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

[alsa-devel] [PATCH v4 0/2] ASoC: add driver support for CX2092X DSP
by simon.ho.cnxt＠gmail.com 06 Apr '17

06 Apr '17

From: Simon Ho <simon.ho(a)conexant.com> Hi Pierre, Mark and Takashi I corrected the driver based on your input as below. - Corrected coding style. - Removed ".owner" assignment. - Removed SPI interface support since SPI fearure is removed from this new chip. Thanks, Simon Simon Ho (2): doc: cx2092x: Add DT bingings doc for CX2092X DSP ASoC: Add support for Conexant CX2092X DSP .../devicetree/bindings/sound/cnxt,cx2092x.txt | 25 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2092x.c | 436 +++++++++++++++++++++ sound/soc/codecs/cx2092x.h | 19 + 5 files changed, 488 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cnxt,cx2092x.txt create mode 100644 sound/soc/codecs/cx2092x.c create mode 100644 sound/soc/codecs/cx2092x.h -- 2.7.4

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[alsa-devel] [PATCH 2/2] pcm:plugin: Fix sound capture via MMAP access
by sutar.mounesh＠gmail.com 05 Apr '17

05 Apr '17

From: Timo Wischer <twischer(a)de.adit-jv.com> Distorted sound is heard if the capture device of an ioplug plugin is used, which is accessed through any ALSA plugin (like copy, linear, ...) via MMAP access. E.g. aplay -> loop -> copy -> arecord -M -> aplay -> hw:0 As mentioned in the ALSA API (see pcm/pcm.c:942): The function #snd_pcm_avail_update() have to be called before any mmap begin+commit operation. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index aaea2ab..ec57f6e 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -490,7 +490,14 @@ static snd_pcm_sframes_t snd_pcm_plugin_avail_update(snd_pcm_t *pcm) snd_pcm_uframes_t slave_offset; snd_pcm_uframes_t slave_frames = ULONG_MAX; snd_pcm_sframes_t result; - + /* As mentioned in the ALSA API (see pcm/pcm.c:942): + * The function #snd_pcm_avail_update() + * have to be called before any mmap begin+commit operation. + * Otherwise the snd_pcm_areas_copy will not called a second time. + * But this is needed, if the ring buffer wrap is reached and + * there is more data available. + */ + slave_size = snd_pcm_avail_update(slave); result = snd_pcm_mmap_begin(slave, &slave_areas, &slave_offset, &slave_frames); if (result < 0) { err = result; -- 2.7.4

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[alsa-devel] [PATCH 1/2] plugin:dshare: wrong state reporting
by sutar.mounesh＠gmail.com 05 Apr '17

05 Apr '17

From: Andreas Pape <apape(a)de.adit-jv.com> If plugin dshare detects underrun, it reports this to the user via return value -EPIPE and setting dshare state to 'xrun' which is correct. But, if user after this wants to check the stream state, it is misleadingly reported as 'running' instead of 'xrun'. With this behavior aplay e.g. will not do a proper underrun handling (restarting stream) but terminates streaming. This is due to plugin dshare always returns state of the slave pcm, in pcm_ops->state() which is not correct. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Signed-off-by: Mounesh Sutar <sutar.mounesh(a)gmail.com> diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 331f799..4c57fb4 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -244,7 +244,7 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) default: break; } - + status->state = snd_pcm_state(pcm); status->trigger_tstamp = dshare->trigger_tstamp; status->avail = snd_pcm_mmap_playback_avail(pcm); status->avail_max = status->avail > dshare->avail_max ? status->avail : dshare->avail_max; -- 2.7.4

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[alsa-devel] [PATCH] ALSA: firewire-lib/bebob/oxfw: improve response evaluation for AV/C commands
by Takashi Sakamoto 05 Apr '17

05 Apr '17

In ALSA firewire stack, some AV/C commands are supported, including vendor's extensions. Drivers includes response parser of each command, according to its requirements, while the parser is written with loose fashion in two points; error check and length check. This doesn't cause any issues such as kernel corruption, but should be improved. This commit modifies evaluations of return value on each parsers. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_command.c | 30 ++++++++++++++++++++++-------- sound/firewire/fcp.c | 12 +++++++++--- sound/firewire/oxfw/oxfw-command.c | 12 +++++++++--- 3 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/firewire/bebob/bebob_command.c b/sound/firewire/bebob/bebob_command.c index 9402cc1..f9b4225 100644 --- a/sound/firewire/bebob/bebob_command.c +++ b/sound/firewire/bebob/bebob_command.c @@ -31,13 +31,15 @@ int avc_audio_set_selector(struct fw_unit *unit, unsigned int subunit_id, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(8)); - if (err > 0 && err < 9) + if (err < 0) + ; + else if (err < 9) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; else if (buf[0] == 0x0a) /* REJECTED */ err = -EINVAL; - else if (err > 0) + else err = 0; kfree(buf); @@ -67,7 +69,9 @@ int avc_audio_get_selector(struct fw_unit *unit, unsigned int subunit_id, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(8)); - if (err > 0 && err < 9) + if (err < 0) + ; + else if (err < 9) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -120,7 +124,9 @@ int avc_bridgeco_get_plug_type(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(9)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 11) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -150,7 +156,9 @@ int avc_bridgeco_get_plug_ch_pos(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, 256, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(9)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 11) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -187,7 +195,9 @@ int avc_bridgeco_get_plug_section_type(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, 12, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(9) | BIT(10)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 12) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -221,7 +231,9 @@ int avc_bridgeco_get_plug_input(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 16, buf, 16, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7)); - if ((err >= 0) && (err < 8)) + if (err < 0) + ; + else if (err < 16) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -260,7 +272,9 @@ int avc_bridgeco_get_plug_strm_fmt(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, *len, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(10)); - if ((err >= 0) && (err < 12)) + if (err < 0) + ; + else if (err < 12) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index cce1976..61dda82 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -63,7 +63,9 @@ int avc_general_set_sig_fmt(struct fw_unit *unit, unsigned int rate, /* do transaction and check buf[1-5] are the same against command */ err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5)); - if (err >= 0 && err < 8) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -106,7 +108,9 @@ int avc_general_get_sig_fmt(struct fw_unit *unit, unsigned int *rate, /* do transaction and check buf[1-4] are the same against command */ err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2) | BIT(3) | BIT(4)); - if (err >= 0 && err < 8) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -154,7 +158,9 @@ int avc_general_get_plug_info(struct fw_unit *unit, unsigned int subunit_type, buf[3] = 0xff & subfunction; err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2)); - if (err >= 0 && err < 8) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; diff --git a/sound/firewire/oxfw/oxfw-command.c b/sound/firewire/oxfw/oxfw-command.c index 12ef325..ac3e2e3 100644 --- a/sound/firewire/oxfw/oxfw-command.c +++ b/sound/firewire/oxfw/oxfw-command.c @@ -34,7 +34,9 @@ int avc_stream_set_format(struct fw_unit *unit, enum avc_general_plug_dir dir, err = fcp_avc_transaction(unit, buf, len + 10, buf, len + 10, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7) | BIT(8)); - if ((err > 0) && (err < len + 10)) + if (err < 0) + ; + else if (err < len + 10) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -77,7 +79,9 @@ int avc_stream_get_format(struct fw_unit *unit, err = fcp_avc_transaction(unit, buf, 12, buf, *len, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5) | BIT(6) | BIT(7)); - if ((err > 0) && (err < 10)) + if (err < 0) + ; + else if (err < 12) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; @@ -139,7 +143,9 @@ int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate, /* do transaction and check buf[1-5] are the same against command */ err = fcp_avc_transaction(unit, buf, 8, buf, 8, BIT(1) | BIT(2) | BIT(3) | BIT(4) | BIT(5)); - if ((err > 0) && (err < 8)) + if (err < 0) + ; + else if (err < 8) err = -EIO; else if (buf[0] == 0x08) /* NOT IMPLEMENTED */ err = -ENOSYS; -- 2.9.3

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[alsa-devel] [PATCH] ALSA: firewire-motu: remove invalid bitshift for register value
by Takashi Sakamoto 05 Apr '17

05 Apr '17

In protocol version 3, drivers can read current sampling clock status from register 0x'ffff'f000'0b14. 8 bits of LSB of this register represents type of signal as source of clock. Current driver code includes invalid bitshift to handle the parameter. This commit fixes the bug. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 5992e30034c4 ("ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index b463da9..ddb6472 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -14,7 +14,6 @@ #define V3_CLOCK_RATE_MASK 0x0000ff00 #define V3_CLOCK_RATE_SHIFT 8 #define V3_CLOCK_SOURCE_MASK 0x000000ff -#define V3_CLOCK_SOURCE_SHIFT 8 #define V3_OPT_IFACE_MODE_OFFSET 0x0c94 #define V3_ENABLE_OPT_IN_IFACE_A 0x00000001 @@ -101,7 +100,7 @@ static int v3_get_clock_source(struct snd_motu *motu, return err; data = be32_to_cpu(reg); - val = (data & V3_CLOCK_SOURCE_MASK) >> V3_CLOCK_SOURCE_SHIFT; + val = data & V3_CLOCK_SOURCE_MASK; if (val == 0x00) { *src = SND_MOTU_CLOCK_SOURCE_INTERNAL; } else if (val == 0x01) { -- 2.9.3

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[alsa-devel] [PATCH] ALSA: oxfw: fix regression to handle Stanton SCS.1m/1d
by Takashi Sakamoto 05 Apr '17

05 Apr '17

At a commit 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card"), ALSA oxfw driver fails to handle SCS.1m/1d, due to -EBUSY at a call of snd_card_register(). The cause is that the driver manages to register two rawmidi instances with the same device number 0. This is a regression introduced since kernel 4.7. This commit fixes the regression, by fixing up device property after discovering stream formats. Fixes: 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card") Cc: <stable(a)vger.kernel.org> # 4.7+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 74d7fb6..413ab63 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -227,11 +227,11 @@ static void do_registration(struct work_struct *work) if (err < 0) goto error; - err = detect_quirks(oxfw); + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; - err = snd_oxfw_stream_discover(oxfw); + err = detect_quirks(oxfw); if (err < 0) goto error; -- 2.9.3

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[alsa-devel] [PATCH 0/4] ALSA: firewire-digi00x: integration to support console models
by Takashi Sakamoto 05 Apr '17

05 Apr '17

Hi, Disidesign Digi00x series includes two types of unit; rack and console. ALSA Digi00x driver is originally designed to handle both types. Recently I got console model and realized that it can't handle console type, as a result of packet sniffing. This patchset is to add proper support for unit of the console type, with new information from the reverse-engineering work. Takashi Sakamoto (4): ALSA: firewire-digi00x: add support for console models of Digi00x series ALSA: firewire-digi00x: handle all MIDI messages on streaming packets ALSA: firewire-digi00x: allow user space applications to read/write MIDI messages for all ports ALSA: firewire-digi00x: remove transaction handler for unknown purpose sound/firewire/digi00x/amdtp-dot.c | 55 ++++--- sound/firewire/digi00x/digi00x-midi.c | 208 ++++++++++----------------- sound/firewire/digi00x/digi00x-transaction.c | 88 ++---------- sound/firewire/digi00x/digi00x.c | 13 +- sound/firewire/digi00x/digi00x.h | 7 +- 5 files changed, 145 insertions(+), 226 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH 00/14] ALSA: fireface: new driver for RME Fireface series
by Takashi Sakamoto 05 Apr '17

05 Apr '17

Hi, This patchset updates my previous RFCv3, and go for ALSA upstream. [alsa-devel] [RFC v3][PATCH 00/13] ALSA: fireface: new driver for RME Fireface series http://mailman.alsa-project.org/pipermail/alsa-devel/2017-February/117811.h… This patchset newly adds a driver into ALSA firewire stack, to support for Fireface 400 in RME FireWire series. Unfortunately, units on this series are completely against IEC 61883-1/6. This brings much commits and descriptions into this patchset. I'd ask you to review much amount of codes, as well as MOTU FireWire series, to finish my work for AMDTP variants[0]. As I noted to my former patchset, units of the series have unique feature; resampling between packet/data block processing layer and sample processing layer. This patchset relies on this layer, in short the driver doesn't handle sampling clock frequency, just handle sampling transmission frequency and expects units to generates audio from resampled data. Users should use proper application to configure sampling clock frequency when they want to match the frequency and sampling transmission frequency. Changes from RFCv3: * Add function pointer to AMDTP structure to reduce branching on packet processing. * Improve commit description. [0]: [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants Takashi Sakamoto (14): ALSA: fireface: add skeleton for RME Fireface series ALSA: fireface: postpone sound card registration ALSA: fireface: add model specific structure ALSA: fireface: add an abstraction layer for model-specific protocols ALSA: fireface: add transaction support ALSA: fireface: add support for MIDI functionality ALSA: fireface: add proc node to help debugging ALSA: firewire-lib: use the same prototype for functions to handle packet ALSA: firewire-lib: add no-header packet processing ALSA: fireface: add unique data processing layer ALSA: fireface: add stream management functionality ALSA: fireface: add support for PCM functionality ALSA: fireface: add hwdep interface ALSA: fireface: add support for Fireface 400 include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 2 +- sound/firewire/Kconfig | 8 + sound/firewire/Makefile | 1 + sound/firewire/amdtp-stream-trace.h | 6 +- sound/firewire/amdtp-stream.c | 107 ++++++-- sound/firewire/amdtp-stream.h | 6 + sound/firewire/fireface/Makefile | 3 + sound/firewire/fireface/amdtp-ff.c | 155 +++++++++++ sound/firewire/fireface/ff-hwdep.c | 191 +++++++++++++ sound/firewire/fireface/ff-midi.c | 131 +++++++++ sound/firewire/fireface/ff-pcm.c | 409 ++++++++++++++++++++++++++++ sound/firewire/fireface/ff-proc.c | 63 +++++ sound/firewire/fireface/ff-protocol-ff400.c | 371 +++++++++++++++++++++++++ sound/firewire/fireface/ff-stream.c | 282 +++++++++++++++++++ sound/firewire/fireface/ff-transaction.c | 295 ++++++++++++++++++++ sound/firewire/fireface/ff.c | 209 ++++++++++++++ sound/firewire/fireface/ff.h | 146 ++++++++++ 18 files changed, 2365 insertions(+), 23 deletions(-) create mode 100644 sound/firewire/fireface/Makefile create mode 100644 sound/firewire/fireface/amdtp-ff.c create mode 100644 sound/firewire/fireface/ff-hwdep.c create mode 100644 sound/firewire/fireface/ff-midi.c create mode 100644 sound/firewire/fireface/ff-pcm.c create mode 100644 sound/firewire/fireface/ff-proc.c create mode 100644 sound/firewire/fireface/ff-protocol-ff400.c create mode 100644 sound/firewire/fireface/ff-stream.c create mode 100644 sound/firewire/fireface/ff-transaction.c create mode 100644 sound/firewire/fireface/ff.c create mode 100644 sound/firewire/fireface/ff.h -- 2.9.3

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[alsa-devel] [PATCH v2 1/2] ASoC: fsl_ssi: Remove an outdated comment
by Fabio Estevam 05 Apr '17

05 Apr '17

From: Fabio Estevam <fabio.estevam(a)nxp.com> The comment for the FSLSSI_I2S_RATES definition states that the driver currently only supports I2S slave mode, which is no longer correct. Remove such outdated comment. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Only remove the outdated part of the comment sound/soc/fsl/fsl_ssi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index fde08660..e0639e17 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -57,10 +57,8 @@ /** * FSLSSI_I2S_RATES: sample rates supported by the I2S * - * This driver currently only supports the SSI running in I2S slave mode, - * which means the codec determines the sample rate. Therefore, we tell - * ALSA that we support all rates and let the codec driver decide what rates - * are really supported. + * Tell ALSA that we support all rates and let the codec driver decide what + * rates are really supported. */ #define FSLSSI_I2S_RATES SNDRV_PCM_RATE_CONTINUOUS -- 2.7.4

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[alsa-devel] [PATCH] ASoC: twl6040: Add control for HS and HF mono to stereo selection
by Peter Ujfalusi 05 Apr '17

05 Apr '17

The new controls will give user the ability to route the left PDM channel data to the right headset/handsfree DAC. HS mono to stereo switch: PDM channel 1 (or mono) data to both HS DAC. HF mono to stereo switch: PDM channel 3 data to both HF DAC. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 748036e851ea..2b6ad09e0886 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -606,6 +606,14 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { twl6040_headset_power_get_enum, twl6040_headset_power_put_enum), + /* Left HS PDM data routed to Right HSDAC */ + SOC_SINGLE("Headset Mono to Stereo Playback Switch", + TWL6040_REG_HSRCTL, 7, 1, 0), + + /* Left HF PDM data routed to Right HFDAC */ + SOC_SINGLE("Handsfree Mono to Stereo Playback Switch", + TWL6040_REG_HFRCTL, 5, 1, 0), + SOC_ENUM_EXT("PLL Selection", twl6040_power_mode_enum, twl6040_pll_get_enum, twl6040_pll_put_enum), }; -- 2.12.2

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