March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC][PATCH 0/3] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 05 Mar '14

05 Mar '14

Xiubo Li (3): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: codec: remove the set_cache_io() entirely from CODEC ASoC probe. ASoC: core: remove the default regmap I/O setting up from soc_probe_codec() include/sound/soc.h | 3 +++ sound/soc/codecs/cq93vc.c | 6 ++++-- sound/soc/codecs/mc13783.c | 10 ++-------- sound/soc/codecs/wm5102.c | 6 ++---- sound/soc/codecs/wm5110.c | 6 ++---- sound/soc/codecs/wm8350.c | 6 ++++-- sound/soc/codecs/wm8400.c | 6 ++++-- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8997.c | 7 ++----- sound/soc/soc-core.c | 21 +++++++++++---------- 10 files changed, 36 insertions(+), 39 deletions(-) -- 1.8.4

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[alsa-devel] [PATCH] ASoC: ak4104: Remove superfluous codec->control_data initialization
by Lars-Peter Clausen 05 Mar '14

05 Mar '14

The driver uses automatic IO setup, which will also initialize the control_data field of the CODEC, no need to do it manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ak4104.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b4819dc..10adf25 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -174,8 +174,6 @@ static int ak4104_probe(struct snd_soc_codec *codec) struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ak4104->regmap; - /* set power-up and non-reset bits */ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, -- 1.8.0

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[alsa-devel] [PATCH] ASoC: io: Remove hw_read() operation
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> We now no longer have any users of hw_read() in the kernel so remove the code in order to prevent any new users being added. Users should be using regmap. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 - sound/soc/soc-io.c | 13 ------------- 2 files changed, 14 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 56c4c71..dc36331 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -706,7 +706,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ hw_write_t hw_write; - unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); void *reg_cache; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c3b6a0a..3c1de9c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -26,18 +26,6 @@ static int hw_write(struct snd_soc_codec *codec, unsigned int reg, return regmap_write(codec->control_data, reg, value); } -static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) -{ - int ret; - unsigned int val; - - ret = regmap_read(codec->control_data, reg, &val); - if (ret == 0) - return val; - else - return -1; -} - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -64,7 +52,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int ret; codec->write = hw_write; - codec->read = hw_read; switch (control) { case SND_SOC_REGMAP: -- 1.9.0

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[alsa-devel] [PATCH] ASoC: fsl: Don't select FIQ from Eukrea
by Mark Brown 05 Mar '14

05 Mar '14

From: Mark Brown <broonie(a)linaro.org> There is no point in using FIQ if DMA is available (it is selected) and selecting FIQ currently breaks the build on non-i.MX platforms. If FIQ is actually required the build will need to be restricted or have a select for the relevant FIQ code adding. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a7d6ab6..b244830 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -175,7 +175,6 @@ config SND_SOC_EUKREA_TLV320 || (OF && ARM) depends on I2C select SND_SOC_TLV320AIC23 - select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI select SND_SOC_FSL_SSI -- 1.9.0

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[alsa-devel] [PATCH] ASoC: rsnd: Fix warnings due to improper printk formats
by Laurent Pinchart 04 Mar '14

04 Mar '14

Use the %pap printk specifier to print resource_size_t variables. This fixes warnings on platforms where resource_size_t has a different size than int. Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas(a)ideasonboard.com> --- sound/soc/sh/rcar/gen.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index add088b..f505b1f 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -283,13 +283,13 @@ static int rsnd_gen2_probe(struct platform_device *pdev, return ret; dev_dbg(dev, "Gen2 device probed\n"); - dev_dbg(dev, "SRU : %08x => %p\n", scu_res->start, + dev_dbg(dev, "SRU : %pap => %p\n", &scu_res->start, gen->base[RSND_GEN2_SCU]); - dev_dbg(dev, "ADG : %08x => %p\n", adg_res->start, + dev_dbg(dev, "ADG : %pap => %p\n", &adg_res->start, gen->base[RSND_GEN2_ADG]); - dev_dbg(dev, "SSIU : %08x => %p\n", ssiu_res->start, + dev_dbg(dev, "SSIU : %pap => %p\n", &ssiu_res->start, gen->base[RSND_GEN2_SSIU]); - dev_dbg(dev, "SSI : %08x => %p\n", ssi_res->start, + dev_dbg(dev, "SSI : %pap => %p\n", &ssi_res->start, gen->base[RSND_GEN2_SSI]); return 0; -- Regards, Laurent Pinchart

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[alsa-devel] SNDRV_PCM_INFO_BATCH
by David Henningsson 04 Mar '14

04 Mar '14

So, PulseAudio removed timer scheduling for drivers with SNDRV_PCM_INFO_BATCH, and released 5.0. Meanwhile, USB drivers still have SNDRV_PCM_INFO_BATCH, so this leads to performance regressions for USB sound cards. The question remains - what does SNDRV_PCM_INFO_BATCH really mean? In the documentation, we have this: "int snd_pcm_hw_params_is_batch (const snd_pcm_hw_params_t *params) Check, if hardware does double buffering for data transfers for given configuration." What PulseAudio was looking for was more something like "what accuracy does the pointer callback give us". When we asked this question at the latest audio meeting, SNDRV_PCM_INFO_BATCH was given as the answer, i e, if this flag is set, pointer callback have no more accuracy than period boundaries. At least that's how I understood it. So, either the documentation for snd_pcm_hw_params_is_batch is correct, in which case PulseAudio should revert the tsched patch. Or the answer given at the audio meeting was correct, in which case we need to fix the USB driver to stop sending the SNDRV_PCM_INFO_BATCH flag. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [GIT PULL][PATCH 00/39] Enhancement of support for Firewire devices
by Takashi Sakamoto 04 Mar '14

04 Mar '14

This is a pull-request of my 39 patches to enhance support for Firewire devices. These patches extend firewire-lib, add new drivers (fireworks/bebob). These drivers can handle playback/capture of PCM samples/MIDI messages and can support 70-80 models totally. These drivers are designed not to interrupt FFADO (user-land driver project) functionality except for playbacking/capturing. These drivers basically give no way to control device's internal mixer. For this purpose, users need to use applications such like ffado-dbus-server/ffado-mixer, or develop software to execute AV/C commands and device specific commands. Updates since RFC v3: [alsa-devel] [RFC v3] [PATCH 00/52] Enhancement for support of firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-January/071820.ht… firewire-lib: - improve operation for oPCR - add fallbacks at transaction timeout for device's quirk fireworks: - add counter to maintain streams - use memdup_user() for hwdep interface bebob: - add counter to maintain streams - fix failure to handle bus reset ---------------------------------------------------------------- The following changes since commit dde9c38779c1f30253485a160facc211b20ca222: Merge branch 'for-linus' (2014-02-27 12:46:40 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to 0ed07c65fef2f80fcfc6223b0d689fca56fa6323: bebob: Add support for M-Audio special Firewire series (2014-02-28 11:30:52 +0900) ---------------------------------------------------------------- Takashi Sakamoto (39): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks firewire-lib: Add a fallback at RCODE_CANCELLED fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series bebob: Add support for M-Audio special Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 891 ++++++++++++++++------ sound/firewire/amdtp.h | 175 ++++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 450 +++++++++++ sound/firewire/bebob/bebob.h | 262 +++++++ sound/firewire/bebob/bebob_command.c | 354 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 922 +++++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 170 +++++ sound/firewire/bebob/bebob_pcm.c | 451 +++++++++++ sound/firewire/bebob/bebob_proc.c | 179 +++++ sound/firewire/bebob/bebob_stream.c | 869 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 229 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 160 +++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 324 ++++++++ sound/firewire/fireworks/fireworks.h | 223 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 294 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 179 +++++ sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 208 +++++ sound/firewire/fireworks/fireworks_stream.c | 337 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/lib.c | 6 +- sound/firewire/lib.h | 1 + sound/firewire/speakers.c | 93 +-- 37 files changed, 8356 insertions(+), 402 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

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[alsa-devel] [PATCH RFC 0/5] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 04 Mar '14

04 Mar '14

This patch set adds audio support for AM43xx-ePOS-EVM. I'll mail the dts and defconfig changes once these have been merged. Jyri Sarha (5): ASoC: tlv320aic31xx: Add basic codec driver implementation ASoC: tlv320aic31xx: Add codec driver to Makefile and Kconfig ASoC: tlv320aic31xx: Add DT binding document ASoC: davinci-evm: Add AM43xx-EPOS-EVM audio support ASoC: davinci: Add SND_AM43XX_SOC_EPOS_EVM build option .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/tlv320aic31xx.txt | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1360 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 265 ++++ sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 41 + 9 files changed, 1748 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

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[alsa-devel] [PATCH] Wrap hw_ptr to boundary in pcm_ioplug
by MONTANARO Luciano (MM) 04 Mar '14

04 Mar '14

The function snd_pcm_ioplug_hw_ptr_update() always increased the hw_ptr by delta, without wrapping it to the boundary. This would lead to problems when after many hours, the hw_ptr would overflow. --- src/pcm/pcm_ioplug.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a90c844..c1c3a98 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -60,7 +60,7 @@ static void snd_pcm_ioplug_hw_ptr_update(snd_pcm_t *pcm) delta = hw - io->last_hw; else delta = pcm->buffer_size + hw - io->last_hw; - io->data->hw_ptr += delta; + snd_pcm_mmap_hw_forward(io->data->pcm, delta); io->last_hw = hw; } else io->data->state = SNDRV_PCM_STATE_XRUN; -- 1.7.0.4 -- Luciano Montanaro EntryNAV - Audio Software Integrator Magneti Marelli Electronic Systems Viale Carlo Emanuele II 150, Venaria Reale (To) Tel. (+39) 011 6879765 VISITA IL NOSTRO NUOVO SITO WEB! - VISIT OUR NEW WEB SITE! www.magnetimarelli.com Confidential Notice: This message - including its attachments - may contain proprietary, confidential and/or legally protected information and is intended solely for the use of the designated addressee(s) above. If you are not the intended recipient be aware that any downloading, copying, disclosure, distribution or use of the contents of the above information is strictly prohibited. If you have received this communication by mistake, please forward the message back to the sender at the email address above, delete the message from all mailboxes and any other electronic storage medium and destroy all copies. Disclaimer Notice: Internet communications cannot be guaranteed to be safe or error-free. Therefore we do not assure that this message is complete or accurate and we do not accept liability for any errors or omissions in the contents of this message.

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[alsa-devel] Fix for busy loop at boundary wrapping with ioplug based plugins
by MONTANARO Luciano (MM) 04 Mar '14

04 Mar '14

Hello everybody, I have found an odd problem during long testing of our device, that suddenly starts misbehaving after several hours of use. What happened is: A client opens a device based on the jack plugin for reading, and keeps reading from it. After 10 hours 55 minutes, the snd_pcm_readi() functions starts to report data to be always available for read, and makes the reader use all the cpu it can. My patch will follow shortly, please tell me what you think of it. Best regards, Luciano -- Luciano Montanaro EntryNAV - Audio Software Integrator Magneti Marelli Electronic Systems Viale Carlo Emanuele II 150, Venaria Reale (To) Tel. (+39) 011 6879765 VISITA IL NOSTRO NUOVO SITO WEB! - VISIT OUR NEW WEB SITE! www.magnetimarelli.com Confidential Notice: This message - including its attachments - may contain proprietary, confidential and/or legally protected information and is intended solely for the use of the designated addressee(s) above. If you are not the intended recipient be aware that any downloading, copying, disclosure, distribution or use of the contents of the above information is strictly prohibited. If you have received this communication by mistake, please forward the message back to the sender at the email address above, delete the message from all mailboxes and any other electronic storage medium and destroy all copies. Disclaimer Notice: Internet communications cannot be guaranteed to be safe or error-free. Therefore we do not assure that this message is complete or accurate and we do not accept liability for any errors or omissions in the contents of this message.

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