April 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3] ucm: add cset-tlv
by Hsin-Yu Chao 13 Apr '16

13 Apr '16

This patch enables UCM to set a file in TLV format to kcontrol by: cset-tlv "name='<kcontrol-name>' <path-to-file>" This new 'cset-tlv' command will be used to write audio DSP to specific alsa control, where the driver expectes a file in TLV format. The TLV file to set to kcontrol will be checked first by file size not larger than 16 MB, and then examine if the length field reports correct number of bytes in the TLV file. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> --- src/ucm/main.c | 91 +++++++++++++++++++++++++++++++++++++++++++++-------- src/ucm/parser.c | 10 ++++++ src/ucm/ucm_local.h | 1 + 3 files changed, 89 insertions(+), 13 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 7e44603..d0929a2 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -161,6 +161,54 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int read_tlv_file(char **res, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + unsigned int *tlv; + + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + if (sz > 16 * 1024 * 1024 || sz < 8) { + uc_error("File size should be less than 16 MB"); + goto __fail; + } + *res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, *res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + free(*res); + *res = NULL; + } + /* Check if the tlv file specifies valid size. */ + tlv = (unsigned int *)(*res); + if (tlv[1] + 2 * sizeof(unsigned int) != sz) { + uc_error("Invalid tlv size"); + free(*res); + *res = NULL; + } + + __fail: + close(fd); + return err; +} + static int binary_file_parse(snd_ctl_elem_value_t *dst, snd_ctl_elem_info_t *info, const char *filepath) @@ -226,6 +274,7 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) snd_ctl_elem_id_t *id; snd_ctl_elem_value_t *value; snd_ctl_elem_info_t *info; + char *res = NULL; snd_ctl_elem_id_malloc(&id); snd_ctl_elem_value_malloc(&value); @@ -241,23 +290,36 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) err = -EINVAL; goto __fail; } - snd_ctl_elem_value_set_id(value, id); snd_ctl_elem_info_set_id(info, id); - err = snd_ctl_elem_read(ctl, value); - if (err < 0) - goto __fail; err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) - err = binary_file_parse(value, info, pos); - else - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); - if (err < 0) - goto __fail; - err = snd_ctl_elem_write(ctl, value); - if (err < 0) - goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_TLV) { + if (!snd_ctl_elem_info_is_tlv_writable(info)) { + err = -EINVAL; + goto __fail; + } + err = read_tlv_file(&res, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_tlv_write(ctl, id, (unsigned int *)res); + if (err < 0) + goto __fail; + } else { + snd_ctl_elem_value_set_id(value, id); + err = snd_ctl_elem_read(ctl, value); + if (err < 0) + goto __fail; + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (err < 0) + goto __fail; + err = snd_ctl_elem_write(ctl, value); + if (err < 0) + goto __fail; + } err = 0; __fail: if (id != NULL) @@ -266,6 +328,8 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) free(value); if (info != NULL) free(info); + if (res != NULL) + free(res); return err; } @@ -298,6 +362,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, break; case SEQUENCE_ELEMENT_TYPE_CSET: case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: + case SEQUENCE_ELEMENT_TYPE_CSET_TLV: if (cdev == NULL) { char *playback_ctl = NULL; char *capture_ctl = NULL; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 9e1cb41..d781e1b 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -316,6 +316,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-tlv") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_TLV; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-tlv requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 3a5d2c2..b89de2a 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -48,6 +48,7 @@ #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 #define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 +#define SEQUENCE_ELEMENT_TYPE_CSET_TLV 6 struct ucm_value { struct list_head list; -- 2.8.0.rc3.226.g39d4020

2 2

[alsa-devel] [PATCH] ALSA: hda - Remove temporary buffer for ELD update
by Hyungwon Hwang 13 Apr '16

13 Apr '16

With the introduce of pcm_lock and per_pin->lock, this temporary buffer is not needed anymore. Signed-off-by: Hyungwon Hwang <hyungwon.hwang7(a)gmail.com> --- sound/pci/hda/patch_hdmi.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5af372d..a24d5b4 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1395,8 +1395,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, struct hda_jack_tbl *jack; struct hda_codec *codec = per_pin->codec; struct hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld = &spec->temp_eld; - struct hdmi_eld *pin_eld = &per_pin->sink_eld; + struct hdmi_eld *eld = &per_pin->sink_eld; hda_nid_t pin_nid = per_pin->pin_nid; /* * Always execute a GetPinSense verb here, even when called from @@ -1413,15 +1412,15 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, present = snd_hda_pin_sense(codec, pin_nid); mutex_lock(&per_pin->lock); - pin_eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); - if (pin_eld->monitor_present) + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + if (eld->monitor_present) eld->eld_valid = !!(present & AC_PINSENSE_ELDV); else eld->eld_valid = false; codec_dbg(codec, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (spec->ops.pin_get_eld(codec, pin_nid, eld->eld_buffer, @@ -1441,7 +1440,7 @@ static bool hdmi_present_sense_via_verbs(struct hdmi_spec_per_pin *per_pin, else update_eld(codec, per_pin, eld); - ret = !repoll || !pin_eld->monitor_present || pin_eld->eld_valid; + ret = !repoll || !eld->monitor_present || eld->eld_valid; jack = snd_hda_jack_tbl_get(codec, pin_nid); if (jack) -- 2.5.0

2 2

[alsa-devel] [PATCH 1/1] ASoC: au1x: use correct format specifier
by Heinrich Schuchardt 13 Apr '16

13 Apr '16

Documentation/printk-formats.txt has unsigned long: use %lu or %lx size_t: use %zu or %zx runtime->dma_bytes is of type size_t. runtime->min_align is of type unsigned long. Signed-off-by: Heinrich Schuchardt <xypron.glpk(a)gmx.de> --- sound/soc/au1x/dbdma2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 5741c0a..b5d1caa 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, stype = substream->stream; pcd = to_dmadata(substream); - DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " - "runtime->min_align %d\n", + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu " + "runtime->min_align %lu\n", (unsigned long)runtime->dma_area, (unsigned long)runtime->dma_addr, runtime->dma_bytes, runtime->min_align); -- 2.1.4

1 0

[alsa-devel] [PATCH 1/1] ASoC: fsl: imx-pcm-fiq: use correct format specifier
by Heinrich Schuchardt 13 Apr '16

13 Apr '16

Documentation/printk-formats.txt has size_t: use %zu or %zx runtime->dma_bytes is of type size_t. Signed-off-by: Heinrich Schuchardt <xypron.glpk(a)gmx.de> --- sound/soc/fsl/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index e63cd5e..dac6688 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret, + pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret, runtime->dma_area, &runtime->dma_addr, runtime->dma_bytes); -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Update documentation
by Takashi Iwai 12 Apr '16

12 Apr '16

Update the URLs for alsa-info.sh and hda-emu. Also drop the alsa-driver snapshot URL since it's been discontinued recently. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Documentation/sound/alsa/HD-Audio.txt | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index e7193aac669c..d4510ebf2e8c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -655,17 +655,6 @@ development branches in general while the development for the current and next kernels are found in for-linus and for-next branches, respectively. -If you are using the latest Linus tree, it'd be better to pull the -above GIT tree onto it. If you are using the older kernels, an easy -way to try the latest ALSA code is to build from the snapshot -tarball. There are daily tarballs and the latest snapshot tarball. -All can be built just like normal alsa-driver release packages, that -is, installed via the usual spells: configure, make and make -install(-modules). See INSTALL in the package. The snapshot tarballs -are found at: - -- ftp://ftp.suse.com/pub/people/tiwai/snapshot/ - Sending a Bug Report ~~~~~~~~~~~~~~~~~~~~ @@ -699,7 +688,12 @@ problems. alsa-info ~~~~~~~~~ The script `alsa-info.sh` is a very useful tool to gather the audio -device information. You can fetch the latest version from: +device information. It's included in alsa-utils package. The latest +version can be found on git repository: + +- git://git.alsa-project.org/alsa-utils.git + +The script can be fetched directly from the following URL, too: - http://www.alsa-project.org/alsa-info.sh @@ -836,15 +830,11 @@ can get a proc-file dump at the current state, get a list of control (mixer) elements, set/get the control element value, simulate the PCM operation, the jack plugging simulation, etc. -The package is found in: - -- ftp://ftp.suse.com/pub/people/tiwai/misc/ - -A git repository is available: +The program is found in the git repository below: - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git -See README file in the tarball for more details about hda-emu +See README file in the repository for more details about hda-emu program. -- 2.8.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Add missing capture_hook calls for dyn-ADC PCM streams
by Takashi Iwai 12 Apr '16

12 Apr '16

The calls for capture_hook were missing in dyn_adc_capture_pcm_prepare and cleanup callbacks. Luckily there are no users of the capture hooks with dyn-adc PCM, so far, thus this doesn't change the behavior of existing devices, but it's a fix for a future usage. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 7ca5b89f088a..dc2c13687118 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5432,6 +5432,7 @@ static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_PREPARE); return 0; } @@ -5442,6 +5443,7 @@ static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_gen_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = 0; + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_CLEANUP); return 0; } -- 2.8.1

1 0

[alsa-devel] [PATCH v2 0/7] ASoC: sti: unip tdm mode
by Moise Gergaud 12 Apr '16

12 Apr '16

This patchset is V2 of "ASoC: sti: tdm: uni tdm mode" V2 consists of splitting the V1 patch into 7 patches. Moise Gergaud (7): ASoC: sti: macro for uniperif tdm regs access ASoC: sti: rename unip player type into common player & reader type ASoC: sti: define tdm type & default tdm hw config ASoC: sti: helper functions for unip tdm slots configuration ASoC: sti: helper functions to fix tdm runtime params ASoC: sti: unip player tdm mode ASoC: sti: unip reader tdm mode sound/soc/sti/sti_uniperif.c | 144 ++++++++++++++++++++++++- sound/soc/sti/uniperif.h | 197 +++++++++++++++++++++++++++++++++- sound/soc/sti/uniperif_player.c | 140 +++++++++++++++++------- sound/soc/sti/uniperif_reader.c | 229 +++++++++++++++++++++++++++++----------- 4 files changed, 606 insertions(+), 104 deletions(-) -- 1.9.1

2 14

[alsa-devel] [PATCH v2] ASoC: rockchip: i2s: configure the sdio pins' iomux mode
by Sugar Zhang 12 Apr '16

12 Apr '16

There are 3 i2s sdio pins, which iomux mode is as follows: - sdi3_sdo1 - sdi2_sdo2 - sdi1_sdo3 we need to configure these pins' iomux mode via the GRF register when use multi channel playback/capture. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Changes in v2: - fix a typo error. as suggested by Heiko: - use the general grf-binding without any offset - keep the actual offset in the driver on a per-soc basis. .../devicetree/bindings/sound/rockchip-i2s.txt | 5 ++ sound/soc/rockchip/rockchip_i2s.c | 67 +++++++++++++++++++--- sound/soc/rockchip/rockchip_i2s.h | 7 +++ 3 files changed, 70 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index 6e86d8a..4ea29aa 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -23,6 +23,11 @@ Required properties: - rockchip,playback-channels: max playback channels, if not set, 8 channels default. - rockchip,capture-channels: max capture channels, if not set, 2 channels default. +Required properties for controller which support multi channels +playback/capture: + +- rockchip,grf: the phandle of the syscon node for GRF register. + Example for rk3288 I2S controller: i2s@ff890000 { diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e204..d4f7eac 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -11,8 +11,10 @@ */ #include <linux/module.h> +#include <linux/mfd/syscon.h> #include <linux/delay.h> #include <linux/of_gpio.h> +#include <linux/of_device.h> #include <linux/clk.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> @@ -23,6 +25,11 @@ #define DRV_NAME "rockchip-i2s" +struct rk_i2s_pins { + u32 reg_offset; + u32 shift; +}; + struct rk_i2s_dev { struct device *dev; @@ -33,8 +40,10 @@ struct rk_i2s_dev { struct snd_dmaengine_dai_dma_data playback_dma_data; struct regmap *regmap; + struct regmap *grf; bool is_master_mode; + const struct rk_i2s_pins *pins; }; static int i2s_runtime_suspend(struct device *dev) @@ -277,6 +286,30 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, I2S_TXCR_VDW_MASK | I2S_TXCR_CSR_MASK, val); + if (!IS_ERR(i2s->grf) && i2s->pins) { + regmap_read(i2s->regmap, I2S_TXCR, &val); + val &= I2S_TXCR_CSR_MASK; + + switch (val) { + case I2S_CHN_4: + val = I2S_IO_4CH_OUT_6CH_IN; + break; + case I2S_CHN_6: + val = I2S_IO_6CH_OUT_4CH_IN; + break; + case I2S_CHN_8: + val = I2S_IO_8CH_OUT_2CH_IN; + break; + default: + val = I2S_IO_2CH_OUT_8CH_IN; + break; + } + + val <<= i2s->pins->shift; + val |= (I2S_IO_DIRECTION_MASK << i2s->pins->shift) << 16; + regmap_write(i2s->grf, i2s->pins->reg_offset, val); + } + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDL_MASK, I2S_DMACR_TDL(16)); regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDL_MASK, @@ -462,9 +495,23 @@ static const struct regmap_config rockchip_i2s_regmap_config = { .cache_type = REGCACHE_FLAT, }; +static const struct rk_i2s_pins rk3399_i2s_pins = { + .reg_offset = 0xe220, + .shift = 11, +}; + +static const struct of_device_id rockchip_i2s_match[] = { + { .compatible = "rockchip,rk3066-i2s", }, + { .compatible = "rockchip,rk3188-i2s", }, + { .compatible = "rockchip,rk3288-i2s", }, + { .compatible = "rockchip,rk3399-i2s", .data = &rk3399_i2s_pins }, + {}, +}; + static int rockchip_i2s_probe(struct platform_device *pdev) { struct device_node *node = pdev->dev.of_node; + const struct of_device_id *of_id; struct rk_i2s_dev *i2s; struct snd_soc_dai_driver *soc_dai; struct resource *res; @@ -478,6 +525,17 @@ static int rockchip_i2s_probe(struct platform_device *pdev) return -ENOMEM; } + i2s->dev = &pdev->dev; + + i2s->grf = syscon_regmap_lookup_by_phandle(node, "rockchip,grf"); + if (!IS_ERR(i2s->grf)) { + of_id = of_match_device(rockchip_i2s_match, &pdev->dev); + if (!of_id || !of_id->data) + return -EINVAL; + + i2s->pins = of_id->data; + } + /* try to prepare related clocks */ i2s->hclk = devm_clk_get(&pdev->dev, "i2s_hclk"); if (IS_ERR(i2s->hclk)) { @@ -517,7 +575,6 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; i2s->capture_dma_data.maxburst = 4; - i2s->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, i2s); pm_runtime_enable(&pdev->dev); @@ -583,14 +640,6 @@ static int rockchip_i2s_remove(struct platform_device *pdev) return 0; } -static const struct of_device_id rockchip_i2s_match[] = { - { .compatible = "rockchip,rk3066-i2s", }, - { .compatible = "rockchip,rk3188-i2s", }, - { .compatible = "rockchip,rk3288-i2s", }, - { .compatible = "rockchip,rk3399-i2s", }, - {}, -}; - static const struct dev_pm_ops rockchip_i2s_pm_ops = { SET_RUNTIME_PM_OPS(i2s_runtime_suspend, i2s_runtime_resume, NULL) diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index dc6e2c7..8e239d3 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -236,4 +236,11 @@ enum { #define I2S_TXDR (0x0024) #define I2S_RXDR (0x0028) +/* io direction cfg register */ +#define I2S_IO_DIRECTION_MASK (7) +#define I2S_IO_8CH_OUT_2CH_IN (0) +#define I2S_IO_6CH_OUT_4CH_IN (4) +#define I2S_IO_4CH_OUT_6CH_IN (6) +#define I2S_IO_2CH_OUT_8CH_IN (7) + #endif /* _ROCKCHIP_IIS_H */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Add DSP firmware manifest parsing
by Vinod Koul 12 Apr '16

12 Apr '16

From: "Nc,Shreyas" <shreyas.nc(a)intel.com> Module params like module_id and loadable flag can changed in the DSP Firmware. These are kept in the firmware manifest and driver should read these values from this manifest. So, add support to parse the DSP firmware manifest and read these module params. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 134 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 109 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 4 + sound/soc/intel/skylake/skl-sst.c | 11 ++- sound/soc/intel/skylake/skl-topology.c | 15 ++-- 6 files changed, 268 insertions(+), 7 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index c28f5d0e1d99..e2fb5b763b21 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o + skl-sst.o skl-dsp-parse.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-dsp-parse.c b/sound/soc/intel/skylake/skl-dsp-parse.c new file mode 100644 index 000000000000..1736dec68e3c --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.c @@ -0,0 +1,134 @@ +/* + * skl-dsp-parse.c - Implements DSP firmware manifest parsing + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <asm/types.h> +#include <linux/types.h> +#include <linux/string.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/slab.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-tplg-interface.h" +#include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" + +/* + * Get the module id for the module by checking + * the table for the UUID for the module + */ +int snd_skl_get_module_info(struct skl_sst *ctx, char *uuid, + struct skl_dfw_module *dfw_config) +{ + int i, num; + struct uuid_tbl *tbl; + + tbl = ctx->tbl; + num = ctx->num_modules; + + for (i = 0; i < num; i++) { + if (!strcmp(uuid, tbl[i].uuid)) { + dfw_config->module_id = tbl[i].module_id; + dfw_config->is_loadable = tbl[i].is_loadable; + + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_skl_get_module_info); + +/* + * Get the uuid from the DSP firmware structure in a string + * of the format XXXXXXXX-XXXX-XXXX-XXXXXXXXXXXX + */ +static char *get_canonical_uuid(char *buf, + struct adsp_module_entry *entry) +{ + unsigned int i; + + sprintf(buf, "%08X", entry->uuid.id_1); + sprintf(buf+strlen(buf), "-"); + + for (i = 0; i < ARRAY_SIZE(entry->uuid.id_2); i++) { + sprintf(buf+strlen(buf), + "%04X", entry->uuid.id_2[i]); + sprintf(buf+strlen(buf), "-"); + } + + for (i = 0; i < ARRAY_SIZE(entry->uuid.id_3); i++) { + sprintf(buf+strlen(buf), + "%02X", entry->uuid.id_3[i]); + if (i == 1) + sprintf(buf+strlen(buf), "-"); + } + + return buf; +} + +/* + * Parse the firmware binary to get the UUID, module id + * and loadable flags + */ +int parse_fw_bin(struct sst_dsp *ctx) +{ + struct adsp_fw_hdr *adsp_hdr; + struct adsp_module_entry *mod_entry; + int i, num_entry; + char uuid_str[UUID_STR_SIZE]; + const char *buf; + struct skl_sst *skl = ctx->thread_context; + struct uuid_tbl *tbl; + + + /* Get the FW pointer to derive ADSP header */ + buf = ctx->fw->data; + + adsp_hdr = (struct adsp_fw_hdr *)(buf + SKL_ADSP_FW_BIN_HDR_OFFSET); + + mod_entry = (struct adsp_module_entry *) + (buf + SKL_ADSP_FW_BIN_HDR_OFFSET + adsp_hdr->header_len); + + num_entry = adsp_hdr->num_module_entries; + + tbl = devm_kzalloc(ctx->dev, + num_entry * sizeof(struct uuid_tbl), GFP_KERNEL); + + if (!tbl) + return -ENOMEM; + + /* + * Populate the UUID table to store module_id + * and loadable flags for the module + */ + + for (i = 0; i < num_entry; i++, mod_entry++) { + get_canonical_uuid(uuid_str, mod_entry); + + strncpy(tbl[i].uuid, uuid_str, strlen(uuid_str)); + + tbl[i].module_id = i; + tbl[i].is_loadable = mod_entry->type.load_type; + } + + skl->tbl = tbl; + skl->num_modules = num_entry; + return 0; +} +EXPORT_SYMBOL_GPL(parse_fw_bin); diff --git a/sound/soc/intel/skylake/skl-dsp-parse.h b/sound/soc/intel/skylake/skl-dsp-parse.h new file mode 100644 index 000000000000..2da1bbad9145 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.h @@ -0,0 +1,109 @@ +/* + * skl-dsp-parse.h + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <asm/types.h> +#include <linux/types.h> +#include <linux/uuid.h> +#include "skl-tplg-interface.h" +#include "../common/sst-dsp-priv.h" + +#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 +#define UUID_STR_SIZE 37 +#define DEFAULT_HASH_SHA256_LEN 32 + +struct skl_dfw_module_mod { + char name[100]; + struct skl_dfw_module skl_dfw_mod; +}; + +struct UUID { + u32 id_1; + u16 id_2[2]; + u8 id_3[8]; +}; + +union seg_flags { + u32 ul; + struct { + u32 contents : 1; + u32 alloc : 1; + u32 load : 1; + u32 read_only : 1; + u32 code : 1; + u32 data : 1; + u32 _rsvd0 : 2; + u32 type : 4; + u32 _rsvd1 : 4; + u32 length : 16; + } r; +} __packed; + +struct segment_desc { + union seg_flags flags; + u32 v_base_addr; + u32 file_offset; +}; + +struct module_type { + u32 load_type : 4; + u32 auto_start : 1; + u32 domain_ll : 1; + u32 domain_dp : 1; + u32 rsvd_ : 25; +} __packed; + +struct adsp_module_entry { + u32 struct_id; + u8 name[8]; + struct UUID uuid; + struct module_type type; + u8 hash1[DEFAULT_HASH_SHA256_LEN]; + u32 entry_point; + u16 cfg_offset; + u16 cfg_count; + u32 affinity_mask; + u16 instance_max_count; + u16 instance_bss_size; + struct segment_desc segments[3]; +} __packed; + +struct adsp_fw_hdr { + u32 header_id; + u32 header_len; + u8 name[8]; + u32 preload_page_count; + u32 fw_image_flags; + u32 feature_mask; + u16 major_version; + u16 minor_version; + u16 hotfix_version; + u16 build_version; + u32 num_module_entries; + u32 hw_buf_base_addr; + u32 hw_buf_length; + u32 load_offset; +} __packed; + +struct uuid_tbl { + char uuid[UUID_STR_SIZE]; + int module_id; + int is_loadable; +}; + +int parse_fw_bin(struct sst_dsp *ctx); +int snd_skl_get_module_info(struct skl_sst *ctx, char *uuid, + struct skl_dfw_module *dfw_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index d59d1ba62a43..67d56be8cb77 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -60,6 +60,10 @@ struct skl_sst { void (*enable_miscbdcge)(struct device *dev, bool enable); /*Is CGCTL.MISCBDCGE disabled*/ bool miscbdcg_disabled; + + /* Populate module information */ + struct uuid_tbl *tbl; + int num_modules; }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 348a734f8e24..9fbe922d05ce 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -24,6 +24,7 @@ #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" #include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" #define SKL_BASEFW_TIMEOUT 300 #define SKL_INIT_TIMEOUT 1000 @@ -69,7 +70,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { - int ret = 0, i; + int ret = 0, i, err; struct skl_sst *skl = ctx->thread_context; u32 reg; @@ -83,6 +84,14 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) skl_dsp_disable_core(ctx); return -EIO; } + + err = parse_fw_bin(ctx); + if (err < 0) { + dev_err(ctx->dev, + "Firmware parsing err: %d\n", ret); + release_firmware(ctx->fw); + return err; + } } ret = skl_dsp_boot(ctx); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 545b4e77b8aa..d64edfa45b6e 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -28,6 +28,7 @@ #include "skl-tplg-interface.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "skl-dsp-parse.h" #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) @@ -1550,7 +1551,15 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, return -ENOMEM; w->priv = mconfig; + memcpy(mconfig->guid, dfw_config->uuid, + ARRAY_SIZE(dfw_config->uuid)); + + ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config); + if (ret < 0) + return ret; + mconfig->id.module_id = dfw_config->module_id; + mconfig->is_loadable = dfw_config->is_loadable; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; mconfig->ibs = dfw_config->ibs; @@ -1558,7 +1567,7 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, mconfig->core_id = dfw_config->core_id; mconfig->max_in_queue = dfw_config->max_in_queue; mconfig->max_out_queue = dfw_config->max_out_queue; - mconfig->is_loadable = dfw_config->is_loadable; + skl_tplg_fill_fmt(mconfig->in_fmt, dfw_config->in_fmt, MODULE_MAX_IN_PINS); skl_tplg_fill_fmt(mconfig->out_fmt, dfw_config->out_fmt, @@ -1579,10 +1588,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, mconfig->time_slot = dfw_config->time_slot; mconfig->formats_config.caps_size = dfw_config->caps.caps_size; - if (dfw_config->is_loadable) - memcpy(mconfig->guid, dfw_config->uuid, - ARRAY_SIZE(dfw_config->uuid)); - mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) * sizeof(*mconfig->m_in_pin), GFP_KERNEL); -- 2.4.3

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[alsa-devel] Too fast playback everywhere, except audacity
by Giedrius Trainavičius 12 Apr '16

12 Apr '16

Hello, I am developing a driver for a DAC on raspberry pi 3 by mostly making use of existing PCM5102A, and bcm2708 I2S driver code. What I've done is just configuring the cpu dai link, and for the sampling rate changes, I am setting some gpios which configures the clock of an ADC (PCM1804) which is a master. The main CPU and the DAC are slaves to the ADC and I have a 24.576MHz oscillator. The sampling rates I want to support are 48000, 96000 and 192000. I am able to react just fine to the requested rate in hw_params callback. Trying out the different rates in audacity works just fine. However, if I try to playback any wav files using mplayer or aplay or youtube, all of their playback requests for 48000, but they get played way too fast. Also if I request mplayer to use alsa output, it gets a buffer underrun and gets stuck. If I set audacity to use some other output device than alsa directly, it plays corrupted audio and hangs on stopping playback. How does audacity manage to play the audio at the correct speed? Is it doing software resampling to the 'project rate'? I've tried getting resampling to work using 'plug', but completely unsuccessfully - I can't get any sound out of it, it says it failed setting the requested parameters. Any pointers to what I should be investigating? I am new to the linux kernel module development, as well as sound drivers. Thank you in advance, Giedrius.

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