April 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v6 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 31 May '16

31 May '16

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.6-rc1 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v6 Changes from v5: - Fix code style, e.g. change "if" statements to "switch" statements - Fix SPDIF output enable register field - Set BUFFER_PAIR_ENABLE for both mono, stereo and TDM modes - Reflect PCM bit formats for SPDIF, SSP and TDM as supported by hardware Changes from v4: - Fix power suspend function and add power resume function - Move clock initialization code from audio driver to clock framework Changes from v3: - Fix the subject lines to match the style for the subsystem Changes from v2: - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using standard kernel trace infrastructure. - Fix interrupt handler. Acknowledge only those interrupts that are handledby ISR. - Modify configure_vco() and the pll_macro_entry() struct to make it better readable. The functionality did not change. - Remove casts on macros - Remove surround sound channel grouping from the driver. Changes from v1: - Address code review comments. Fix print format of type size_t and pointer. Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 67 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1529 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ 6 files changed, 2610 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

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[alsa-devel] [PATCH v4 0/7] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 19 May '16

19 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 4 of the patch: * minor correction in the binding example ("interrupt names") * re-added index note to the "mpu" property.

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[alsa-devel] [PATCH] alsa-lib:Make snd_atomic_write_* truly atomic
by Andreas x Larsson 17 May '16

17 May '16

Under some usecases a race condition appears inside the snd_atomic_write_* functions. The 'begin' and 'end' variables are updated with the ++ operator which is not atomic but needs to be. This can be achieved with the gcc atomic_* built-ins. Combined with __ATOMIC_SEQ_CST as the memory model, memory barriers are introduced so those can also be removed from the code. Signed-off-by: Andreas x Larsson <andrelan(a)axis.com> Signed-off-by: Arvid Nihlgård Lindell <arvidnl(a)axis.com> --- alsa-lib/include/iatomic.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/alsa-lib/include/iatomic.h b/alsa-lib/include/iatomic.h index acdd3e2..7fafe20 100644 --- a/alsa-lib/include/iatomic.h +++ b/alsa-lib/include/iatomic.h @@ -140,14 +140,12 @@ static __inline__ void snd_atomic_write_init(snd_atomic_write_t *w) static __inline__ void snd_atomic_write_begin(snd_atomic_write_t *w) { - w->begin++; - wmb(); + __atomic_add_fetch(&w->begin, 1,__ATOMIC_SEQ_CST); } static __inline__ void snd_atomic_write_end(snd_atomic_write_t *w) { - wmb(); - w->end++; + __atomic_add_fetch(&w->end, 1, __ATOMIC_SEQ_CST); } static __inline__ void snd_atomic_read_init(snd_atomic_read_t *r, snd_atomic_write_t *w) -- 2.1.4

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[alsa-devel] ADAU1761 default register value problem
by Ricard Wanderlof 16 May '16

16 May '16

Hi Lars, A while ago I posted a question to the list regarding the default values set up for the ADAU1761 (and similar CODECs), but I didn't get a response. I really need to try and figure out a way forward here though so I'll bring this up again. The problem I have is that the driver assumes that the CODEC has been initialized with the default register settings when the system boots. However, if the system has not been power cycled but just rebooted, this is not necessarily the case, as the CODEC has no external reset pin but derives its reset signal internally from the power level. Furthermore, if the system upon reboot tries to set a register value to one of the default settings (for example as the result of a hw_params() call), then the regcache mechanism will disregard the write as it assumes the default value is already set, which is not necessarily the case. As I see it the solution to this problem is simply not to assume anything about the register settings, so that the first write to a given register always takes effect. I can work on a patch to accomplish this, however, since so much work apparently has been laid down in getting the default values into the driver in the first place, I'm assuming there's something more to the story that I'm missing, so I'd like to get a better grip on this before I dive in. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v2 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 13 May '16

13 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 68 +++++++++++++++++++++++++++++++++++++--------- sound/soc/soc-pcm.c | 3 +- 3 files changed, 61 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..fe5bc93 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..075f0d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,13 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); + if (custom_stop_condition) { + con = custom_stop_condition(widget, dir); + if (con >= 0) + return con; + con = 0; + } + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1117,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1130,41 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1172,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1211,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1310,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1970,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

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[alsa-devel] [PATCH v3 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 13 May '16

13 May '16

Updated driver for TI's TAS5720L/M digital audio amplifiers incorporating previous feedback (thanks Mark Brown). Biggest change is that the driver is no longer using interrupts for codec fault reporting, but instead uses a polling-based approach to check for (and clear) codec fault conditions every 200ms. This is much more graceful and actually lower overhead than trying to keep the actual TAS5720 fault interrupts (that come in every 300us!) under control which is what v2 of the driver was doing. Because of the removal of interrupts the DT doc had to be updated as well, so I did not carry forward Rob Herring's earlier Acked-by because of said change. -- Andreas Dannenberg Texas Instruments Inc Changes since v2: - Switched fault handling from using interrupts to polling mode - Remove interrupt related description from DT bindings doc - Remove unlikely() to simplify an expression - Remove unnesseary tas5720_set_dai_sysclk() function stub Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 26 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 638 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 766 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

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[alsa-devel] [PATCH v2 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. ;) Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 63 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/5] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (5): ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 69 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 60 insertions(+), 9 deletions(-) -- 1.9.1

4 17

[alsa-devel] [PATCH v3 0/7] ARM/ASoC: OMAP3: Fix McBSP2/3 sidetone support
by Peter Ujfalusi 13 May '16

13 May '16

Hi, Based on the ongoing discussion on v2 (ARM: OMAP3: Fix McBSP2/3 hwmod setup for sidetone) I have dropped the removal of the sidetone hwmod and only corrected it. The series now includes all related changes needed to have correct sidetone support whenever we boot in legacy or in DT mode. This time the ASoC part have dependency on earlier patches so they can not be applied separately. Regards, Peter --- Peter Ujfalusi (7): ARM: OMAP3: hwmod data: Fix McBSP2/3 sidetone data ARM: dts: omap3: Add clocks to McBSP nodes ARM: OMAP3: McBSP: New callback for McBSP2/3 ICLK idle configuration ARM: OMAP3: pdata-quirks: Add support for McBSP2/3 sidetone handling ASoc: omap-mcbsp: Rename omap_mcbsp_sysfs_remove() to omap_mcbsp_cleanup() ASoC: omap-mcbsp: sidetone: Use the new callback for iclk handling ARM: OMAP2+: McBSP: Remove the old iclk allow/deny idle code arch/arm/boot/dts/omap3.dtsi | 10 ++++++++++ arch/arm/mach-omap2/mcbsp.c | 31 ++++++++++++++++------------- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 24 ++++------------------ arch/arm/mach-omap2/pdata-quirks.c | 18 +++++++++++++++++ include/linux/platform_data/asoc-ti-mcbsp.h | 4 +++- sound/soc/omap/mcbsp.c | 21 +++++++++++++------ sound/soc/omap/mcbsp.h | 3 ++- sound/soc/omap/omap-mcbsp.c | 5 ++++- 8 files changed, 73 insertions(+), 43 deletions(-) -- 2.8.1

3 13

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: note buggy I2S support when the codec masters LRCLK
by Peter Rosin 13 May '16

13 May '16

The only difference in register settings between I2S and DSP mode A, for the SND_SOC_DAIFMT_CBM_CFM case, is the start condition (which is SSC_START_FALLING_RF for I2S and SSC_START_RISING_RF for DSP mode A). As this is not the only difference between the two formats, it is an indicator that something is seriously wrong. And it is the I2S support that is broken. While the start condition is correct for the left channel word in the I2S case, it is not correct that the right channel word follows immediately after the left channel word. The start of the right channel word should be triggered by a rising edge on LRCLK in the I2S case, something which simply does not happen. The correct thing to do would be to simply remove the alleged support for I2S/CBM_CFM, but that is likely to cause regressions. So, just make a note about the buggy I2S support in the source. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 4ab1e2013238..3108c6e20ad5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -558,7 +558,15 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* I2S format, CODEC supplies BCLK and LRC clocks. */ + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * NOTE! This is not really I2S, as the SSC does not allow + * for any extra empty BCLK cycles after the left channel + * word. I.e. the SSC is only looking at the falling edge + * of LRCLK and ignores the rising edge. + * In reality, this is DSP mode A with inverted LRCLK. + * Use DSP mode A instead, if your codec supports it. + */ rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_FALLING_RF) -- 2.1.4

2 3

[alsa-devel] [PATCH] ASoC: rsnd: count .probe/.remove for rsnd_mod_call()
by Kuninori Morimoto 13 May '16

13 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_mod_call is counting its calling count to avoid unbalanced function pair calling for error cases (ex init <-> quit). SSI parent is now controlled as "mod" on current rsnd driver. Because of this reason, SSI .remove function will be called twice if it was used as SSI parent when user tried unbind. But probe/remove pair were not counted. This patch counts probe/remove functions to avoid it. Special thans Hiep Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsnd.h | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc89a67..a8f61d7 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -276,8 +276,9 @@ struct rsnd_mod { /* * status * - * 0xH0000CB0 + * 0xH0000CBA * + * A 0: probe 1: remove * B 0: init 1: quit * C 0: start 1: stop * @@ -287,19 +288,19 @@ struct rsnd_mod { * H 0: fallback * H 0: hw_params */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 #define __rsnd_mod_shift_stop 8 -#define __rsnd_mod_shift_probe 28 /* always called */ -#define __rsnd_mod_shift_remove 28 /* always called */ #define __rsnd_mod_shift_irq 28 /* always called */ #define __rsnd_mod_shift_pcm_new 28 /* always called */ #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ -#define __rsnd_mod_add_probe 0 -#define __rsnd_mod_add_remove 0 +#define __rsnd_mod_add_probe 1 +#define __rsnd_mod_add_remove -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -310,7 +311,7 @@ struct rsnd_mod { #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_call_probe 0 -#define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_remove 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 1.9.1

3 3

[alsa-devel] [PATCH 1/2] ASoC: nau8825: non-clock jack detection for power saving at standby
by John Hsu 12 May '16

12 May '16

The driver changes jack type detection interruption to non-clock archi- tecture for less 1mW power saving. The architecture is called manual mode jack detection. It has no hardware debounce, no jack type detection, but only detecting jack insertion. After jack insertion, the driver will switch to auto mode jack detection with internal clock which can detect microphone, jack type and do hardware debounce. The manual architecture has these main changes including codec initiation, interruption, clock control, and power management. When codec initiation or system resume, the clock is closed as jack insertion detection at man- ual mode, and bypass debounce circuit. These configurations move to resume setup function when setup bias level after resume. When jack insertion detection happens, the manual mode turns off and make configuration about jack type detection interruption at auto mode in auto irq setup function which can detect microphone and jack type. The inter- ruption will switch to manual mode again with clock free until jack ejec- tion happens. The system clock configuration adds clock disable option which can disable internal VCO clock. Before the system clock change, there is an restric- tion added to make sure clock disabled and not config any clock when no headset connected. In power management, we involve the solution about races and jack detec- tion in resume from Ben Zhang in the following patch and list his comment. [PATCH] ASoC: nau8825: Fix jack detection across suspend "Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power." "Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly." Change SAR widget to supply type which can prevent the codec keeping at SND_SOC_BIAS_ON during suspend. The codec suspend function can just invoke normally. Before the system suspends, the driver turns off all interruptions. Keep the interruption quiet before resume setup completes. The ADC channel will be disabled which is needed for interruptions at audo mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 281 +++++++++++++++++++++++++++++++++++---------- sound/soc/codecs/nau8825.h | 9 +- 2 files changed, 226 insertions(+), 64 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f35019a..9386624 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -30,10 +30,16 @@ #include "nau8825.h" + +#define NUVOTON_CODEC_DAI "nau8825-hifi" + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 +static int nau8825_configure_sysclk(struct nau8825 *nau8825, + int clk_id, unsigned int freq); + struct nau8825_fll { int mclk_src; int ratio; @@ -368,9 +374,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -614,8 +623,10 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. + */ return 0; } @@ -642,6 +653,22 @@ static void nau8825_restart_jack_detection(struct regmap *regmap) NAU8825_JACK_DET_RESTART, 0); } +static void nau8825_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8825_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + } +} + static void nau8825_eject_jack(struct nau8825 *nau8825) { struct snd_soc_dapm_context *dapm = nau8825->dapm; @@ -656,6 +683,67 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Mask all interruptions except jack insertion interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0xfffe); + + /* Enable insertion interruption only and bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); + + /* Close clock for jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); +} + +/* Enable audo mode interruptions with internal clock. */ +static void nau8825_setup_auto_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable headset jack type detection complete interruption and + * jack ejection interruption. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interruptions */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, 0); + + /* Unmask all interruptions */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + + /* Restart the jack detection process at auto mode */ + nau8825_restart_jack_detection(regmap); } static int nau8825_button_decode(int value) @@ -753,7 +841,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to clear interruption\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -789,6 +880,26 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == + NAU8825_JACK_INSERTION_DETECTED) { + /* One more step to check GPIO status directly. Thus, the + * driver can confirm the real insertion interruption because + * the intrruption at manual mode has bypassed debounce + * circuit which can get rid of unstable status. + */ + if (nau8825_is_jack_inserted(regmap)) { + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_INSERT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_INSERT_EN, NAU8825_IRQ_INSERT_EN); + /* Enable interruption for jack type detection at audo + * mode which can detect microphone and jack type. + */ + nau8825_setup_auto_irq(nau8825); + } } if (!clear_irq) @@ -938,8 +1049,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) } static const struct regmap_config nau8825_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = NAU8825_REG_DATA_LEN, + .reg_bits = NAU8825_REG_ADDR_LEN, .max_register = NAU8825_REG_MAX, .readable_reg = nau8825_readable_reg, @@ -958,11 +1069,10 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* Unmask interruptions. Handler uses dapm object so we can enable - * interruptions only after dapm is fully initialized. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. */ - regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); - nau8825_restart_jack_detection(nau8825->regmap); return 0; } @@ -1134,7 +1244,26 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, struct regmap *regmap = nau8825->regmap; int ret; + if (!nau8825_is_jack_inserted(nau8825->regmap) && + clk_id != NAU8825_CLK_DIS) { + /* For power saving less 1mW, the jack type detection inter- + * ruption changes to non-clock architecture. Therefore, the + * clock should be disabled and not allowed to config any clock + * when no headset connected. + */ + dev_warn(nau8825->dev, "Souldn't enable any clock when no headset connected\n"); + return 0; + } + switch (clk_id) { + case NAU8825_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, 0); + + break; case NAU8825_CLK_MCLK: regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); @@ -1209,6 +1338,30 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable insertion interruption only and bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, 0); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1238,11 +1391,22 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + + /* Setup codec configuration after resume */ + nau8825_resume_setup(nau8825); } break; case SND_SOC_BIAS_OFF: + /* Turn off all interruptions before system shutdown. Keep the + * interruption quiet before resume setup completes. + */ + regmap_write(nau8825->regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff); + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); @@ -1252,12 +1416,50 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return 0; } +#ifdef CONFIG_PM_SLEEP +static int nau8825_codec_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(nau8825->regmap, true); + + return 0; +} + +static int nau8825_codec_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(nau8825->regmap, false); + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + /* Check the jack plug status directly. If the headset is un- + * plugged during S3 when the chip has no power, there will + * be no jack detection irq event after restarting jack detec- + * tion, because the chip just thinks no headset has ever + * been plugged in. + */ + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + enable_irq(nau8825->irq); + + return 0; +} +#else +#define nau8825_codec_suspend NULL +#define nau8825_codec_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_codec_suspend, + .resume = nau8825_codec_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1351,29 +1553,13 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Because resume and initiation all need + * to enable the mechanism, move these configuration to resume setup + * function. */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, @@ -1442,36 +1628,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1498,7 +1654,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..1100385 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -93,6 +93,9 @@ #define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 #define NAU8825_REG_GENERAL_STATUS 0x82 #define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8825_REG_ADDR_LEN 16 +#define NAU8825_REG_DATA_LEN 16 /* ENA_CTRL (0x1) */ #define NAU8825_ENABLE_DACR_SFT 10 @@ -145,6 +148,7 @@ /* JACK_DET_CTRL (0xd) */ #define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_DET_DB_BYPASS (1 << 8) #define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 #define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) #define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 @@ -157,6 +161,7 @@ #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) +#define NAU8825_IRQ_INSERT_EN (1 << 0) /* IRQ_STATUS (0x10) */ #define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) @@ -177,6 +182,7 @@ #define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) #define NAU8825_IRQ_EJECT_DIS (1 << 2) +#define NAU8825_IRQ_INSERT_DIS (1 << 0) /* SAR_CTRL (0x13) */ #define NAU8825_SAR_ADC_EN_SFT 12 @@ -333,7 +339,8 @@ /* System Clock Source */ enum { - NAU8825_CLK_MCLK = 0, + NAU8825_CLK_DIS = 0, + NAU8825_CLK_MCLK, NAU8825_CLK_INTERNAL, NAU8825_CLK_FLL_MCLK, NAU8825_CLK_FLL_BLK, -- 2.6.4

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[alsa-devel] [PATCH] ASoC: pcm: allow changing the playback/capture rates for symmetric links
by Peter Rosin 09 May '16

09 May '16

The below program fails on a dai link with symmetric rates without this patch. The patch makes it work. #include <sys/soundcard.h> #include <unistd.h> #include <fcntl.h> #include <stropts.h> #include <stdio.h> int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22050; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4fa86..1e876ff23524 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -237,6 +237,14 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int rate, channels, sample_bits, symmetry, i; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (!cpu_dai->capture_active) + return 0; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!cpu_dai->playback_active) + return 0; + } + rate = params_rate(params); channels = params_channels(params); sample_bits = snd_pcm_format_physical_width(params_format(params)); -- 2.1.4

3 12

[alsa-devel] [PATCH 0/2] wm5110: fixes to ANC clock control
by Richard Fitzgerald 09 May '16

09 May '16

The register for ANC clock control has a combination of volatile and non-readback bits and these two patches fix the way we write to this register. Both patches must be taken together to ensure correct operation. Richard Fitzgerald (2): mfd: wm5110: ARIZONA_CLOCK_CONTROL should be volatile ASoC: arizona: No need to update_bits when writing AEC clock control drivers/mfd/wm5110-tables.c | 1 + sound/soc/codecs/arizona.c | 3 +-- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: Add DSP firmware manifest parsing
by Vinod Koul 09 May '16

09 May '16

From: Shreyas NC <shreyas.nc(a)intel.com> Module params like module_id and loadable flag can be changed in the DSP Firmware. These are kept in the firmware manifest and driver should read these values from this manifest. So, add support to parse the DSP firmware manifest and read these module params. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Changes in v3: add bound check for firmware file sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 118 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 107 +++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 4 ++ sound/soc/intel/skylake/skl-sst.c | 11 ++- sound/soc/intel/skylake/skl-topology.c | 5 ++ 6 files changed, 245 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index c28f5d0e1d99..e2fb5b763b21 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o + skl-sst.o skl-dsp-parse.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-dsp-parse.c b/sound/soc/intel/skylake/skl-dsp-parse.c new file mode 100644 index 000000000000..a3cd5d7ab246 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.c @@ -0,0 +1,118 @@ +/* + * skl-dsp-parse.c - Implements DSP firmware parsing + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-tplg-interface.h" +#include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" + +/* + * Get the module id for the module by checking + * the table for the UUID for the module + */ +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config) +{ + int i, num; + struct uuid_tbl *tbl; + uuid_le *uuid_mod; + + tbl = ctx->tbl; + num = ctx->num_modules; + uuid_mod = (uuid_le *)uuid; + + for (i = 0; i < num; i++) { + if (uuid_le_cmp(*uuid_mod, tbl[i].uuid) == 0) { + dfw_config->module_id = tbl[i].module_id; + dfw_config->is_loadable = tbl[i].is_loadable; + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_skl_get_module_info); + +/* + * Parse the firmware binary to get the UUID, module id + * and loadable flags + */ +int snd_skl_parse_fw_bin(struct sst_dsp *ctx) +{ + struct adsp_fw_hdr *adsp_hdr; + struct adsp_module_entry *mod_entry; + int i, num_entry; + uuid_le *uuid_bin; + const char *buf; + struct skl_sst *skl = ctx->thread_context; + struct uuid_tbl *tbl; + + /* Get the FW pointer to derive ADSP header */ + buf = ctx->fw->data; + + adsp_hdr = (struct adsp_fw_hdr *)(buf + SKL_ADSP_FW_BIN_HDR_OFFSET); + + mod_entry = (struct adsp_module_entry *) + (buf + SKL_ADSP_FW_BIN_HDR_OFFSET + adsp_hdr->header_len); + + num_entry = adsp_hdr->num_module_entries; + + tbl = devm_kzalloc(ctx->dev, + num_entry * sizeof(struct uuid_tbl), GFP_KERNEL); + + if (!tbl) + return -ENOMEM; + + /* + * Read the UUID(GUID) from FW Manifest. + * The 16 byte UUID is of the format: + * XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXX + * Populate the UUID table to store module_id + * and loadable flags for the module. + */ + + for (i = 0; i < num_entry; i++, mod_entry++) { + + /* + * we check if current pointer is larger than file size from + * base value to check excceding the file while parsing + */ + if ((const char *)mod_entry >= buf + ctx->fw->size) { + dev_err(ctx->dev, + "Exceeds file bound: Entry %d Ptr %p\n", + i, mod_entry); + + return -EIO; + } + + uuid_bin = (uuid_le *)mod_entry->uuid.id; + memcpy(&tbl[i].uuid, uuid_bin, sizeof(tbl[i].uuid)); + + tbl[i].module_id = i; + tbl[i].is_loadable = mod_entry->type.load_type; + } + + skl->tbl = tbl; + skl->num_modules = num_entry; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_skl_parse_fw_bin); diff --git a/sound/soc/intel/skylake/skl-dsp-parse.h b/sound/soc/intel/skylake/skl-dsp-parse.h new file mode 100644 index 000000000000..cfdb24c45035 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.h @@ -0,0 +1,107 @@ +/* + * skl-dsp-parse.h + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <asm/types.h> +#include <linux/types.h> +#include <linux/uuid.h> +#include "skl-tplg-interface.h" +#include "../common/sst-dsp-priv.h" + +#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 +#define UUID_STR_SIZE 37 +#define DEFAULT_HASH_SHA256_LEN 32 + +struct skl_dfw_module_mod { + char name[100]; + struct skl_dfw_module skl_dfw_mod; +}; + +struct UUID { + u8 id[16]; +}; + +union seg_flags { + u32 ul; + struct { + u32 contents : 1; + u32 alloc : 1; + u32 load : 1; + u32 read_only : 1; + u32 code : 1; + u32 data : 1; + u32 _rsvd0 : 2; + u32 type : 4; + u32 _rsvd1 : 4; + u32 length : 16; + } r; +} __packed; + +struct segment_desc { + union seg_flags flags; + u32 v_base_addr; + u32 file_offset; +}; + +struct module_type { + u32 load_type : 4; + u32 auto_start : 1; + u32 domain_ll : 1; + u32 domain_dp : 1; + u32 rsvd_ : 25; +} __packed; + +struct adsp_module_entry { + u32 struct_id; + u8 name[8]; + struct UUID uuid; + struct module_type type; + u8 hash1[DEFAULT_HASH_SHA256_LEN]; + u32 entry_point; + u16 cfg_offset; + u16 cfg_count; + u32 affinity_mask; + u16 instance_max_count; + u16 instance_bss_size; + struct segment_desc segments[3]; +} __packed; + +struct adsp_fw_hdr { + u32 header_id; + u32 header_len; + u8 name[8]; + u32 preload_page_count; + u32 fw_image_flags; + u32 feature_mask; + u16 major_version; + u16 minor_version; + u16 hotfix_version; + u16 build_version; + u32 num_module_entries; + u32 hw_buf_base_addr; + u32 hw_buf_length; + u32 load_offset; +} __packed; + +struct uuid_tbl { + uuid_le uuid; + int module_id; + int is_loadable; +}; + +int snd_skl_parse_fw_bin(struct sst_dsp *ctx); +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index d59d1ba62a43..67d56be8cb77 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -60,6 +60,10 @@ struct skl_sst { void (*enable_miscbdcge)(struct device *dev, bool enable); /*Is CGCTL.MISCBDCGE disabled*/ bool miscbdcg_disabled; + + /* Populate module information */ + struct uuid_tbl *tbl; + int num_modules; }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index bec4a7c486fd..df7964fd09c3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -25,6 +25,7 @@ #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" #include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" #define SKL_BASEFW_TIMEOUT 300 #define SKL_INIT_TIMEOUT 1000 @@ -70,7 +71,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { - int ret = 0, i; + int ret = 0, i, err; struct skl_sst *skl = ctx->thread_context; u32 reg; @@ -84,6 +85,14 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) skl_dsp_disable_core(ctx); return -EIO; } + + err = snd_skl_parse_fw_bin(ctx); + if (err < 0) { + dev_err(ctx->dev, + "Firmware parsing err: %d\n", ret); + release_firmware(ctx->fw); + return err; + } } ret = skl_dsp_boot(ctx); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a1f4478fabcb..f690b016203c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -28,6 +28,7 @@ #include "skl-tplg-interface.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "skl-dsp-parse.h" #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) @@ -1566,6 +1567,10 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, w->priv = mconfig; memcpy(&mconfig->guid, &dfw_config->uuid, 16); + ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config); + if (ret < 0) + return ret; + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; -- 1.9.1

2 2

[alsa-devel] [PATCH v2 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 09 May '16

09 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. History: v2: Fix memory leak on realloc failure. The original obj pointer is reserved and so it can be freed at last by tplg_elem_free(). Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 26 +++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 133 insertions(+), 58 deletions(-) -- 2.5.0

2 7

[alsa-devel] [PATCH] ALSA: dice: add support for M-Audio Profire 610 and perhaps 2626
by Takashi Sakamoto 09 May '16

09 May '16

M-Audio Profire 610 has an unexpected value in version field of its config ROM, thus ALSA dice driver is not assigned to the model due to a mismatch of modalias. This commit adds an entry to support the model. I expect the entry is also for Profire 2626. I note that Profire 610 uses TCD2220 (so-called Dice Jr.), and supports a part of Extended Application Protocol (EAP). $ cd linux-firewire-utils/src $ ./crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ------------------------------------------------------------ 400 04047689 bus_info_length 4, crc_length 4, crc 30345 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 000d6c04 company_id 000d6c | 410 04400002 device_id 0404400002 | EUI-64 000d6c0404400002 root directory ------------------------------------------------------------ 414 000695fe directory_length 6, crc 38398 418 03000d6c vendor 41c 8100000a --> descriptor leaf at 444 420 17000011 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ------------------------------------------------------------ 430 0004fb14 directory_length 4, crc 64276 434 12000d6c specifier id 438 130100d1 version 43c 17000011 model 440 8100000c --> descriptor leaf at 470 descriptor leaf at 444 ------------------------------------------------------------ 444 0004b8e4 leaf_length 4, crc 47332 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4d2d4175 "M-Au" 454 64696f00 "dio" descriptor leaf at 458 ------------------------------------------------------------ 458 00053128 leaf_length 5, crc 12584 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 50726f46 "ProF" 468 69726520 "ire " 46c 36313000 "610" descriptor leaf at 470 ------------------------------------------------------------ 470 00053128 leaf_length 5, crc 12584 474 00000000 textual descriptor 478 00000000 minimal ASCII 47c 50726f46 "ProF" 480 69726520 "ire " 484 36313000 "610" $ cat /proc/asound/card1/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc0:000100000000 notification: 00000040 nick name: FW610 clock select: internal 48000 enable: 1 status: locked 48000 ext status: 00000040 sample rate: 48000 version: 1.0.4.0 clock caps: 32000 44100 48000 88200 96000 176400 192000 aes1 aes4 aes adat tdif wc arx1 arx2 internal clock source names: SPDIF\AES34\AES56\TOS\AES_ANY\ADAT\ADAT_AUX\Word Clock\Unused\Unused\Unused\Unused\Internal\\ ... Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96fe68f4..25e9f77 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -317,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); -- 2.7.4

2 4

[alsa-devel] [PATCH 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 08 May '16

08 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 24 ++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 131 insertions(+), 58 deletions(-) -- 2.5.0

5 11

[alsa-devel] sound: deadlock involving snd_hrtimer_callback
by Dmitry Vyukov 08 May '16

08 May '16

Hi Takashi, I've incorporated your hrtimer fixes (but also updated to ddce192106e4f984123884f8e878f66ace94b573) and now I am seeing lots of the following deadlock messages: [ INFO: possible circular locking dependency detected ] 4.6.0-rc4+ #351 Not tainted ------------------------------------------------------- swapper/0/0 is trying to acquire lock: (&(&timer->lock)->rlock){-.-...}, at: [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 but task is already holding lock: (&(&stime->lock)->rlock){-.....}, at: [<ffffffff85383d3f>] snd_hrtimer_callback+0x4f/0x2b0 sound/core/hrtimer.c:54 which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&(&stime->lock)->rlock){-.....}: [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff85383baa>] snd_hrtimer_start+0x4a/0xf0 sound/core/hrtimer.c:112 [<ffffffff85379c74>] snd_timer_start1+0x2b4/0x5a0 sound/core/timer.c:457 [<ffffffff8537d33d>] snd_timer_start+0x5d/0xa0 sound/core/timer.c:571 [< inline >] seq_timer_start sound/core/seq/seq_timer.c:393 [<ffffffff853fc260>] snd_seq_timer_start+0x1a0/0x2b0 sound/core/seq/seq_timer.c:405 [< inline >] snd_seq_queue_process_event sound/core/seq/seq_queue.c:687 [<ffffffff853f6f14>] snd_seq_control_queue+0x304/0x8b0 sound/core/seq/seq_queue.c:748 [<ffffffff853fcbb5>] event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 [<ffffffff853ef7d4>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:636 [<ffffffff853efc38>] snd_seq_deliver_event+0x118/0x800 sound/core/seq/seq_clientmgr.c:833 [<ffffffff853f1026>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2418 [<ffffffff85405c2b>] send_timer_event.isra.0+0x10b/0x150 sound/core/seq/oss/seq_oss_timer.c:153 [<ffffffff8540615a>] snd_seq_oss_timer_start+0x1ca/0x310 sound/core/seq/oss/seq_oss_timer.c:174 [< inline >] old_event sound/core/seq/oss/seq_oss_event.c:125 [<ffffffff854095ff>] snd_seq_oss_process_event+0xa1f/0x2ce0 sound/core/seq/oss/seq_oss_event.c:100 [< inline >] insert_queue sound/core/seq/oss/seq_oss_rw.c:179 [<ffffffff8540c201>] snd_seq_oss_write+0x321/0x810 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff85403f99>] odev_write+0x59/0xa0 sound/core/seq/oss/seq_oss.c:177 [<ffffffff817f3fc3>] __vfs_write+0x113/0x4b0 fs/read_write.c:529 [<ffffffff817f5a97>] vfs_write+0x167/0x4a0 fs/read_write.c:578 [< inline >] SYSC_write fs/read_write.c:625 [<ffffffff817f98d1>] SyS_write+0x111/0x220 fs/read_write.c:617 [<ffffffff867c0440>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 -> #0 (&(&timer->lock)->rlock){-.-...}: [< inline >] check_prev_add kernel/locking/lockdep.c:1823 [< inline >] check_prevs_add kernel/locking/lockdep.c:1933 [< inline >] validate_chain kernel/locking/lockdep.c:2238 [<ffffffff81474035>] __lock_acquire+0x3625/0x4d00 kernel/locking/lockdep.c:3298 [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [< inline >] arch_safe_halt ./arch/x86/include/asm/paravirt.h:118 [<ffffffff8120f402>] default_idle+0x52/0x370 arch/x86/kernel/process.c:307 [<ffffffff81210bba>] arch_cpu_idle+0xa/0x10 arch/x86/kernel/process.c:298 [<ffffffff81454c28>] default_idle_call+0x48/0xa0 kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call kernel/sched/idle.c:151 [< inline >] cpu_idle_loop kernel/sched/idle.c:242 [<ffffffff8145520f>] cpu_startup_entry+0x58f/0x7b0 kernel/sched/idle.c:291 [<ffffffff8679ac6d>] rest_init+0x18d/0x1a0 init/main.c:408 [<ffffffff88ce7759>] start_kernel+0x63a/0x660 init/main.c:661 [<ffffffff88ce6364>] x86_64_start_reservations+0x38/0x3a arch/x86/kernel/head64.c:195 [<ffffffff88ce64be>] x86_64_start_kernel+0x158/0x167 arch/x86/kernel/head64.c:176 other info that might help us debug this: Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&(&stime->lock)->rlock); lock(&(&timer->lock)->rlock); lock(&(&stime->lock)->rlock); lock(&(&timer->lock)->rlock); *** DEADLOCK *** 1 lock held by swapper/0/0: #0: (&(&stime->lock)->rlock){-.....}, at: [<ffffffff85383d3f>] snd_hrtimer_callback+0x4f/0x2b0 sound/core/hrtimer.c:54 stack backtrace: CPU: 0 PID: 0 Comm: swapper/0 Not tainted 4.6.0-rc4+ #351 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88003ec07a50 ffffffff82c8f92f ffffffff00000000 fffffbfff0fd64b8 ffffffff8997bad0 ffffffff8997bad0 ffffffff8998aa70 ffffffff87e27510 ffffffff87e26d00 ffff88003ec07aa0 ffffffff8146cf58 Call Trace: <IRQ> [< inline >] __dump_stack lib/dump_stack.c:15 <IRQ> [<ffffffff82c8f92f>] dump_stack+0x12e/0x18f lib/dump_stack.c:51 [<ffffffff8146cf58>] print_circular_bug+0x288/0x340 kernel/locking/lockdep.c:1196 [< inline >] check_prev_add kernel/locking/lockdep.c:1823 [< inline >] check_prevs_add kernel/locking/lockdep.c:1933 [< inline >] validate_chain kernel/locking/lockdep.c:2238 [<ffffffff81474035>] __lock_acquire+0x3625/0x4d00 kernel/locking/lockdep.c:3298 [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [< inline >] arch_safe_halt ./arch/x86/include/asm/paravirt.h:118 [<ffffffff8120f402>] default_idle+0x52/0x370 arch/x86/kernel/process.c:307 [<ffffffff81210bba>] arch_cpu_idle+0xa/0x10 arch/x86/kernel/process.c:298 [<ffffffff81454c28>] default_idle_call+0x48/0xa0 kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call kernel/sched/idle.c:151 [< inline >] cpu_idle_loop kernel/sched/idle.c:242 [<ffffffff8145520f>] cpu_startup_entry+0x58f/0x7b0 kernel/sched/idle.c:291 [<ffffffff8679ac6d>] rest_init+0x18d/0x1a0 init/main.c:408 [<ffffffff88ce7759>] start_kernel+0x63a/0x660 init/main.c:661 [<ffffffff88ce6364>] x86_64_start_reservations+0x38/0x3a arch/x86/kernel/head64.c:195 [<ffffffff88ce64be>] x86_64_start_kernel+0x158/0x167 arch/x86/kernel/head64.c:176 They are also followed by rcu stalls and actual spinlock lockups: BUG: spinlock lockup suspected on CPU#3, syz-fuzzer/7040 lock: 0xffff880032ae2bd8, .magic: dead4ead, .owner: syz-executor/11206, .owner_cpu: 0 CPU: 3 PID: 7040 Comm: syz-fuzzer Not tainted 4.6.0-rc4+ #351 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88006d507c40 ffffffff82c8f92f ffffffff2c963698 fffffbfff0fd64b8 ffff880032ae2bd8 ffff88002c963040 ffff880061238040 ffff880032ae2be0 000000009a9d0630 ffff88006d507c78 ffffffff8147f58d Call Trace: <IRQ> [< inline >] __dump_stack lib/dump_stack.c:15 <IRQ> [<ffffffff82c8f92f>] dump_stack+0x12e/0x18f lib/dump_stack.c:51 [<ffffffff8147f58d>] spin_dump+0x14d/0x280 kernel/locking/spinlock_debug.c:67 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:117 [<ffffffff8147f8ad>] do_raw_spin_lock+0x15d/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff867c0067>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 <EOI>

2 11

[alsa-devel] Unlikely big volume range Plantronics BT300
by Dennis Kadioglu 08 May '16

08 May '16

Hi, when I connect the bluetooth BT300 of my Plantronics Headset I can see the following warning: [172855.985539] usb 2-1.3: Warning! Unlikely big volume range (=720), cval->res is probably wrong. [172855.985544] usb 2-1.3: [5] FU [PCM Playback Volume] ch = 1, val = -10240/1280/16 As far as I understand it, the problem may be solved by applying a quirk which is already applied for Logitech webcams. Can somebody help me in finding out the correct value for cval->res. I have no clue which calculations are necessary for that. Thanks in advance. Dennis Kadioglu

2 2

[alsa-devel] [PATCH 0/7] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 06 May '16

06 May '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since RFC v2: add BT function refine Kconfig and Makefile add prefix 'mt8173' and 'mt2701' for relative drivers. Garlic Tseng (7): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add clock and irq control for 2701 platform driver ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 35 +- sound/soc/mediatek/Makefile | 9 +- sound/soc/mediatek/mt2701/Makefile | 20 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 344 ++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 33 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 228 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1825 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 412 +++++ sound/soc/mediatek/mt2701/mt2701-irq.c | 74 + sound/soc/mediatek/mt2701/mt2701-irq.h | 26 + sound/soc/mediatek/mt2701/mt2701-reg.h | 195 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 101 ++ .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 488 +++--- sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 23 files changed, 3740 insertions(+), 368 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-irq.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-irq.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (67%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

3 13

[alsa-devel] [PATCH 0/2 v6] Add I2S audio support for ARC AXS10x boards
by Jose Abreu 04 May '16

04 May '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds I2S audio for the AXS10x platform. Changes v5 -> v6 * Use SNDRV_DMA_TYPE_CONTINUOUS Changes v4 -> v5 * Resolve undefined references when compiling as module * Dropped adv7511 audio patches * Use DMA properties in I2S to check which mode to use: PIO or DMA (as suggested by Lars-Peter Clausen) Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh(a)kernel.org> Cc: Alexey Brodkin <abrodkin(a)synopsys.com> Cc: linux-snps-arc(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Jose Abreu (2): ASoC: dwc: Add custom PCM driver ASoC: dwc: Update DOCUMENTATION for I2S Driver .../devicetree/bindings/sound/designware-i2s.txt | 9 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware.h | 71 +++++++ sound/soc/dwc/designware_i2s.c | 94 ++++++--- sound/soc/dwc/designware_pcm.c | 228 +++++++++++++++++++++ 6 files changed, 383 insertions(+), 29 deletions(-) create mode 100644 sound/soc/dwc/designware.h create mode 100644 sound/soc/dwc/designware_pcm.c -- 1.9.1

2 9

[alsa-devel] [PATCH v6 0/4] Propagate errors out from compressed streams
by Charles Keepax 04 May '16

04 May '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v5: - Add a debug print to note which state we transition to in snd_compr_stop_error It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 81 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 +++++- sound/soc/soc-compress.c | 5 ++- 4 files changed, 93 insertions(+), 9 deletions(-) -- 2.1.4

1 6

[alsa-devel] [PATCH v3 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 03 May '16

03 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 81 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 70 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] iMX6UL EVK with kernel 4.6
by Petr Kulhavy 02 May '16

02 May '16

Hi, I'm trying to run the latest mainline kernel 4.6 on the iMX6UL EVK. The board mostly works and the only problem is audio. Even though the sound card is detected and the ALSA mixer can be controlled there is no MCLK coming out of the SAI2 interface. All the related peripherals in the DTS (pinmux, SAI2, sound card, codec) seem to be set-up correctly, I also tried out various things like disabling the ASRC, but without success and I'm slowly running out of ideas. Could somebody give me a hint please, what is different to the Freescale kernel branch where the MCLK works? By the way, why aren't the Freescale patches merged to the mainline kernel? Thanks Petr

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