October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest" to the asoc tree
by Mark Brown 31 Oct '15

31 Oct '15

The patch ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bbb64300469d207f62c08ea7358c839991a52877 Mon Sep 17 00:00:00 2001 From: Mengdong Lin <mengdong.lin(a)intel.com> Date: Fri, 30 Oct 2015 15:13:16 +0800 Subject: [PATCH] ASoC: topology: ABI - Rename dai_elems to pcm_elems in manifest This field is the number of PCM objects (a pair of FE DAI and DAI link). Signed-off-by: Lin Mengdong <mengdong.lin(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 26539a7..c4cc1e4 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -243,7 +243,7 @@ struct snd_soc_tplg_manifest { __le32 control_elems; /* number of control elements */ __le32 widget_elems; /* number of widget elements */ __le32 graph_elems; /* number of graph elements */ - __le32 dai_elems; /* number of DAI elements */ + __le32 pcm_elems; /* number of PCM elements */ __le32 dai_link_elems; /* number of DAI link elements */ struct snd_soc_tplg_private priv; } __attribute__((packed)); -- 2.6.1

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Re: [alsa-devel] [PATCH 2/2] sound: soc: Add Cygnus audio driver
by Arnd Bergmann 31 Oct '15

31 Oct '15

On Saturday 31 October 2015 07:20:17 kbuild test robot wrote: > > 785 dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%x @ 0x%p\n", > 786 __func__, size, buf->area); > Both reports are for the same bug: the "0x%x" should be "0x%zx". There is another bug in the same line: "0x%p" should be "%pK", to avoid the duplicate "0x" and to not leak the pointer when kptr_restrict is set. Arnd

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[alsa-devel] [PATCH 0/2] Add audio support for Broadcom Cygnus SoC
by Simran Rai 30 Oct '15

30 Oct '15

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings, core audio driver and dts. 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 v4.3-rc5 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v1 Simran Rai (2): dt-bindings: sound: Add DT bindings for Broadcom Cygnus audio sound: soc: Add Broadcom Cygnus audio driver. .../bindings/sound/brcm,cygnus-audio.txt | 63 + sound/soc/bcm/Kconfig | 18 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 903 ++++++++++++ sound/soc/bcm/cygnus-ssp.c | 1532 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 129 ++ 6 files changed, 2650 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 0/3] ALSA: firewire-digi00x: add another rawmidi character device for MIDI control ports
by Takashi Sakamoto 30 Oct '15

30 Oct '15

Hi, Digi 002/003 family uses two ways of MIDI transmission; by isochronous packet streaming and by asynchronous transmission. The former is for physical MIDI ports and the latter is for physical controls on device surface. Currently, all of these ports are handled by a set of operations in one ALSA rawmidi character device, thus: $ amidi -l Dir Device Name IO hw:1,0,0 Digi 002Rack control IO hw:1,0,1 Digi 002Rack MIDI 1 O hw:1,0,2 Digi 002Rack MIDI 2 Although, it's preferable to have two rawmidi character devices because these transmission ways are completely different and it's better to assign each set of operations. This patchset adds another rawmidi character device to ALSA digi00x driver. As a result, the driver register two rawmidi character device for each MIDI ports: $ amidi -l Dir Device Name IO hw:1,0,0 Digi 002Rack MIDI 1 O hw:1,0,1 Digi 002Rack MIDI 2 IO hw:1,1,0 Digi 002Rack control I'm sorry to post this patchset in this time, but next release is a first version with digi00x driver and it better to includes this intrusive patchset before releasing. Takashi Sakamoto (3): ALSA: firewire-digi00x: rename identifiers of MIDI operation for physical ports ALSA: firewire-digi00x: add MIDI operations for MIDI control port ALSA: firewire-digi00x: add another rawmidi character device for MIDI control ports sound/firewire/digi00x/digi00x-midi.c | 197 ++++++++++++++++++++++------------ 1 file changed, 130 insertions(+), 67 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH v2 0/3] Add support for Cirrus Logic CS47L24 and WM1831 codecs
by Richard Fitzgerald 30 Oct '15

30 Oct '15

mfd patch based on Lee Jones's for-next gpio patch based on Linus Walleij's for-next ASoC patch based on Mark Brown's for-next Lee/Mark - can one of you take all three patches (with maintainer acks) so that the patch set is kept together? Richard Fitzgerald (3): mfd: arizona: Support Cirrus Logic CS47L24 and WM1831 gpio: arizona: Support Cirrus Logic CS47L24 and WM1831 ASoC: cs47l24: Add driver for Cirrus Logic CS47L24 and WM1831 codecs Documentation/devicetree/bindings/mfd/arizona.txt | 4 +- drivers/gpio/gpio-arizona.c | 4 + drivers/mfd/Kconfig | 19 +- drivers/mfd/Makefile | 3 + drivers/mfd/arizona-core.c | 74 +- drivers/mfd/arizona-irq.c | 40 +- drivers/mfd/arizona-spi.c | 7 + drivers/mfd/arizona.h | 4 + drivers/mfd/cs47l24-tables.c | 1629 +++++++++++++++++++++ include/linux/mfd/arizona/core.h | 3 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs47l24.c | 1148 +++++++++++++++ sound/soc/codecs/cs47l24.h | 23 + 14 files changed, 2942 insertions(+), 26 deletions(-) create mode 100644 drivers/mfd/cs47l24-tables.c create mode 100644 sound/soc/codecs/cs47l24.c create mode 100644 sound/soc/codecs/cs47l24.h -- 1.9.1

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[alsa-devel] [PATCH alsa-lib] pcm: dmix: Handle slave PCM xrun and unexpected states properly
by Takashi Iwai 30 Oct '15

30 Oct '15

Currently, dmix & co plugins ignore the XRUN state of the slave PCM. It's (supposedly) because dmix deals with the PCM in a free-wheel mode, which is equivalent with XRUN. But, this difference (whether the correct freewheel or XRUN) should be done by the kernel, and we may have an XRUN state indeed (e.g. via xrun injection). This patch fixes this lack of behavior, to handle PCM xrun and does prepare when the slave PCM is in such a state. Also, the patch consolidates the prepare callback for all dmix, dsnoop and dshare plugins, and fix/cleanup a bit for dshare/dsnoop codes to align with dsnoop code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 22 ++++++++++++++++++++++ src/pcm/pcm_direct.h | 3 +++ src/pcm/pcm_dmix.c | 16 +++------------- src/pcm/pcm_dshare.c | 22 +++++++--------------- src/pcm/pcm_dsnoop.c | 23 +++++++---------------- 5 files changed, 42 insertions(+), 44 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 195fddf06cda..fd3877c30d04 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -807,6 +807,28 @@ int snd_pcm_direct_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map) return snd_pcm_set_chmap(dmix->spcm, map); } +int snd_pcm_direct_prepare(snd_pcm_t *pcm) +{ + snd_pcm_direct_t *dmix = pcm->private_data; + int err; + + switch (snd_pcm_state(dmix->spcm)) { + case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: + case SND_PCM_STATE_DISCONNECTED: + err = snd_pcm_prepare(dmix->spcm); + if (err < 0) + return err; + snd_pcm_start(dmix->spcm); + break; + } + snd_pcm_direct_check_interleave(dmix, pcm); + dmix->state = SND_PCM_STATE_PREPARED; + dmix->appl_ptr = dmix->last_appl_ptr = 0; + dmix->hw_ptr = 0; + return snd_pcm_direct_set_timer_params(dmix); +} + int snd_pcm_direct_resume(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; diff --git a/src/pcm/pcm_direct.h b/src/pcm/pcm_direct.h index 9b1ddbcf424a..611ad29a4821 100644 --- a/src/pcm/pcm_direct.h +++ b/src/pcm/pcm_direct.h @@ -224,6 +224,8 @@ struct snd_pcm_direct { snd1_pcm_direct_mmap #define snd_pcm_direct_munmap \ snd1_pcm_direct_munmap +#define snd_pcm_direct_prepare \ + snd1_pcm_direct_prepare #define snd_pcm_direct_resume \ snd1_pcm_direct_resume #define snd_pcm_direct_timer_stop \ @@ -304,6 +306,7 @@ int snd_pcm_direct_sw_params(snd_pcm_t *pcm, snd_pcm_sw_params_t * params); int snd_pcm_direct_channel_info(snd_pcm_t *pcm, snd_pcm_channel_info_t * info); int snd_pcm_direct_mmap(snd_pcm_t *pcm); int snd_pcm_direct_munmap(snd_pcm_t *pcm); +int snd_pcm_direct_prepare(snd_pcm_t *pcm); int snd_pcm_direct_resume(snd_pcm_t *pcm); int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix); void snd_pcm_direct_clear_timer_queue(snd_pcm_direct_t *dmix); diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 58e4975d5225..b26a5c790e7e 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -450,9 +450,10 @@ static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { + case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: - return state; case SND_PCM_STATE_DISCONNECTED: + dmix->state = state; return state; default: break; @@ -533,17 +534,6 @@ static int snd_pcm_dmix_hwsync(snd_pcm_t *pcm) } } -static int snd_pcm_dmix_prepare(snd_pcm_t *pcm) -{ - snd_pcm_direct_t *dmix = pcm->private_data; - - snd_pcm_direct_check_interleave(dmix, pcm); - dmix->state = SND_PCM_STATE_PREPARED; - dmix->appl_ptr = dmix->last_appl_ptr = 0; - dmix->hw_ptr = 0; - return snd_pcm_direct_set_timer_params(dmix); -} - static void reset_slave_ptr(snd_pcm_t *pcm, snd_pcm_direct_t *dmix) { dmix->slave_appl_ptr = dmix->slave_hw_ptr = *dmix->spcm->hw.ptr; @@ -920,7 +910,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .state = snd_pcm_dmix_state, .hwsync = snd_pcm_dmix_hwsync, .delay = snd_pcm_dmix_delay, - .prepare = snd_pcm_dmix_prepare, + .prepare = snd_pcm_direct_prepare, .reset = snd_pcm_dmix_reset, .start = snd_pcm_dmix_start, .drop = snd_pcm_dmix_drop, diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index 02370dc7082d..58e47bbeac67 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -237,11 +237,14 @@ static int snd_pcm_dshare_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; - switch (snd_pcm_state(dshare->spcm)) { + snd_pcm_state_t state; + state = snd_pcm_state(dshare->spcm); + switch (state) { + case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: - return SND_PCM_STATE_SUSPENDED; case SND_PCM_STATE_DISCONNECTED: - return SND_PCM_STATE_DISCONNECTED; + dshare->state = state; + return state; default: break; } @@ -296,17 +299,6 @@ static int snd_pcm_dshare_hwsync(snd_pcm_t *pcm) } } -static int snd_pcm_dshare_prepare(snd_pcm_t *pcm) -{ - snd_pcm_direct_t *dshare = pcm->private_data; - - snd_pcm_direct_check_interleave(dshare, pcm); - dshare->state = SND_PCM_STATE_PREPARED; - dshare->appl_ptr = dshare->last_appl_ptr = 0; - dshare->hw_ptr = 0; - return snd_pcm_direct_set_timer_params(dshare); -} - static int snd_pcm_dshare_reset(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; @@ -595,7 +587,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .state = snd_pcm_dshare_state, .hwsync = snd_pcm_dshare_hwsync, .delay = snd_pcm_dshare_delay, - .prepare = snd_pcm_dshare_prepare, + .prepare = snd_pcm_direct_prepare, .reset = snd_pcm_dshare_reset, .start = snd_pcm_dshare_start, .drop = snd_pcm_dshare_drop, diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 8a2e87ad0ae1..576c35b111cd 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -201,12 +201,14 @@ static int snd_pcm_dsnoop_status(snd_pcm_t *pcm, snd_pcm_status_t * status) static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; - switch (snd_pcm_state(dsnoop->spcm)) { + snd_pcm_state_t state; + state = snd_pcm_state(dsnoop->spcm); + switch (state) { + case SND_PCM_STATE_XRUN: case SND_PCM_STATE_SUSPENDED: - return SND_PCM_STATE_SUSPENDED; case SND_PCM_STATE_DISCONNECTED: - dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; - return -ENODEV; + dsnoop->state = state; + return state; default: break; } @@ -258,17 +260,6 @@ static int snd_pcm_dsnoop_hwsync(snd_pcm_t *pcm) } } -static int snd_pcm_dsnoop_prepare(snd_pcm_t *pcm) -{ - snd_pcm_direct_t *dsnoop = pcm->private_data; - - snd_pcm_direct_check_interleave(dsnoop, pcm); - dsnoop->state = SND_PCM_STATE_PREPARED; - dsnoop->appl_ptr = 0; - dsnoop->hw_ptr = 0; - return snd_pcm_direct_set_timer_params(dsnoop); -} - static int snd_pcm_dsnoop_reset(snd_pcm_t *pcm) { snd_pcm_direct_t *dsnoop = pcm->private_data; @@ -497,7 +488,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .state = snd_pcm_dsnoop_state, .hwsync = snd_pcm_dsnoop_hwsync, .delay = snd_pcm_dsnoop_delay, - .prepare = snd_pcm_dsnoop_prepare, + .prepare = snd_pcm_direct_prepare, .reset = snd_pcm_dsnoop_reset, .start = snd_pcm_dsnoop_start, .drop = snd_pcm_dsnoop_drop, -- 2.6.2

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[alsa-devel] [PATCH 0/4] dmaengine: Add transfer termination synchronization support
by Lars-Peter Clausen 30 Oct '15

30 Oct '15

The DMAengine API has a long standing issue that is inherent to the API itself. For a client that calls dmaengine_terminate_all() it is not possible to properly synchronize the completion of any currently running complete callbacks to the current context. This means it is possible to end up with a use-after-free race condition if client frees resources that are accessed in a complete callback before the complete callback has finished running. This patch series introduces a new explicit synchronization primitive to the DMAengine API which allows clients to ensure that all complete callbacks have finished running. This allows them to safely free any resources that might be accessed in a complete callback. The series for now only implements synchronization support for a single driver and only updates single client to make use of the new API. If there is agreement on the general approach more will follow. - Lars Lars-Peter Clausen (4): dmaengine: Add transfer termination synchronization support dmaengine: virt-dma: Add synchronization helper function dmaengine: axi_dmac: Add synchronization support ALSA: pcm_dmaengine: Properly synchronize DMA on shutdown Documentation/dmaengine/client.txt | 38 ++++++++++++++- Documentation/dmaengine/provider.txt | 20 +++++++- drivers/dma/dma-axi-dmac.c | 8 ++++ drivers/dma/dmaengine.c | 5 +- drivers/dma/virt-dma.h | 13 ++++++ include/linux/dmaengine.h | 90 ++++++++++++++++++++++++++++++++++++ sound/core/pcm_dmaengine.c | 9 ++-- 7 files changed, 175 insertions(+), 8 deletions(-) -- 2.1.4

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Re: [alsa-devel] ASoC: Intel: Add Skylake HDA platform driver
by Vinod Koul 30 Oct '15

30 Oct '15

> > Hello Jeeja KP, > > > > This is a semi-automatic email about new static checker warnings. > > > > The patch a40e693c7f5e: "ASoC: Intel: Add Skylake HDA platform driver" > > from Jul 9, 2015, leads to the following Smatch complaint: > > > > sound/soc/intel/skylake/skl-pcm.c:802 skl_get_position() > > warn: variable dereferenced before check 'substream->runtime' (see > > line 792) > > > > sound/soc/intel/skylake/skl-pcm.c > > 791 struct snd_pcm_substream *substream = hstr->substream; > > 792 struct hdac_ext_bus *ebus = get_bus_ctx(substream); > > ^^^^^^^^^^^^^^^^^^^^^^ Dereference inside > > function. > > > > 793 unsigned int pos; > > 794 int delay; > > 795 > > 796 /* use the position buffer as default */ > > 797 pos = > > snd_hdac_stream_get_pos_posbuf(hdac_stream(hstream)); > > 798 > > 799 if (pos >= hdac_stream(hstream)->bufsize) > > 800 pos = 0; > > 801 > > 802 if (substream->runtime) { > > ^^^^^^^^^^^^^^^^^^ > > Checked too late. Right, will move ebus init here. Thanks for reporting this > > > > 803 delay = skl_get_delay_from_lpib(ebus, hstream, pos) > > 804 + codec_delay; > > > > regards, > > dan carpenter -- ~Vinod

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[alsa-devel] [PATCH 1/1] BAT: Add default device name for playback and capture
by han.lu＠intel.com 30 Oct '15

30 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> If the playback device and the capture device was not set through '-D', '-P' or '-C' options, set the name of the two devices to "default". So by default, BAT will work through PulseAudio sound server on Ubuntu/Debian, and work though CRAS sound server on Chromium OS/Chrome OS. If the playback device or the capture device was set through '-D', '-P' or '-C' options by user, the device name "default" will not be set. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/bat.c b/bat/bat.c index 086b9fa..4310be0 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -473,6 +473,10 @@ static int bat_init(struct bat *bat) return err; } + /* Set default playback and capture devices */ + if (bat->playback.device == NULL && bat->capture.device == NULL) + bat->playback.device = bat->capture.device = "default"; + /* Determine capture file */ if (bat->local) { bat->capture.file = bat->playback.file; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: fix typo in sizeof
by Vincent Stehlé 30 Oct '15

30 Oct '15

The size of the pointer to a data structure to send is erroneously passed to sst_ipc_tx_message_wait() as its tx_bytes argument. It should be given the size of the pointed skl_ipc_dxstate_info structure instead. Coincidentally, both the pointer and the structure have the same size of 8 bytes on a 64 bit machine, which "masks" the issue. Compiling for 32 bit reveals the issue more clearly. Fix the typo for correctness, and to make the code robust to future evolutions of the skl_ipc_dxstate_info structure size. This fixes the following coccicheck error: sound/soc/intel/skylake/skl-sst-ipc.c:641:8-14: ERROR: application of sizeof to pointer Signed-off-by: Vincent Stehlé <vincent.stehle(a)laposte.net> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Jeeja KP <jeeja.kp(a)intel.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: trivial(a)kernel.org --- sound/soc/intel/skylake/skl-sst-ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c index 937a0a3..9aee835 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ b/sound/soc/intel/skylake/skl-sst-ipc.c @@ -638,7 +638,7 @@ int skl_ipc_set_dx(struct sst_generic_ipc *ipc, u8 instance_id, dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__, header.primary, header.extension); ret = sst_ipc_tx_message_wait(ipc, *ipc_header, - dx, sizeof(dx), NULL, 0); + dx, sizeof(*dx), NULL, 0); if (ret < 0) { dev_err(ipc->dev, "ipc: set dx failed, err %d\n", ret); return ret; -- 2.6.1

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