June 2018 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Lukas Wunner 29 Jun '18

29 Jun '18

> On Thu, Jun 28, 2018 at 2:22 AM, Jim Qu <Jim.Qu(a)amd.com> wrote: > > On modern laptop, there are more and more platforms > > have two GPUs, and each of them maybe have audio codec > > for HDMP/DP output. For some dGPU which is no output, > > audio codec usually is disabled. > > > > In currect HDA audio driver, it will set all codec as > > VGA_SWITCHEROO_DIS, so if system runtime pm is enabled, > > the audio which is binded to UMA will be suspended. Please explain what you mean by UMA here. Are you referring to dual GPU systems where both GPUs are by AMD? And the integrated GPU is incorrectly assigned VGA_SWITCHEROO_DIS? Starting with v4.17, the only reason the audio driver registers with vga_switcheroo is because it can be powered down if manual power control is used. However manual power control is not the default, it's primary use case nowadays is MacBook Pros, and there are no MacBook Pros with dual AMD GPUs. So I don't understand what real world use case you're trying to fix. Please explain. > > Newer dGPUs use PCI_CLASS_DISPLAY_OTHER rather than > > PCI_CLASS_DISPLAY_VGA. Make sure we check for both when > > checking for the dGPU in get_bound_vga() Unlike the other changes in your patch, this one would seem perfectly valid to me. However it's a separate issue, so please put it in a separate patch. Instead of adding a check for PCI_CLASS_DISPLAY_OTHER, I'd prefer you replacing the existing check for PCI_CLASS_DISPLAY_VGA with a check for PCI_BASE_CLASS_DISPLAY, like so: - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) + if ((p->class >> 16) == PCI_BASE_CLASS_DISPLAY) Thanks, Lukas

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[alsa-devel] Driver for Maxim MAX9744 Stereo Amplifier
by Tom Rinehart 29 Jun '18

29 Jun '18

I am trying to develop a driver the the Maxim MAX9744 20W Class D Stereo Amplifier. (see https://www.maximintegrated.com/en/products/analog/audio/MAX9744.html <https://www.maximintegrated.com/en/products/analog/audio/MAX9744.html>). It is very similar to the MAX9768 10W Class D Mono amplifier which has a linux driver. (see https://www.maximintegrated.com/en/products/analog/audio/MAX9768.html <https://www.maximintegrated.com/en/products/analog/audio/MAX9768.html>) From a control perspective, both look to be the same from their data sheets. Volume is controlled through i2c, and it has mute and shutdown controlled through gpio. The MAX9744 has a volume control that changes both channels together. When I try to use the MAX9768 driver it shows up as a mono amplifier in alsamixer. What do I need to change in the driver to have it recognize stereo input and output but with the volume control tied together (not independent for each channel)? It seems like I should be able to modify the MAX9768 driver to work with the MAX9744. I have an Adafruit board with the MAX9744 amplifier on it so it is easy to test: https://www.adafruit.com/product/1752 <https://www.adafruit.com/product/1752>.

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[alsa-devel] [PATCH] ASoC: pxa: remove the dmaengine compat need
by Daniel Mack 28 Jun '18

28 Jun '18

From: Robert Jarzmik <robert.jarzmik(a)free.fr> As the pxa architecture switched towards the dmaengine slave map, the old compatibility mechanism to acquire the dma requestor line number and priority are not needed anymore. This patch simplifies the dma resource acquisition, using the more generic function dma_request_slave_channel(). Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Reviewed-by: Daniel Mack <daniel(a)zonque.org> Acked-by: Mark Brown <broonie(a)kernel.org> --- Mark, That's the patch that's Robert says is missing in your tree. As a matter of fact, that's the only reason why I pinged you and asked you to pull it in, because the cleanups for ASoC/pxa I've sent build upon it. Not sure what happened and caused this confusion :-/ Sorry. Thanks, Daniel sound/arm/pxa2xx-ac97.c | 14 ++------------ sound/arm/pxa2xx-pcm-lib.c | 6 +++--- sound/soc/pxa/pxa2xx-ac97.c | 32 +++++--------------------------- sound/soc/pxa/pxa2xx-i2s.c | 6 ++---- 4 files changed, 12 insertions(+), 46 deletions(-) diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 4bc244c40f80..236a63cdaf9f 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -63,28 +63,18 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_legacy_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_out_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_in_req, }; static struct snd_pcm *pxa2xx_ac97_pcm; diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index e8da3b8ee721..dcbe7ecc1835 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -125,9 +125,9 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return snd_dmaengine_pcm_open_request_chan(substream, - pxad_filter_fn, - dma_params->filter_data); + return snd_dmaengine_pcm_open( + substream, dma_request_slave_channel(rtd->cpu_dai->dev, + dma_params->chan_name)); } EXPORT_SYMBOL(__pxa2xx_pcm_open); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 803818aabee9..1b41c0f2a8fb 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -68,61 +68,39 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static struct pxad_param pxa2xx_ac97_pcm_stereo_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 11, -}; - static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_in", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, -}; - -static struct pxad_param pxa2xx_ac97_pcm_stereo_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 12, }; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "pcm_pcm_stereo_out", .maxburst = 32, - .filter_data = &pxa2xx_ac97_pcm_stereo_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_out_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 10, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_out = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_out", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_out_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mono_in_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 9, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_aux_mono_in = { .addr = __PREG(MODR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mono_in", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mono_in_req, }; -static struct pxad_param pxa2xx_ac97_pcm_aux_mic_mono_req = { - .prio = PXAD_PRIO_LOWEST, - .drcmr = 8, -}; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .addr = __PREG(MCDR), .addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES, + .chan_name = "pcm_aux_mic_mono", .maxburst = 16, - .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; static int pxa2xx_ac97_hifi_startup(struct snd_pcm_substream *substream, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 3fb60baf6eab..e7184de0de04 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -82,20 +82,18 @@ static struct pxa_i2s_port pxa_i2s; static struct clk *clk_i2s; static int clk_ena = 0; -static unsigned long pxa2xx_i2s_pcm_stereo_out_req = 3; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_out = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "tx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_out_req, }; -static unsigned long pxa2xx_i2s_pcm_stereo_in_req = 2; static struct snd_dmaengine_dai_dma_data pxa2xx_i2s_pcm_stereo_in = { .addr = __PREG(SADR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, + .chan_name = "rx", .maxburst = 32, - .filter_data = &pxa2xx_i2s_pcm_stereo_in_req, }; static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, -- 2.17.1

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[alsa-devel] [PATCH v2 0/2] Add support for audio on SDM845 SoC
by Rohit kumar 28 Jun '18

28 Jun '18

This provides initial patchset to support audio on Qualcomm Techonologies Inc. SDM845 SoC. Currently, it supports audio playback/capture over Primary MI2S and Quaternary TDM ports. Changes since v1: Addressed comments posted by Vinod and Srinivas. Rohit kumar (2): ASoC: qcom: dt-bindings: Add sdm845 machine bindings ASoC: qcom: add sdm845 sound card support .../devicetree/bindings/sound/qcom,sdm845.txt | 82 ++++ sound/soc/qcom/Kconfig | 9 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sdm845.c | 514 +++++++++++++++++++++ 4 files changed, 607 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 sound/soc/qcom/sdm845.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH] ASoC: dpcm: extend channel merging to the backend cpu dai
by Jerome Brunet 28 Jun '18

28 Jun '18

Extend dpcm_merge_chan to also check backend cpu dai channels capabilities. Apply the same policy as soc_pcm_init_runtime_hw() for multicodec links and only check cpu dai in this case. Cc: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Hi Mark, As you've probably noticed, this patch applies on top Jiada's patch pf4c277b817cc ("ASoC: soc-pcm: DPCM cares BE channel constraint") Regards. Jerome sound/soc/soc-pcm.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b82d9ff1f47e..5141409ccaed 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1728,12 +1728,26 @@ static void dpcm_runtime_base_chan(struct snd_pcm_substream *substream, list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *cpu_dai_drv = be->cpu_dai->driver; struct snd_soc_dai_driver *codec_dai_drv; struct snd_soc_pcm_stream *codec_stream; - int i; + struct snd_soc_pcm_stream *cpu_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + cpu_stream = &cpu_dai_drv->playback; + else + cpu_stream = &cpu_dai_drv->capture; + + *channels_min = max(*channels_min, cpu_stream->channels_min); + *channels_max = min(*channels_max, cpu_stream->channels_max); + + /* + * chan min/max cannot be enforced if there are multiple CODEC + * DAIs connected to a single CPU DAI, use CPU DAI's directly + */ + if (be->num_codecs == 1) { + codec_dai_drv = be->codec_dais[0]->driver; - for (i = 0; i < be->num_codecs; i++) { - codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; else -- 2.14.4

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Re: [alsa-devel] [PATCH] vgaswitchroo: set audio client id according to bound gpu client id
by Alex Deucher 28 Jun '18

28 Jun '18

On Thu, Jun 28, 2018 at 2:22 AM, Jim Qu <Jim.Qu(a)amd.com> wrote: > On modern laptop, there are more and more platforms > have two GPUs, and each of them maybe have audio codec > for HDMP/DP output. For some dGPU which is no output, > audio codec usually is disabled. > > In currect HDA audio driver, it will set all codec as > VGA_SWITCHEROO_DIS, so if system runtime pm is enabled, > the audio which is binded to UMA will be suspended. > > In HDA driver side, it is difficult to know which GPU > the audio has binded to. So set the bound gpu pci dev > to vgaswitchroo, let vgaswitchroo make decision. > > Newer dGPUs use PCI_CLASS_DISPLAY_OTHER rather than > PCI_CLASS_DISPLAY_VGA. Make sure we check for both when > checking for the dGPU in get_bound_vga() > > The patch also combine the HDA change to avoid break > building. > > Change-Id: I9906c1bd4dd5b36108d7133bb1cf724d13f1cd6d > Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> > Signed-off-by: Jim Qu <Jim.Qu(a)amd.com> > --- > drivers/gpu/vga/vga_switcheroo.c | 11 +++++++++-- > include/linux/vga_switcheroo.h | 4 ++-- > sound/pci/hda/hda_intel.c | 15 +++++++++------ > 3 files changed, 20 insertions(+), 10 deletions(-) > Please resend and send it to dri-devel and alsa-devel(a)alsa-project.org since this patch touches multiple subsystems. I would propose we merge this through the drm tree if possible. Alex > diff --git a/drivers/gpu/vga/vga_switcheroo.c b/drivers/gpu/vga/vga_switcheroo.c > index fc4adf3..7b7b0fd 100644 > --- a/drivers/gpu/vga/vga_switcheroo.c > +++ b/drivers/gpu/vga/vga_switcheroo.c > @@ -329,7 +329,7 @@ EXPORT_SYMBOL(vga_switcheroo_register_client); > * vga_switcheroo_register_audio_client - register audio client > * @pdev: client pci device > * @ops: client callbacks > - * @id: client identifier > + * @bound_vga: client bound vga pci device > * > * Register audio client (audio device on a GPU). The client is assumed > * to use runtime PM. Beforehand, vga_switcheroo_client_probe_defer() > @@ -339,8 +339,15 @@ EXPORT_SYMBOL(vga_switcheroo_register_client); > */ > int vga_switcheroo_register_audio_client(struct pci_dev *pdev, > const struct vga_switcheroo_client_ops *ops, > - enum vga_switcheroo_client_id id) > + struct pci_dev *bound_vga) > { > + enum vga_switcheroo_client_id id = VGA_SWITCHEROO_DIS; > + > + if (bound_vga) { > + if (vgasr_priv.handler->get_client_id) > + id = vgasr_priv.handler->get_client_id(bound_vga); > + } > + > return register_client(pdev, ops, id | ID_BIT_AUDIO, false, true); > } > EXPORT_SYMBOL(vga_switcheroo_register_audio_client); > diff --git a/include/linux/vga_switcheroo.h b/include/linux/vga_switcheroo.h > index 77f0f0a..a0c7b86 100644 > --- a/include/linux/vga_switcheroo.h > +++ b/include/linux/vga_switcheroo.h > @@ -151,7 +151,7 @@ int vga_switcheroo_register_client(struct pci_dev *dev, > bool driver_power_control); > int vga_switcheroo_register_audio_client(struct pci_dev *pdev, > const struct vga_switcheroo_client_ops *ops, > - enum vga_switcheroo_client_id id); > + struct pci_dev *bound_vga); > > void vga_switcheroo_client_fb_set(struct pci_dev *dev, > struct fb_info *info); > @@ -180,7 +180,7 @@ static inline int vga_switcheroo_register_handler(const struct vga_switcheroo_ha > enum vga_switcheroo_handler_flags_t handler_flags) { return 0; } > static inline int vga_switcheroo_register_audio_client(struct pci_dev *pdev, > const struct vga_switcheroo_client_ops *ops, > - enum vga_switcheroo_client_id id) { return 0; } > + struct pci_dev *bound_vga) { return 0; } > static inline void vga_switcheroo_unregister_handler(void) {} > static inline enum vga_switcheroo_handler_flags_t vga_switcheroo_handler_flags(void) { return 0; } > static inline int vga_switcheroo_lock_ddc(struct pci_dev *pdev) { return -ENODEV; } > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index 1ae1850..8f992e6 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1319,15 +1319,17 @@ static const struct vga_switcheroo_client_ops azx_vs_ops = { > static int register_vga_switcheroo(struct azx *chip) > { > struct hda_intel *hda = container_of(chip, struct hda_intel, chip); > + struct pci_dev *p; > int err; > > if (!hda->use_vga_switcheroo) > return 0; > - /* FIXME: currently only handling DIS controller > - * is there any machine with two switchable HDMI audio controllers? > - */ > - err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, > - VGA_SWITCHEROO_DIS); > + > + p = get_bound_vga(chip->pci); > + err = vga_switcheroo_register_audio_client(chip->pci, &azx_vs_ops, p); > + > + if (p) > + pci_dev_put(p); > if (err < 0) > return err; > hda->vga_switcheroo_registered = 1; > @@ -1429,7 +1431,8 @@ static struct pci_dev *get_bound_vga(struct pci_dev *pci) > p = pci_get_domain_bus_and_slot(pci_domain_nr(pci->bus), > pci->bus->number, 0); > if (p) { > - if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA) > + if ((p->class >> 8) == PCI_CLASS_DISPLAY_VGA || > + (p->class >> 8) == PCI_CLASS_DISPLAY_OTHER) > return p; > pci_dev_put(p); > } > -- > 2.7.4 > > _______________________________________________ > amd-gfx mailing list > amd-gfx(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/amd-gfx

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[alsa-devel] [PATCH v3 00/14] ARM: pxa: switch to DMA slave maps
by Robert Jarzmik 28 Jun '18

28 Jun '18

As I gathered almost all the required acks, this is an information only post before queuing to the PXA tree. The only missing part is for netdev, for which I'd like an ack from netdev people for patches 0007 and 0008, but that won't prevent me from queuing all the other patches (excepting the patch 0012 which can only be applied once 0007 and 0008 are queued). Cheers. -- Robert Robert Jarzmik (14): dmaengine: pxa: use a dma slave map ARM: pxa: add dma slave map dmaengine: pxa: add a default requestor policy mmc: pxamci: remove the dmaengine compat need media: pxa_camera: remove the dmaengine compat need mtd: rawnand: marvell: remove the dmaengine compat need net: smc911x: remove the dmaengine compat need net: smc91x: remove the dmaengine compat need ASoC: pxa: remove the dmaengine compat need ata: pata_pxa: remove the dmaengine compat need dmaengine: pxa: document pxad_param dmaengine: pxa: make the filter function internal ARM: pxa: remove the DMA IO resources ARM: pxa: change SSP DMA channels allocation arch/arm/mach-pxa/devices.c | 148 +--------------------------------- arch/arm/mach-pxa/devices.h | 6 +- arch/arm/mach-pxa/pxa25x.c | 38 ++++++++- arch/arm/mach-pxa/pxa27x.c | 39 ++++++++- arch/arm/mach-pxa/pxa3xx.c | 41 +++++++++- arch/arm/plat-pxa/ssp.c | 47 ----------- drivers/ata/pata_pxa.c | 10 +-- drivers/dma/pxa_dma.c | 18 ++++- drivers/media/platform/pxa_camera.c | 22 +---- drivers/mmc/host/pxamci.c | 29 +------ drivers/mtd/nand/raw/marvell_nand.c | 17 +--- drivers/net/ethernet/smsc/smc911x.c | 13 +-- drivers/net/ethernet/smsc/smc91x.c | 9 +-- drivers/net/ethernet/smsc/smc91x.h | 1 - include/linux/dma/pxa-dma.h | 20 +++-- include/linux/platform_data/mmp_dma.h | 4 + include/linux/pxa2xx_ssp.h | 2 - sound/arm/pxa2xx-ac97.c | 14 +--- sound/arm/pxa2xx-pcm-lib.c | 6 +- sound/soc/pxa/pxa-ssp.c | 5 +- sound/soc/pxa/pxa2xx-ac97.c | 32 ++------ sound/soc/pxa/pxa2xx-i2s.c | 6 +- 22 files changed, 176 insertions(+), 351 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH v2] ASoC: dpcm: improve runtime update predictability
by Jerome Brunet 28 Jun '18

28 Jun '18

As it is, dpcm_runtime_update() performs the old path and new path update of a frontend before going on to the next frontend DAI. Depending the order of the FEs within the rtd list, the result of the update might be different. For example: * Frontend A connected to backend C, with a 48kHz playback * Frontend B connected to backend D, with a 44.1kHz playback * FE A appears before FE B in the rtd list of the card. If we reparent BE C to FE B (disconnecting BE D): * old path update of FE A will run first, and BE C will get hw_free() and shutdown() * new path update of FE B will run after and BE C, which is stopped, so it will be configured at 44.1kHz, as expected If we reparent BE D to FE A (disconnecting BE C): * new path update of FE A will run first but since BE D is still running at 44.1kHz, it won't be reconfigured (no call to startup() or hw_params()) * old path update of FE B runs after, nothing happens * In this case, we end up with a BE playing at 44.1kHz a stream which is supposed to be played at 48Khz (too slow) To improve this situation, this patch performs all the FE old paths update before going on to update the new paths. With this, the result should no longer depend on the order of the FE within the card rtd list. Please note that there might be a small performance penalty since dpcm_process_paths() is called twice per stream direction. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Changes since v1 [0]: * Add missing static as reported by kbuild robot. [0]: https://lkml.kernel.org/r/20180625082923.25636-1-jbrunet@baylibre.com sound/soc/soc-pcm.c | 165 +++++++++++++++++++++++++++------------------------- 1 file changed, 86 insertions(+), 79 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5e7ae47a9658..4eda18cfe895 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2543,106 +2543,113 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) return ret; } -/* Called by DAPM mixer/mux changes to update audio routing between PCMs and - * any DAI links. - */ -int soc_dpcm_runtime_update(struct snd_soc_card *card) +static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { - struct snd_soc_pcm_runtime *fe; - int old, new, paths; + struct snd_soc_dapm_widget_list *list; + int count, paths; - mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - list_for_each_entry(fe, &card->rtd_list, list) { - struct snd_soc_dapm_widget_list *list; + if (!fe->dai_link->dynamic) + return 0; - /* make sure link is FE */ - if (!fe->dai_link->dynamic) - continue; + /* only check active links */ + if (!fe->cpu_dai->active) + return 0; - /* only check active links */ - if (!fe->cpu_dai->active) - continue; + /* DAPM sync will call this to update DSP paths */ + dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", + new ? "new" : "old", fe->dai_link->name); - /* DAPM sync will call this to update DSP paths */ - dev_dbg(fe->dev, "ASoC: DPCM runtime update for FE %s\n", - fe->dai_link->name); + /* skip if FE doesn't have playback capability */ + if (!fe->cpu_dai->driver->playback.channels_min || + !fe->codec_dai->driver->playback.channels_min) + goto capture; - /* skip if FE doesn't have playback capability */ - if (!fe->cpu_dai->driver->playback.channels_min - || !fe->codec_dai->driver->playback.channels_min) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active - || !fe->codec_dai->playback_active) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - mutex_unlock(&card->mutex); - return paths; - } + /* skip if FE isn't currently playing */ + if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + goto capture; - /* update any new playback paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 1); - if (new) { - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "playback"); + return paths; + } - /* update any old playback paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, 0); - if (old) { + /* update any playback paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); + } + + dpcm_path_put(&list); + capture: - /* skip if FE doesn't have capture capability */ - if (!fe->cpu_dai->driver->capture.channels_min - || !fe->codec_dai->driver->capture.channels_min) - continue; + /* skip if FE doesn't have capture capability */ + if (!fe->cpu_dai->driver->capture.channels_min || + !fe->codec_dai->driver->capture.channels_min) + return 0; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active - || !fe->codec_dai->capture_active) - continue; + /* skip if FE isn't currently capturing */ + if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + return 0; - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - mutex_unlock(&card->mutex); - return paths; - } + paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, "capture"); + return paths; + } - /* update any new capture paths */ - new = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 1); - if (new) { + /* update any old capture paths */ + count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); + if (count) { + if (new) dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - - /* update any old capture paths */ - old = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, 0); - if (old) { + else dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); - } - dpcm_path_put(&list); + dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); } - mutex_unlock(&card->mutex); + dpcm_path_put(&list); + return 0; } + +/* Called by DAPM mixer/mux changes to update audio routing between PCMs and + * any DAI links. + */ +int soc_dpcm_runtime_update(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *fe; + int ret = 0; + + mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* shutdown all old paths first */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 0); + if (ret) + goto out; + } + + /* bring new paths up */ + list_for_each_entry(fe, &card->rtd_list, list) { + ret = soc_dpcm_fe_runtime_update(fe, 1); + if (ret) + goto out; + } + +out: + mutex_unlock(&card->mutex); + return ret; +} int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) { struct snd_soc_dpcm *dpcm; -- 2.14.4

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[alsa-devel] [PATCH] ASoC: qdsp6: qdafe: fix some off by one bugs
by Dan Carpenter 28 Jun '18

28 Jun '18

The > should be >= or we could read one element beyond the end of the port_maps[] array. Fixes: 7fa2d70f9766 ("ASoC: qdsp6: q6afe: Add q6afe driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 01f43218984b..69a7896cb713 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -777,7 +777,7 @@ static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data) */ int q6afe_get_port_id(int index) { - if (index < 0 || index > AFE_PORT_MAX) + if (index < 0 || index >= AFE_PORT_MAX) return -EINVAL; return port_maps[index].port_id; @@ -1014,7 +1014,7 @@ int q6afe_port_stop(struct q6afe_port *port) port_id = port->id; index = port->token; - if (index < 0 || index > AFE_PORT_MAX) { + if (index < 0 || index >= AFE_PORT_MAX) { dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); return -EINVAL; } @@ -1355,7 +1355,7 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) unsigned long flags; int cfg_type; - if (id < 0 || id > AFE_PORT_MAX) { + if (id < 0 || id >= AFE_PORT_MAX) { dev_err(dev, "AFE port token[%d] invalid!\n", id); return ERR_PTR(-EINVAL); }

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[alsa-devel] [PATCH] ASoC: dpcm: don't merge format from invalid codec dai
by Jerome Brunet 28 Jun '18

28 Jun '18

When merging codec formats, dpcm_runtime_base_format() should skip the codecs which are not supporting the current stream direction. At the moment, if a BE link has more than one codec, and only one of these codecs has no capture DAI, it becomes impossible to start a capture stream because the merged format would be 0. Skipping invalid codec DAI solves the problem. Fixes: b073ed4e2126 ("ASoC: soc-pcm: DPCM cares BE format") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- 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 5141409ccaed..a58d8e7bf52b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1694,6 +1694,14 @@ static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) int i; for (i = 0; i < be->num_codecs; i++) { + /* + * Skip CODECs which don't support the current stream + * type. See soc_pcm_init_runtime_hw() for more details + */ + if (!snd_soc_dai_stream_valid(be->codec_dais[i], + stream)) + continue; + codec_dai_drv = be->codec_dais[i]->driver; if (stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; -- 2.14.4

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