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

[alsa-devel] [PATCH] ALSA: usb-audio: Simplify PCM open/close callbacks
by Takashi Iwai 27 May '18

27 May '18

The stream direction in open and close callbacks can be retrieved from substream->direction, hence we don't have to stick with the unique PCM ops hard-coded for each direction. Rewrite the common open/close callback functions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 34 ++++++++-------------------------- 1 file changed, 8 insertions(+), 26 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 20bed1c7a312..d5b9c30d3bb1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1227,8 +1227,9 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre return err; } -static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_open(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; @@ -1248,8 +1249,9 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) return setup_hw_info(runtime, subs); } -static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) +static int snd_usb_pcm_close(struct snd_pcm_substream *substream) { + int direction = substream->stream; struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; @@ -1611,26 +1613,6 @@ static void retire_playback_urb(struct snd_usb_substream *subs, spin_unlock_irqrestore(&subs->lock, flags); } -static int snd_usb_playback_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_playback_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_PLAYBACK); -} - -static int snd_usb_capture_open(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_open(substream, SNDRV_PCM_STREAM_CAPTURE); -} - -static int snd_usb_capture_close(struct snd_pcm_substream *substream) -{ - return snd_usb_pcm_close(substream, SNDRV_PCM_STREAM_CAPTURE); -} - static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -1692,8 +1674,8 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream } static const struct snd_pcm_ops snd_usb_playback_ops = { - .open = snd_usb_playback_open, - .close = snd_usb_playback_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, @@ -1705,8 +1687,8 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { }; static const struct snd_pcm_ops snd_usb_capture_ops = { - .open = snd_usb_capture_open, - .close = snd_usb_capture_close, + .open = snd_usb_pcm_open, + .close = snd_usb_pcm_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_usb_hw_params, .hw_free = snd_usb_hw_free, -- 2.16.3

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[alsa-devel] [PATCH] ALSA: aloop: Reduced duplicated PCM ops definition
by Takashi Iwai 27 May '18

27 May '18

The PCM ops defined for playback and capture are identical. Just use the single one for both. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/aloop.c | 19 +++---------------- 1 file changed, 3 insertions(+), 16 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index eab7f594ebe7..78a2fdc38531 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -768,20 +768,7 @@ static int loopback_close(struct snd_pcm_substream *substream) return 0; } -static const struct snd_pcm_ops loopback_playback_ops = { - .open = loopback_open, - .close = loopback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = loopback_hw_params, - .hw_free = loopback_hw_free, - .prepare = loopback_prepare, - .trigger = loopback_trigger, - .pointer = loopback_pointer, - .page = snd_pcm_lib_get_vmalloc_page, - .mmap = snd_pcm_lib_mmap_vmalloc, -}; - -static const struct snd_pcm_ops loopback_capture_ops = { +static const struct snd_pcm_ops loopback_pcm_ops = { .open = loopback_open, .close = loopback_close, .ioctl = snd_pcm_lib_ioctl, @@ -804,8 +791,8 @@ static int loopback_pcm_new(struct loopback *loopback, substreams, substreams, &pcm); if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_capture_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &loopback_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &loopback_pcm_ops); pcm->private_data = loopback; pcm->info_flags = 0; -- 2.16.3

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[alsa-devel] [PATCH] ASoC: AMD: make channel 1 dma as circular
by Akshu Agrawal 27 May '18

27 May '18

channel 1: SYSMEM<->ACP channel 2: ACP<->I2S Instead of waiting on period interrupt of ch 2 and then starting dma on ch1, we make ch1 dma as circular. This removes dependency of period granularity on hw pointer. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 72 +++++++-------------------------------------- 1 file changed, 10 insertions(+), 62 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index ac32dea..f1d8678 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -337,8 +337,7 @@ static void config_acp_dma(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, - u16 ch_num, bool is_circular) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) { u32 dma_ctrl; @@ -369,11 +368,8 @@ static void acp_dma_start(void __iomem *acp_mmio, break; } - /* enable for ACP SRAM to/from I2S DMA channel */ - if (is_circular == true) - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; - else - dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* circular for both DMA channel */ + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -617,7 +613,6 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { - u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -634,33 +629,13 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(ACP_TO_I2S_DMA_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_13) == - PLAYBACK_START_DMA_DESCR_CH13) - dscr_idx = PLAYBACK_END_DMA_DESCR_CH12; - else - dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - config_acp_dma_channel(acp_mmio, SYSRAM_TO_ACP_CH_NUM, dscr_idx, - 1, 0); - acp_dma_start(acp_mmio, SYSRAM_TO_ACP_CH_NUM, false); - snd_pcm_period_elapsed(irq_data->play_i2ssp_stream); - acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_9) == - PLAYBACK_START_DMA_DESCR_CH9) - dscr_idx = PLAYBACK_END_DMA_DESCR_CH8; - else - dscr_idx = PLAYBACK_START_DMA_DESCR_CH8; - config_acp_dma_channel(acp_mmio, - SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, - dscr_idx, 1, 0); - acp_dma_start(acp_mmio, SYSRAM_TO_ACP_BT_INSTANCE_CH_NUM, - false); snd_pcm_period_elapsed(irq_data->play_i2sbt_stream); acp_reg_write((intr_flag & BIT(ACP_TO_I2S_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -669,38 +644,20 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_15) == - CAPTURE_START_DMA_DESCR_CH15) - dscr_idx = CAPTURE_END_DMA_DESCR_CH14; - else - dscr_idx = CAPTURE_START_DMA_DESCR_CH14; - config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, - 1, 0); - acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); - + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) != 0) { valid_irq = true; - snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); } if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; - if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_11) == - CAPTURE_START_DMA_DESCR_CH11) - dscr_idx = CAPTURE_END_DMA_DESCR_CH10; - else - dscr_idx = CAPTURE_START_DMA_DESCR_CH10; - config_acp_dma_channel(acp_mmio, - ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, - dscr_idx, 1, 0); - acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, - false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -708,7 +665,6 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; - snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -1015,11 +971,9 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; - u32 loops = 4000; u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *prtd = substream->private_data; struct audio_substream_data *rtd = runtime->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); @@ -1034,18 +988,12 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) if (rtd->bytescount == 0) rtd->bytescount = bytescount; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_start(rtd->acp_mmio, rtd->ch1, false); - while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & - BIT(rtd->ch1)) { - if (!loops--) { - dev_err(component->dev, - "acp dma start timeout\n"); - return -ETIMEDOUT; - } - cpu_relax(); - } + acp_dma_start(rtd->acp_mmio, rtd->ch1); + acp_dma_start(rtd->acp_mmio, rtd->ch2); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch2); + acp_dma_start(rtd->acp_mmio, rtd->ch1); } - acp_dma_start(rtd->acp_mmio, rtd->ch2, true); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- 1.9.1

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[alsa-devel] [PATCH 0/2] ALSA: dice: optimization for TC Electronic Studio Konnekt 48
by Takashi Sakamoto 27 May '18

27 May '18

Hi, This patchset adds optimization to current ALSA dice driver for Studio Konnekt 48 produced by TC Electronic 2007. My recent work adds hard-coded stream format parameters for all of models produced by TC Electronic with ASICs of Dice series: * Konnekt 24D * Konnect 8 * Studio Konnekt 48 * Konnekt Live * Desktop Konnekt 6 * Impact Twin * Digital Konnekt 32 (Series of PowerCore is not based on Dice.) Except for Konnekt Live, I actually test with these models, and realized that current ALSA dice driver brings several issues, mainly with noises but with detection of packet discontinuities and jumbo payload for some models and transmission frequencies. I guess they comes from timestamping quirks which I reported: [alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6 http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107715.html I have never found any good solution for this issue. Takashi Sakamoto (2): ALSA: dice: fix stream format parameters for TC Electronic Studio Konnekt 48 ALSA: dice: unuse second stream for MIDI conformant data channel for TC Electronic models sound/firewire/dice/dice-tcelectronic.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) -- 2.14.1

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[alsa-devel] [PATCH] ALSA: seq: fix spelling mistake "Unamed" -> "Unnamed"
by Colin King 27 May '18

27 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in string Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/seq/seq_ports.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index d21ece9f8d73..24d90abfc64d 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -669,7 +669,7 @@ int snd_seq_event_port_attach(int client, /* Set up the port */ memset(&portinfo, 0, sizeof(portinfo)); portinfo.addr.client = client; - strlcpy(portinfo.name, portname ? portname : "Unamed port", + strlcpy(portinfo.name, portname ? portname : "Unnamed port", sizeof(portinfo.name)); portinfo.capability = cap; -- 2.17.0

2 1

[alsa-devel] [RFC/RFT PATCH] ASoC: topology: Improve backwards compatibility with v4 topology files
by Guenter Roeck 25 May '18

25 May '18

From: Guenter Roeck <groeck(a)chromium.org> Commit dc31e741db49 ("ASoC: topology: ABI - Add the types for BE DAI") introduced sound topology files version 5. Initially, this change made the topology code incompatible with v4 topology files. Backwards compatibility with v4 configuration files was subsequently added with commit 288b8da7e992 ("ASoC: topology: Support topology file of ABI v4"). Unfortunately, backwards compatibility was never fully implemented. First, the manifest size in (Skylake) v4 configuration files is set to 0, which causes manifest_new_ver() to bail out with error messages similar to the following. snd_soc_skl 0000:00:1f.3: ASoC: invalid manifest size snd_soc_skl 0000:00:1f.3: tplg component load failed-22 snd_soc_skl 0000:00:1f.3: Failed to init topology! snd_soc_skl 0000:00:1f.3: ASoC: failed to probe component -22 skl_n88l25_m98357a skl_n88l25_m98357a: ASoC: failed to instantiate card -22 skl_n88l25_m98357a: probe of skl_n88l25_m98357a failed with error -22 After this problem is fixed, the following error message is seen instead. snd_soc_skl 0000:00:1f.3: ASoC: old version of manifest snd_soc_skl 0000:00:1f.3: Invalid descriptor token 1093938482 snd_soc_skl 0000:00:1f.3: ASoC: failed to load widget media0_in cpr 0 snd_soc_skl 0000:00:1f.3: tPlg component load failed-22 This message is seen because backwards compatibility for loading widgets was never implemented. Both problems have been widely reported. Various attempts were made to fix the problem, usually by providing v5 configuration files. However, all those attempts result in incomplete ASoC configuration. Let's implement backward compatibility properly to solve the problem for good. Signed-off-by: Guenter Roeck <groeck(a)chromium.org> --- Tested on Caroline (Samsung Chromebook Pro) running v4.17-rc6 plus this patch, with original (v4) configuration file. Also tested on several other Chromebooks with this patch on top of chromeos-4.14. Additional real world test coverage would be useful before moving forward. sound/soc/intel/skylake/skl-topology.c | 235 +++++++++++++++++++++++++ sound/soc/soc-topology.c | 7 +- 2 files changed, 240 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 3b1dca419883..cc81e200e6b7 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -19,6 +19,7 @@ #include <linux/slab.h> #include <linux/types.h> #include <linux/firmware.h> +#include <linux/uuid.h> #include <sound/soc.h> #include <sound/soc-topology.h> #include <uapi/sound/snd_sst_tokens.h> @@ -30,6 +31,79 @@ #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +/* v4 configuration data */ +struct skl_dfw_v4_module_pin { + u16 module_id; + u16 instance_id; +} __packed; + +struct skl_dfw_v4_module_fmt { + u32 channels; + u32 freq; + u32 bit_depth; + u32 valid_bit_depth; + u32 ch_cfg; + u32 interleaving_style; + u32 sample_type; + u32 ch_map; +} __packed; + +struct skl_dfw_v4_module_caps { + u32 set_params:2; + u32 rsvd:30; + u32 param_id; + u32 caps_size; + u32 caps[HDA_SST_CFG_MAX]; +}; + +struct skl_dfw_v4_pipe { + u8 pipe_id; + u8 pipe_priority; + u16 conn_type:4; + u16 rsvd:4; + u16 memory_pages:8; +} __packed; + +struct skl_dfw_v4_module { + char uuid[SKL_UUID_STR_SZ]; + + u16 module_id; + u16 instance_id; + u32 max_mcps; + u32 mem_pages; + u32 obs; + u32 ibs; + u32 vbus_id; + + u32 max_in_queue:8; + u32 max_out_queue:8; + u32 time_slot:8; + u32 core_id:4; + u32 rsvd1:4; + + u32 module_type:8; + u32 conn_type:4; + u32 dev_type:4; + u32 hw_conn_type:4; + u32 rsvd2:12; + + u32 params_fixup:8; + u32 converter:8; + u32 input_pin_type:1; + u32 output_pin_type:1; + u32 is_dynamic_in_pin:1; + u32 is_dynamic_out_pin:1; + u32 is_loadable:1; + u32 rsvd3:11; + + struct skl_dfw_v4_pipe pipe; + struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE]; + struct skl_dfw_v4_module_fmt out_fmt[MAX_OUT_QUEUE]; + struct skl_dfw_v4_module_pin in_pin[MAX_IN_QUEUE]; + struct skl_dfw_v4_module_pin out_pin[MAX_OUT_QUEUE]; + struct skl_dfw_v4_module_caps caps; +} __packed; + #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) #define SKL_FMT_FIXUP_MASK (1 << 2) @@ -2724,6 +2798,160 @@ static int skl_tplg_get_desc_blocks(struct device *dev, return -EINVAL; } +/* + * Add pipeline from topology binary into driver pipeline list + * + * If already added we return that instance + * Otherwise we create a new instance and add into driver list + */ +static int skl_tplg_add_pipe_v4(struct device *dev, + struct skl_module_cfg *mconfig, struct skl *skl, + struct skl_dfw_v4_pipe *dfw_pipe) +{ + struct skl_pipeline *ppl; + struct skl_pipe *pipe; + struct skl_pipe_params *params; + + list_for_each_entry(ppl, &skl->ppl_list, node) { + if (ppl->pipe->ppl_id == dfw_pipe->pipe_id) { + mconfig->pipe = ppl->pipe; + return 0; + } + } + + ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL); + if (!ppl) + return -ENOMEM; + + pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL); + if (!pipe) + return -ENOMEM; + + params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL); + if (!params) + return -ENOMEM; + + pipe->ppl_id = dfw_pipe->pipe_id; + pipe->memory_pages = dfw_pipe->memory_pages; + pipe->pipe_priority = dfw_pipe->pipe_priority; + pipe->conn_type = dfw_pipe->conn_type; + pipe->state = SKL_PIPE_INVALID; + pipe->p_params = params; + INIT_LIST_HEAD(&pipe->w_list); + + ppl->pipe = pipe; + list_add(&ppl->node, &skl->ppl_list); + + mconfig->pipe = pipe; + mconfig->pipe->state = SKL_PIPE_INVALID; + + return 0; +} + +static void skl_fill_module_pin_info_v4(struct skl_dfw_v4_module_pin *dfw_pin, + struct skl_module_pin *m_pin, + bool is_dynamic, int max_pin) +{ + int i; + + for (i = 0; i < max_pin; i++) { + m_pin[i].id.module_id = dfw_pin[i].module_id; + m_pin[i].id.instance_id = dfw_pin[i].instance_id; + m_pin[i].in_use = false; + m_pin[i].is_dynamic = is_dynamic; + m_pin[i].pin_state = SKL_PIN_UNBIND; + } +} + +static void skl_tplg_fill_fmt_v4(struct skl_module_pin_fmt *dst_fmt, + struct skl_dfw_v4_module_fmt *src_fmt, + int pins) +{ + int i; + + for (i = 0; i < pins; i++) { + dst_fmt[i].fmt.channels = src_fmt[i].channels; + dst_fmt[i].fmt.s_freq = src_fmt[i].freq; + dst_fmt[i].fmt.bit_depth = src_fmt[i].bit_depth; + dst_fmt[i].fmt.valid_bit_depth = src_fmt[i].valid_bit_depth; + dst_fmt[i].fmt.ch_cfg = src_fmt[i].ch_cfg; + dst_fmt[i].fmt.ch_map = src_fmt[i].ch_map; + dst_fmt[i].fmt.interleaving_style = + src_fmt[i].interleaving_style; + dst_fmt[i].fmt.sample_type = src_fmt[i].sample_type; + } +} + +static int skl_tplg_get_pvt_data_v4(struct snd_soc_tplg_dapm_widget *tplg_w, + struct skl *skl, struct device *dev, + struct skl_module_cfg *mconfig) +{ + struct skl_dfw_v4_module *dfw = + (struct skl_dfw_v4_module *)tplg_w->priv.data; + int ret; + + dev_dbg(dev, "Parsing Skylake v4 widget topology data\n"); + + ret = guid_parse(dfw->uuid, (guid_t *)mconfig->guid); + if (ret) + return ret; + mconfig->id.module_id = -1; + mconfig->id.instance_id = dfw->instance_id; + mconfig->module->resources[0].cps = dfw->max_mcps; + mconfig->module->resources[0].ibs = dfw->ibs; + mconfig->module->resources[0].obs = dfw->obs; + mconfig->core_id = dfw->core_id; + mconfig->module->max_input_pins = dfw->max_in_queue; + mconfig->module->max_output_pins = dfw->max_out_queue; + mconfig->module->loadable = dfw->is_loadable; + skl_tplg_fill_fmt_v4(mconfig->module->formats[0].inputs, dfw->in_fmt, + MAX_IN_QUEUE); + skl_tplg_fill_fmt_v4(mconfig->module->formats[0].outputs, dfw->out_fmt, + MAX_OUT_QUEUE); + + mconfig->params_fixup = dfw->params_fixup; + mconfig->converter = dfw->converter; + mconfig->m_type = dfw->module_type; + mconfig->vbus_id = dfw->vbus_id; + mconfig->module->resources[0].is_pages = dfw->mem_pages; + + ret = skl_tplg_add_pipe_v4(dev, mconfig, skl, &dfw->pipe); + if (ret) + return ret; + + mconfig->dev_type = dfw->dev_type; + mconfig->hw_conn_type = dfw->hw_conn_type; + mconfig->time_slot = dfw->time_slot; + mconfig->formats_config.caps_size = dfw->caps.caps_size; + + mconfig->m_in_pin = devm_kzalloc(dev, + MAX_IN_QUEUE * sizeof(*mconfig->m_in_pin), + GFP_KERNEL); + if (!mconfig->m_in_pin) + return -ENOMEM; + + mconfig->m_out_pin = devm_kzalloc(dev, + MAX_OUT_QUEUE * sizeof(*mconfig->m_out_pin), + GFP_KERNEL); + if (!mconfig->m_out_pin) + return -ENOMEM; + + skl_fill_module_pin_info_v4(dfw->in_pin, mconfig->m_in_pin, + dfw->is_dynamic_in_pin, + mconfig->module->max_input_pins); + skl_fill_module_pin_info_v4(dfw->out_pin, mconfig->m_out_pin, + dfw->is_dynamic_out_pin, + mconfig->module->max_output_pins); + + if (mconfig->formats_config.caps_size) { + dev_warn(dev, "caps size %d not supported, will be ignored\n", + mconfig->formats_config.caps_size); + mconfig->formats_config.caps_size = 0; + } + + return 0; +} + /* * Parse the private data for the token and corresponding value. * The private data can have multiple data blocks. So, a data block @@ -2739,6 +2967,13 @@ static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, char *data; int ret; + /* + * v4 configuration files have a valid UUID at the start of + * the widget's private data. + */ + if (uuid_is_valid((char *)tplg_w->priv.data)) + return skl_tplg_get_pvt_data_v4(tplg_w, skl, dev, mconfig); + /* Read the NUM_DATA_BLOCKS descriptor */ array = (struct snd_soc_tplg_vendor_array *)tplg_w->priv.data; ret = skl_tplg_get_desc_blocks(dev, array); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 986b8b2f90fb..d66b2e5ccd67 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2293,8 +2293,11 @@ static int manifest_new_ver(struct soc_tplg *tplg, *manifest = NULL; if (src->size != sizeof(*src_v4)) { - dev_err(tplg->dev, "ASoC: invalid manifest size\n"); - return -EINVAL; + dev_warn(tplg->dev, "ASoC: invalid manifest size %d\n", + src->size); + if (src->size) + return -EINVAL; + src->size = sizeof(*src_v4); } dev_warn(tplg->dev, "ASoC: old version of manifest\n"); -- 2.17.0.441.gb46fe60e1d-goog

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[alsa-devel] [PATCH 1/3] ASoC: mediatek: add sub dai to mtk_base_afe
by KaiChieh Chuang 25 May '18

25 May '18

In MediaTek SoC chip we have multiple DAI, such as I2S, ADDA, PCM, etc. Organize each DAI in to one sub dai, with its dai driver, controls, widgets, routes. add mtk_afe_combine_sub_dai() to combine dai driver from each DAI. add mtk_afe_add_sub_dai_control() to register the control, widget, routes to component. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 78 ++++++++++++++++++++++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 5 ++ sound/soc/mediatek/common/mtk-base-afe.h | 18 +++++ 3 files changed, 101 insertions(+) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 8966ee1..c449188 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -13,6 +13,84 @@ #include "mtk-afe-platform-driver.h" #include "mtk-base-afe.h" +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) +{ + struct snd_soc_dai_driver *sub_dai_drivers; + size_t num_dai_drivers = 0, dai_idx = 0; + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + /* calcualte total dai driver size */ + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) + num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + } + + dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); + + /* combine sub_dais */ + afe->num_dai_drivers = num_dai_drivers; + afe->dai_drivers = devm_kcalloc(afe->dev, + num_dai_drivers, + sizeof(struct snd_soc_dai_driver), + GFP_KERNEL); + if (!afe->dai_drivers) + return -ENOMEM; + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) { + sub_dai_drivers = afe->sub_dais[i].dai_drivers; + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + sub_dai_drivers, + afe->sub_dais[i].num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += afe->sub_dais[i].num_dai_drivers; + } + } + + return 0; +} + +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) +{ + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].controls) + snd_soc_add_component_controls(component, + afe->sub_dais[i].controls, + afe->sub_dais[i].num_controls); + + if (afe->sub_dais[i].dapm_widgets) + snd_soc_dapm_new_controls(&component->dapm, + afe->sub_dais[i].dapm_widgets, + afe->sub_dais[i].num_dapm_widgets); + + if (afe->sub_dais[i].dapm_routes) + snd_soc_dapm_add_routes(&component->dapm, + afe->sub_dais[i].dapm_routes, + afe->sub_dais[i].num_dapm_routes); + } + + snd_soc_dapm_new_widgets(component->dapm.card); + + return 0; + +} + static snd_pcm_uframes_t mtk_afe_pcm_pointer (struct snd_pcm_substream *substream) { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.h b/sound/soc/mediatek/common/mtk-afe-platform-driver.h index 1c81d91..0c31fa4 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.h +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.h @@ -12,5 +12,10 @@ #define AFE_PCM_NAME "mtk-afe-pcm" extern const struct snd_soc_component_driver mtk_afe_pcm_platform; +struct mtk_base_afe; +struct snd_soc_component; + +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe); +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component); #endif diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index c2c5a6c..bcf562f0 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -48,6 +48,7 @@ struct mtk_base_irq_data { struct device; struct mtk_base_afe_memif; struct mtk_base_afe_irq; +struct mtk_base_afe_dai; struct regmap; struct snd_pcm_substream; struct snd_soc_dai; @@ -71,6 +72,11 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; + struct mtk_base_afe_dai *sub_dais; + int num_sub_dais; + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + const struct snd_pcm_hardware *mtk_afe_hardware; int (*memif_fs)(struct snd_pcm_substream *substream, unsigned int rate); @@ -94,5 +100,17 @@ struct mtk_base_afe_irq { int irq_occupyed; }; +struct mtk_base_afe_dai { + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + + const struct snd_kcontrol_new *controls; + unsigned int num_controls; + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + unsigned int num_dapm_routes; +}; + #endif -- 1.9.1

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[alsa-devel] [PATCH v2 1/3] ASoC: mediatek: add sub dai to mtk_base_afe
by KaiChieh Chuang 25 May '18

25 May '18

In MediaTek SoC chip we have multiple DAI, such as I2S, ADDA, PCM, etc. Organize each DAI in to one sub dai, with its dai driver, controls, widgets, routes. add mtk_afe_combine_sub_dai() to combine dai driver from each DAI. add mtk_afe_add_sub_dai_control() to register the control, widget, routes to component. Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 78 ++++++++++++++++++++++ .../soc/mediatek/common/mtk-afe-platform-driver.h | 5 ++ sound/soc/mediatek/common/mtk-base-afe.h | 18 +++++ 3 files changed, 101 insertions(+) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index 8966ee1..c449188 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -13,6 +13,84 @@ #include "mtk-afe-platform-driver.h" #include "mtk-base-afe.h" +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe) +{ + struct snd_soc_dai_driver *sub_dai_drivers; + size_t num_dai_drivers = 0, dai_idx = 0; + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + /* calcualte total dai driver size */ + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) + num_dai_drivers += afe->sub_dais[i].num_dai_drivers; + } + + dev_info(afe->dev, "%s(), num of dai %zd\n", __func__, num_dai_drivers); + + /* combine sub_dais */ + afe->num_dai_drivers = num_dai_drivers; + afe->dai_drivers = devm_kcalloc(afe->dev, + num_dai_drivers, + sizeof(struct snd_soc_dai_driver), + GFP_KERNEL); + if (!afe->dai_drivers) + return -ENOMEM; + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].dai_drivers && + afe->sub_dais[i].num_dai_drivers != 0) { + sub_dai_drivers = afe->sub_dais[i].dai_drivers; + /* dai driver */ + memcpy(&afe->dai_drivers[dai_idx], + sub_dai_drivers, + afe->sub_dais[i].num_dai_drivers * + sizeof(struct snd_soc_dai_driver)); + dai_idx += afe->sub_dais[i].num_dai_drivers; + } + } + + return 0; +} + +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component) +{ + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); + int i; + + if (!afe->sub_dais) { + dev_err(afe->dev, "%s(), sub_dais == NULL\n", __func__); + return -EINVAL; + } + + for (i = 0; i < afe->num_sub_dais; i++) { + if (afe->sub_dais[i].controls) + snd_soc_add_component_controls(component, + afe->sub_dais[i].controls, + afe->sub_dais[i].num_controls); + + if (afe->sub_dais[i].dapm_widgets) + snd_soc_dapm_new_controls(&component->dapm, + afe->sub_dais[i].dapm_widgets, + afe->sub_dais[i].num_dapm_widgets); + + if (afe->sub_dais[i].dapm_routes) + snd_soc_dapm_add_routes(&component->dapm, + afe->sub_dais[i].dapm_routes, + afe->sub_dais[i].num_dapm_routes); + } + + snd_soc_dapm_new_widgets(component->dapm.card); + + return 0; + +} + static snd_pcm_uframes_t mtk_afe_pcm_pointer (struct snd_pcm_substream *substream) { diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.h b/sound/soc/mediatek/common/mtk-afe-platform-driver.h index 1c81d91..0c31fa4 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.h +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.h @@ -12,5 +12,10 @@ #define AFE_PCM_NAME "mtk-afe-pcm" extern const struct snd_soc_component_driver mtk_afe_pcm_platform; +struct mtk_base_afe; +struct snd_soc_component; + +int mtk_afe_combine_sub_dai(struct mtk_base_afe *afe); +int mtk_afe_add_sub_dai_control(struct snd_soc_component *component); #endif diff --git a/sound/soc/mediatek/common/mtk-base-afe.h b/sound/soc/mediatek/common/mtk-base-afe.h index c2c5a6c..bcf562f0 100644 --- a/sound/soc/mediatek/common/mtk-base-afe.h +++ b/sound/soc/mediatek/common/mtk-base-afe.h @@ -48,6 +48,7 @@ struct mtk_base_irq_data { struct device; struct mtk_base_afe_memif; struct mtk_base_afe_irq; +struct mtk_base_afe_dai; struct regmap; struct snd_pcm_substream; struct snd_soc_dai; @@ -71,6 +72,11 @@ struct mtk_base_afe { struct mtk_base_afe_irq *irqs; int irqs_size; + struct mtk_base_afe_dai *sub_dais; + int num_sub_dais; + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + const struct snd_pcm_hardware *mtk_afe_hardware; int (*memif_fs)(struct snd_pcm_substream *substream, unsigned int rate); @@ -94,5 +100,17 @@ struct mtk_base_afe_irq { int irq_occupyed; }; +struct mtk_base_afe_dai { + struct snd_soc_dai_driver *dai_drivers; + unsigned int num_dai_drivers; + + const struct snd_kcontrol_new *controls; + unsigned int num_controls; + const struct snd_soc_dapm_widget *dapm_widgets; + unsigned int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + unsigned int num_dapm_routes; +}; + #endif -- 1.9.1

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[alsa-devel] [PATCH v3 0/8] slimbus: fixes and some helpers
by Srinivas Kandagatla 25 May '18

25 May '18

Hi Greg, Here is set of patches for slimbus, there are 6 fixes and two helper functions. Most of these issues were found while testing QCOM NGD SLIMBus controller with WCD9335 codec. Changes since v2 (https://lkml.org/lkml/2018/5/17/251) - removed unnecessary changes in slim_prepare_txn patch thanks, Srini Srinivas Kandagatla (8): slimbus: core: rearrange slim_eaddr structure slimbus: core: add need_tid flag to slim_msg_txn slimbus: messaging: pass correct wbuf slimbus: messaging: remove multiple calls to pm_runtime_mark_last_busy slimbus: messaging: initialize completion correctly slimbus: qcom: remove redundant depends in Kconfig slimbus: messaging: add slim_prepare_txn() helper function slimbus: core: add of_slim_device_get() helper drivers/slimbus/Kconfig | 1 - drivers/slimbus/core.c | 39 ++++++++++++++++++++ drivers/slimbus/messaging.c | 89 ++++++++++++++++++++++++++++++--------------- drivers/slimbus/slimbus.h | 4 ++ include/linux/slimbus.h | 14 ++++--- 5 files changed, 110 insertions(+), 37 deletions(-) -- 2.16.2

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[alsa-devel] [PATCH v5 0/3] ASoC: Add Multi CPU DAI support
by Shreyas NC 25 May '18

25 May '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v5: - Addressed review comments from Charles in calculating delays, soc_pcm_init_runtime_hw() and other nits - rebased on latest for-next branch Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 6 + sound/soc/soc-core.c | 280 ++++++++++++++++++++--------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 489 ++++++++++++++++++++++++++++++++++----------------- 4 files changed, 574 insertions(+), 272 deletions(-) -- 2.7.4

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