October 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Digidesign Eleven Rack support
by Aidan Schurr 31 Oct '13

31 Oct '13

Hi guys, am wondering if support for the Digidesign Eleven Rack is on the cards for the future at all? Alsa mixer currently recognises it but lists it as having no available controls, while aplay -l and arecord -l miss it altogether. I found one thread from around 18 months ago that mentions it uses a non standard implementation of USB audio; does this pretty much preclude it from getting up and finding running? Happy to be a beta tester of that could help in any way.

2 1

[alsa-devel] [PATCH 0/3] dma: edma: Add cyclic DMA support
by Joel Fernandes 31 Oct '13

31 Oct '13

The following series adds Cyclic DMA support to TI EDMA DMA Engine driver. First we split out the calculations for the Slave DMA case into a separate function so that we may reuse it for the calculations of Cyclic DMA parameters. Next patch then adds the actual support for Cyclic DMA, enables interrupts correctly and uses's the callbacks in virt-dma during interrupts thus signalling back to the ALSA layer that a period was transmitted. Some background on motivation for this series: Currently, only user of Cyclic DMA in EDMA is davinci-pcm driver. As of today, this driver directly calls into the EDMA private API (arch/arm/common/edma.c) without going through the DMAEngine. davinci-pcm in future will be modified to use DMA Engine framework for Cyclic DMA instead of directly using the Private API. However that's a much larger effort, involving dealing with ping-pong from SRAM on user's of the Davinci McASP, etc. As a first step, we add Cyclic DMA support to the EDMA driver so that this may be used when the actual conversion of davinci-pcm happens. Tested series along with couple of hacks to davinci-pcm to work with DMA Engine: git@github.com:joelagnel/linux-kernel.git (branch dma/cyclic) Joel Fernandes (3): dma: edma: Split out PaRAM set calculations into its own function dma: edma: Add support for Cyclic DMA dma: edma: Increase maximum SG limit to 20 drivers/dma/edma.c | 350 +++++++++++++++++++++++++++++++++++++++++------------ 1 file changed, 273 insertions(+), 77 deletions(-) -- 1.8.1.2

2 11

[alsa-devel] [PATCH v3 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 31 Oct '13

31 Oct '13

Changelog v3: * PATCH-1: Add comments to indicate the end of v1 and v2 array. * PATCH-3: Use better way to keep watermark as even number. * * Nothing changes for PATCH-2 and PATCH-4 v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. * * ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply to one single branch. Otherwise, * ! it would break currect branches. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 ++++++------ arch/arm/boot/dts/imx6sl.dtsi | 12 ++++++------ drivers/dma/imx-sdma.c | 19 ++++++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 5 +++++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 22 +++++++++++++++++++++- 9 files changed, 62 insertions(+), 18 deletions(-) -- 1.8.4

3 6

[alsa-devel] [PATCH] asoc-tools: Render png image instead of jpeg in vizdapm.sh
by Jarkko Nikula 31 Oct '13

31 Oct '13

Lossless png image suits better to technical drawings than jpeg which may even have artifacts around sharp objects. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- For git://opensource.wolfsonmicro.com/asoc-tools.git --- bin/vizdapm.sh | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/bin/vizdapm.sh b/bin/vizdapm.sh index 11d1361..8d61dd2 100755 --- a/bin/vizdapm.sh +++ b/bin/vizdapm.sh @@ -16,7 +16,7 @@ usage() { cat << EOF -usage: $0 <dapm-debugfs-directory> <graph-dot-file> <graph-jpg-file> +usage: $0 <dapm-debugfs-directory> <graph-dot-file> <graph-png-file> EOF exit 1 } @@ -29,7 +29,7 @@ active= curpath=`pwd` dapm_debugfs="$1" graph_file=${curpath}"/$2" -graph_file_jpg=${curpath}"/$3" +graph_file_png=${curpath}"/$3" trap "{ [ -f "$graph_file" ] && rm -f "$graph_file"; exit; }" SIGINT SIGTERM @@ -62,4 +62,4 @@ done popd >/dev/null echo "}" >> "$graph_file" -dot -Kfdp -Tjpg "$graph_file" -o "$graph_file_jpg" +dot -Kfdp -Tpng "$graph_file" -o "$graph_file_png" -- 1.8.4.rc3

2 1

[alsa-devel] [PATCH] ALSA: hda - Add extra chmap for 2.1 outputs on ASUS laptops
by Takashi Iwai 31 Oct '13

31 Oct '13

ASUS N56VZ and N76VZ laptops have a bass speaker but its output comes only from the right channel. This patch adds the extra chmap specific to these models. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=846531 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 13ad49ca9357..2d7c86d81196 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4426,6 +4426,25 @@ static void alc272_fixup_mario(struct hda_codec *codec, "hda_codec: failed to override amp caps for NID 0x2\n"); } +static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_NA, SNDRV_CHMAP_LFE } }, /* LFE only on right */ + { } +}; + +/* override the 2.1 chmap */ +static void alc662_fixup_bass_chmap(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_BUILD) { + struct alc_spec *spec = codec->spec; + spec->gen.pcm_rec[0].stream[0].chmap = asus_pcm_2_1_chmaps; + } +} + enum { ALC662_FIXUP_ASPIRE, ALC662_FIXUP_IDEAPAD, @@ -4446,6 +4465,7 @@ enum { ALC662_FIXUP_INV_DMIC, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_HEADSET_MODE, + ALC662_FIXUP_BASS_CHMAP, }; static const struct hda_fixup alc662_fixups[] = { @@ -4620,6 +4640,12 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_alc668, }, + [ALC662_FIXUP_BASS_CHMAP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc662_fixup_bass_chmap, + .chained = true, + .chain_id = ALC662_FIXUP_ASUS_MODE4 + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -4633,8 +4659,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), - SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_ASUS_MODE4), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), -- 1.8.4.1

2 4

[alsa-devel] Realtek HD-A codec ALC262: How to unmute mixer node 0x24 Amp-In for Mic Pin 0x18
by Lin, Mengdong 31 Oct '13

31 Oct '13

Hi, We found when recording on ALC262, the audio mixer node 0x24 mutes the input from Mic Pin node 0x18, so we get silent data. Would someone please give me some advice to solve this? Node 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x0b I cannot find user-space amixer controls to unmute node 0x18. The recording data flows from Mic Pin 0x18 -> Mixer 0x24 -> audio input 0x07. The Amp-In value on pin 0x18 (Rear Mic Boost) and input 0x07 (Capture Volume) are okay. Thanks Mengdong

3 9

[alsa-devel] [PATCH v2 0/4] Add dual-fifo mode support of i.MX ssi
by Nicolin Chen 31 Oct '13

31 Oct '13

Changelog v2: * Instead of adding rogue scripts to current SDMA driver based on firmware * V1, we define the new SDMA firmware as version 2 and bisect the PATCH-1 * to two patches: The first is to add version check code to the SDMA driver; * And the second is to add SSI dual FIFO DMATYPE. * * Nothing changes for the last two patches. v1: * SSI can reduce hardware overrun/underrun possibility when using dual * fifo mode. To support this mode, we need to first update sdma sciprt * list, and then enable dual fifo BIT in SSI driver, and last update DT * bindings of i.MX series. * * ! This series of patches has a direct dependency between them. When * ! applying them, we need to apply in one single branch. Otherwise, * ! it would break currect branches. Nicolin Chen (4): dma: imx-sdma: Add sdma firmware version 2 support dma: imx-sdma: Add new dma type for ssi dual fifo script ASoC: fsl_ssi: Add dual fifo mode support ARM: dts: imx: use dual-fifo sdma script for ssi .../devicetree/bindings/dma/fsl-imx-sdma.txt | 1 + arch/arm/boot/dts/imx51.dtsi | 4 ++-- arch/arm/boot/dts/imx53.dtsi | 4 ++-- arch/arm/boot/dts/imx6qdl.dtsi | 12 +++++------ arch/arm/boot/dts/imx6sl.dtsi | 12 +++++------ drivers/dma/imx-sdma.c | 19 ++++++++++++++++- include/linux/platform_data/dma-imx-sdma.h | 3 +++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/fsl_ssi.c | 24 +++++++++++++++++++++- 9 files changed, 62 insertions(+), 18 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCH 0/9] ASoC: Yet another small fixes
by Takashi Iwai 31 Oct '13

31 Oct '13

Hi, this is yet another series of small fixes for ASoC codec bugs spotted by coverity. All minor issues, obviously. Takashi

2 25

[alsa-devel] [PATCH] ALSA: compress: fix drain calls blocking other compress functions
by Vinod Koul 31 Oct '13

31 Oct '13

The drain and drain_notify callback were blocked by low level driver untill the draining was complete. Due to this being invoked with big fat mutex held, others ops like reading timestamp, calling pause, drop were blocked. So to fix this we add a new snd_compr_drain_notify() API. This would be required to be invoked by low level driver when drain or partial drain has been completed by the DSP. Thus we make the drain and partial_drain callback as non blocking and driver returns immediately after notifying DSP. The waiting is done while relasing the lock so that other ops can go ahead. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> CC: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 12 ++++++++++ sound/core/compress_offload.c | 43 +++++++++++++++++++++++++++++++++++--- 2 files changed, 51 insertions(+), 4 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 9031a26..eff5f84 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -48,6 +48,8 @@ struct snd_compr_ops; * the ring buffer * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep + * @wait: drain wait queuq + * @draining: draining wait condition */ struct snd_compr_runtime { snd_pcm_state_t state; @@ -59,6 +61,8 @@ struct snd_compr_runtime { u64 total_bytes_available; u64 total_bytes_transferred; wait_queue_head_t sleep; + wait_queue_head_t wait; + unsigned int draining; void *private_data; }; @@ -171,4 +175,12 @@ static inline void snd_compr_fragment_elapsed(struct snd_compr_stream *stream) wake_up(&stream->runtime->sleep); } +static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) +{ + snd_BUG_ON(!stream); + + stream->runtime->draining = 1; + wake_up(&stream->runtime->wait); +} + #endif diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index bea523a..3582f9f 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -123,6 +123,7 @@ static int snd_compr_open(struct inode *inode, struct file *f) } runtime->state = SNDRV_PCM_STATE_OPEN; init_waitqueue_head(&runtime->sleep); + init_waitqueue_head(&runtime->wait); data->stream.runtime = runtime; f->private_data = (void *)data; mutex_lock(&compr->lock); @@ -682,12 +683,36 @@ static int snd_compr_stop(struct snd_compr_stream *stream) if (!retval) { stream->runtime->state = SNDRV_PCM_STATE_SETUP; wake_up(&stream->runtime->sleep); + stream->runtime->draining = 1; + wake_up(&stream->runtime->wait); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; } return retval; } +static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) +{ + int retval = 0; + + /* + * we are called with lock held. So drop the lock while we wait for + * drain complete notfication from the driver + */ + stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + mutex_unlock(&stream->device->lock); + + retval = wait_event_interruptible(stream->runtime->wait, stream->runtime->draining); + if (retval) + pr_err("Wait for drain failed %d\n", retval); + + wake_up(&stream->runtime->sleep); + mutex_lock(&stream->device->lock); + stream->runtime->state = SNDRV_PCM_STATE_SETUP; + + return retval; +} + static int snd_compr_drain(struct snd_compr_stream *stream) { int retval; @@ -695,12 +720,16 @@ static int snd_compr_drain(struct snd_compr_stream *stream) if (stream->runtime->state == SNDRV_PCM_STATE_PREPARED || stream->runtime->state == SNDRV_PCM_STATE_SETUP) return -EPERM; + + stream->runtime->draining = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_DRAIN); - if (!retval) { - stream->runtime->state = SNDRV_PCM_STATE_DRAINING; + if (retval) { + pr_err("SND_COMPR_TRIGGER_DRAIN failed %d\n", retval); wake_up(&stream->runtime->sleep); + return retval; } - return retval; + + return snd_compress_wait_for_drain(stream); } static int snd_compr_next_track(struct snd_compr_stream *stream) @@ -735,10 +764,16 @@ static int snd_compr_partial_drain(struct snd_compr_stream *stream) if (stream->next_track == false) return -EPERM; + stream->runtime->draining = 0; retval = stream->ops->trigger(stream, SND_COMPR_TRIGGER_PARTIAL_DRAIN); + if (retval) { + pr_err("Partial drain returned failure\n"); + wake_up(&stream->runtime->sleep); + return retval; + } stream->next_track = false; - return retval; + return snd_compress_wait_for_drain(stream); } static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) -- 1.7.0.4

2 18

[alsa-devel] [PATCH] ASoC: wm_adsp: Release firmware on memory allocation failure
by Dimitris Papastamos 30 Oct '13

30 Oct '13

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b3fbe4a..a2144b0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -630,7 +630,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) GFP_KERNEL | GFP_DMA); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } ret = regmap_raw_write(regmap, reg, buf, -- 1.8.4.2

2 1