elem_list_alloca(&list); + snd_ctl_elem_id_alloca(&id); + + err = snd_ctl_elem_list(trial->handle, list); + if (err < 0) + return err; + + /* Certainly some elements are already added. */ + if (snd_ctl_elem_list_get_count(list) == 0) + return -EIO; + + err = snd_ctl_elem_list_alloc_space(list, + snd_ctl_elem_list_get_count(list)); + if (err < 0) + return err; + + err = snd_ctl_elem_list(trial->handle, list); + if (err < 0) + goto end; + + if (trial->element_count > snd_ctl_elem_list_get_count(list)) { + err = -EIO; + goto end; + } + + i = 0; + for (e = 0; e < snd_ctl_elem_list_get_count(list); ++e) { + snd_ctl_elem_list_get_id(list, e, id); + + if (strcmp(snd_ctl_elem_id_get_name(id), + snd_ctl_elem_id_get_name(trial->id)) != 0) + continue; + if (snd_ctl_elem_id_get_interface(id) != + snd_ctl_elem_id_get_interface(trial->id)) + continue; + if (snd_ctl_elem_id_get_device(id) != + snd_ctl_elem_id_get_device(trial->id)) + continue; + if (snd_ctl_elem_id_get_subdevice(id) != + snd_ctl_elem_id_get_subdevice(trial->id)) + continue; + + /* + * Here, I expect the list includes element ID data in numerical + * order. Actually, it does. + */ + if (snd_ctl_elem_id_get_numid(id) != + snd_ctl_elem_id_get_numid(trial->id) + i) + continue; + if (snd_ctl_elem_id_get_index(id) != + snd_ctl_elem_id_get_index(trial->id) + i) + continue; + + ++i; + } + + if (i != trial->element_count) + err = -EIO; +end: + snd_ctl_elem_list_free_space(list); + + return err; +} + static int check_elem_set_props(struct elem_set_trial *trial) { snd_ctl_elem_id_t *id; @@ -701,6 +773,14 @@ int main(void) break; } + /* Check added elements are retrieved in a list. */ + err = check_elem_list(&trial); + if (err < 0) { + printf("Fail to list each element with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + /* Check properties of each element in this element set. */ err = check_elem_set_props(&trial); if (err < 0) { -- 2.11.0

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[alsa-devel] [asoc:topic/dai-id 1/1] sound/soc/soc-core.c:4054:9: error: implicit declaration of function 'of_graph_get_port_parent'
by kbuild test robot 25 May '17

25 May '17

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/dai-id head: a180e8b988437b3e84a1b501ac4d073467602ca6 commit: a180e8b988437b3e84a1b501ac4d073467602ca6 [1/1] ASoC: add snd_soc_get_dai_id() function config: x86_64-randconfig-x010-201721 (attached as .config) compiler: gcc-6 (Debian 6.2.0-3) 6.2.0 20160901 reproduce: git checkout a180e8b988437b3e84a1b501ac4d073467602ca6 # save the attached .config to linux build tree make ARCH=x86_64 All error/warnings (new ones prefixed by >>): sound/soc/soc-core.c: In function 'snd_soc_get_dai_id': >> sound/soc/soc-core.c:4054:9: error: implicit declaration of function 'of_graph_get_port_parent' [-Werror=implicit-function-declaration] node = of_graph_get_port_parent(ep); ^~~~~~~~~~~~~~~~~~~~~~~~ >> sound/soc/soc-core.c:4054:7: warning: assignment makes pointer from integer without a cast [-Wint-conversion] node = of_graph_get_port_parent(ep); ^ cc1: some warnings being treated as errors vim +/of_graph_get_port_parent +4054 sound/soc/soc-core.c 4048 int snd_soc_get_dai_id(struct device_node *ep) 4049 { 4050 struct snd_soc_component *pos; 4051 struct device_node *node; 4052 int ret; 4053 > 4054 node = of_graph_get_port_parent(ep); 4055 4056 /* 4057 * For example HDMI case, HDMI has video/sound port, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Fix type warning in sprintf
by Richard Fitzgerald 24 May '17

24 May '17

The shift member of struct soc_mixer_control is unsigned int. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 20695b6..a7dc760 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2654,7 +2654,7 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; char preload[32]; - snprintf(preload, ARRAY_SIZE(preload), "DSP%d Preload", mc->shift); + snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); dsp->preloaded = ucontrol->value.integer.value[0]; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Fix use-after-free at card unregistration
by Takashi Iwai 24 May '17

24 May '17

soc_cleanup_card_resources() call snd_card_free() at the last of its procedure. This turned out to lead to a use-after-free. PCM runtimes have been already removed via soc_remove_pcm_runtimes(), while it's dereferenced later in soc_pcm_free() called via snd_card_free(). The fix is simple: just move the snd_card_free() call to the beginning of the whole procedure. This also gives another benefit: it guarantees that all operations have been shut down before actually releasing the resources, which was racy until now. Reported-and-tested-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index aae099c0e502..754e3ef8d7ae 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2286,6 +2286,9 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); + /* free the ALSA card at first; this syncs with pending operations */ + snd_card_free(card->snd_card); + /* remove and free each DAI */ soc_remove_dai_links(card); soc_remove_pcm_runtimes(card); @@ -2300,9 +2303,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) if (card->remove) card->remove(card); - snd_card_free(card->snd_card); return 0; - } /* removes a socdev */ -- 2.13.0

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[alsa-devel] Applied "MAINTAINERS: Update email address for patches to Wolfson parts" to the asoc tree
by Mark Brown 24 May '17

24 May '17

The patch MAINTAINERS: Update email address for patches to Wolfson parts has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3a1672bb0186e940fc8521d19069f57690e850c2 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Date: Wed, 24 May 2017 11:01:32 +0100 Subject: [PATCH] MAINTAINERS: Update email address for patches to Wolfson parts A somewhat overdue update of the address for sending patches on Wolfson parts to since our acquision a couple of years ago by Cirrus Logic. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index f7d568b8f133..517829e3bd1d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13858,7 +13858,7 @@ S: Odd fixes F: drivers/net/wireless/wl3501* WOLFSON MICROELECTRONICS DRIVERS -L: patches(a)opensource.wolfsonmicro.com +L: patches(a)opensource.cirrus.com T: git https://github.com/CirrusLogic/linux-drivers.git W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported -- 2.11.0

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[alsa-devel] [PATCH] ASoC: rsnd: SSI PIO adjust to 24bit mode
by Kuninori Morimoto 24 May '17

24 May '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 90431eb49bff ("ASoC: rsnd: don't use PDTA bit for 24bit on SSI") fixups 24bit mode data alignment, but PIO was not cared. This patch fixes PIO mode 24bit data alignment Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark, this is for mark/fix/rcar branch, not mark/topic/rcar sound/soc/sh/rcar/ssi.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 57daac0..0822ae0b 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -567,6 +567,13 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); u32 *buf = (u32 *)(runtime->dma_area + rsnd_dai_pointer_offset(io, 0)); + int shift = 0; + + switch (runtime->sample_bits) { + case 32: + shift = 8; + break; + } /* * 8/16/32 data can be assesse to TDR/RDR register @@ -574,9 +581,9 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, * see rsnd_ssi_init() */ if (rsnd_io_is_play(io)) - rsnd_mod_write(mod, SSITDR, *buf); + rsnd_mod_write(mod, SSITDR, (*buf) << shift); else - *buf = rsnd_mod_read(mod, SSIRDR); + *buf = (rsnd_mod_read(mod, SSIRDR) >> shift); elapsed = rsnd_dai_pointer_update(io, sizeof(*buf)); } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: cht_bsw_max98090_ti: Remove unused function cht_get_codec_dai()
by Matthias Kaehlcke 24 May '17

24 May '17

Looks like the function has never been used since it was added by commit 17119a465706 ("ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_max98090_ti"). Removing it fixes the following warning when building with clang: sound/soc/intel/boards/cht_bsw_max98090_ti.c:42:35: error: unused function 'cht_get_codec_dai' [-Werror,-Wunused-function] Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index 742bc0d4e681..20755ecc7f9e 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -39,18 +39,6 @@ struct cht_mc_private { bool ts3a227e_present; }; -static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - - list_for_each_entry(rtd, &card->rtd_list, list) { - if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, - strlen(CHT_CODEC_DAI))) - return rtd->codec_dai; - } - return NULL; -} - static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), -- 2.13.0.219.gdb65acc882-goog

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[alsa-devel] [PATCH] ASoC: cs4271: Remove unnecessary additional variable definition
by Charles Keepax 24 May '17

24 May '17

The function already defines a ret variable at the top and makes no particular use of the shadowed definition, as such remove the redundant definition. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4271.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e78b5f0..d882477 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -674,8 +674,6 @@ static int cs4271_common_probe(struct device *dev, cs4271->gpio_nreset = cs4271plat->gpio_nreset; if (gpio_is_valid(cs4271->gpio_nreset)) { - int ret; - ret = devm_gpio_request(dev, cs4271->gpio_nreset, "CS4271 Reset"); if (ret < 0) -- 2.1.4

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