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

[alsa-devel] [PATCH 0/2] ALSA: fireworks: refactoring of substream counting
by Takashi Sakamoto 24 Oct '15

24 Oct '15

Hi, This patchset is for refactoring of substream counting. Currently the counting has atomic_t type. When utilizing lock by mutex in snd_efw structure, the counting can have simpler type, unsigned int. Takashi Sakamoto (2): fireworks: move mutex from function callees to callers fireworks: change type of substream counter from atomic_t to unsigned int sound/firewire/fireworks/fireworks.h | 4 ++-- sound/firewire/fireworks/fireworks_midi.c | 16 +++++++++++---- sound/firewire/fireworks/fireworks_pcm.c | 28 +++++++++++++++++-------- sound/firewire/fireworks/fireworks_stream.c | 32 +++++++++++------------------ 4 files changed, 46 insertions(+), 34 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 0/8] Add ASoC support for AMD APUs [v4]
by Alex Deucher 24 Oct '15

24 Oct '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL v4 changes: - patch "ASoC : dwc : support dw i2s in slave mode" accepted - Add a _dai_fmt() operation that checks to make sure that the mode we're setting corresponds to what we read back from the hardware. - Split specific quirks into separate patches - Set the specific quirks that apply to AMD chips in the acp driver Patch 7 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream5 Thanks, Alex Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (7): ASoC : dwc : add check for master/slave format ASoC : dwc : add different playback/capture base ASoC : dwc : add quirk for dwc controller instances ASoC : dwc : add quirk for different register offset ASoC : dwc : reconfigure dwc in 'resume' from 'suspend' ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add AMD ASoC ACP-I2S driver drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 127 ++ drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 275 +++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 41 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + include/linux/mfd/amd_acp.h | 43 + include/sound/designware_i2s.h | 6 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 518 ++++++ sound/soc/amd/acp.c | 736 +++++++++ sound/soc/amd/acp.h | 147 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 206 ++- 24 files changed, 6121 insertions(+), 64 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

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[alsa-devel] [RESEND PATCH 1/8] ASoC: dwc: add check for master/slave format
by Alex Deucher 23 Oct '15

23 Oct '15

From: Maruthi Srinivas Bayyavarapu <Maruthi.Bayyavarapu(a)amd.com> DW i2s controller's master/slave config can be read from a read-only register. Machine driver can try to set a master/slave format on cpu-dai using 'set_fmt' of dai ops. A check is added to verify codec is master when dwc is slave and vice-versa. Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu(a)amd.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/dwc/designware_i2s.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 3a52f82..f427325 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -341,12 +341,43 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, return ret; } +static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + if (dev->capability & DW_I2S_SLAVE) + ret = 0; + else + ret = -EINVAL; + break; + case SND_SOC_DAIFMT_CBS_CFS: + if (dev->capability & DW_I2S_MASTER) + ret = 0; + else + ret = -EINVAL; + break; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + dev_dbg(dev->dev, "dwc : Invalid master/slave format\n"); + ret = -EINVAL; + break; + } + return ret; +} + static struct snd_soc_dai_ops dw_i2s_dai_ops = { .startup = dw_i2s_startup, .shutdown = dw_i2s_shutdown, .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, + .set_fmt = dw_i2s_set_fmt, }; static const struct snd_soc_component_driver dw_i2s_component = { -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: rt5645: Add the HWEQ for the speaker output
by Oder Chiou 23 Oct '15

23 Oct '15

The patch adds the HWEQ function for the speaker output. User can set the HWEQ parameters using the ALSA binary control byte-by-byte. We use the following struct array to store the HWEQ parameters for implementing as simple as possible. struct rt5645_eq_param_s { unsigned short reg; unsigned short val; }; It supports the variant length of the HWEQ parameters that are required. We add the validating function in the function "rt5645_hweq_put" of the ALSA binary control to avoid the user that puts the invalid parameters. In the HWEQ enable function of speaker event, we also add the validating function to prevent that the invalid parameters are applied to codec. The HWEQ parameters should be controlled by DAPM for a specific sequence, so the parameters will be applied to the codec in the speaker power up event of DAPM, and will be disabled in the speaker power down event of DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 110 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 110 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d9797c..2813237 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -42,6 +42,8 @@ #define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) +#define RT5645_HWEQ_NUM 57 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -224,6 +226,11 @@ static const struct reg_default rt5645_reg[] = { { 0xff, 0x6308 }, }; +struct rt5645_eq_param_s { + unsigned short reg; + unsigned short val; +}; + static const char *const rt5645_supply_names[] = { "avdd", "cpvdd", @@ -240,6 +247,7 @@ struct rt5645_priv { struct snd_soc_jack *btn_jack; struct delayed_work jack_detect_work; struct regulator_bulk_data supplies[ARRAY_SIZE(rt5645_supply_names)]; + struct rt5645_eq_param_s *eq_param; int codec_type; int sysclk; @@ -477,6 +485,86 @@ static const DECLARE_TLV_DB_RANGE(spk_clsd_tlv, 7, 7, TLV_DB_SCALE_ITEM(228, 0, 0) ); +static int rt5645_hweq_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s); + + return 0; +} + +static int rt5645_hweq_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + eq_param[i].reg = cpu_to_be16(rt5645->eq_param[i].reg); + eq_param[i].val = cpu_to_be16(rt5645->eq_param[i].val); + } + + return 0; +} + +static bool rt5645_validate_hweq(unsigned short reg) +{ + if ((reg >= 0x1a4 && reg <= 0x1cd) | (reg >= 0x1e5 && reg <= 0x1f8) | + (reg == RT5645_EQ_CTRL2)) + return true; + + return false; +} + +static int rt5645_hweq_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + struct rt5645_eq_param_s *eq_param = + (struct rt5645_eq_param_s *)ucontrol->value.bytes.data; + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + eq_param[i].reg = be16_to_cpu(eq_param[i].reg); + eq_param[i].val = be16_to_cpu(eq_param[i].val); + } + + /* The final setting of the table should be RT5645_EQ_CTRL2 */ + for (i = RT5645_HWEQ_NUM - 1; i >= 0; i--) { + if (eq_param[i].reg == 0) + continue; + else if (eq_param[i].reg != RT5645_EQ_CTRL2) + return 0; + else + break; + } + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (!rt5645_validate_hweq(eq_param[i].reg) && + eq_param[i].reg != 0) + return 0; + else if (eq_param[i].reg == 0) + break; + } + + memcpy(rt5645->eq_param, eq_param, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s)); + + return 0; +} + +#define RT5645_HWEQ(xname) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = rt5645_hweq_info, \ + .get = rt5645_hweq_get, \ + .put = rt5645_hweq_put \ +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -541,6 +629,7 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* I2S2 function select */ SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, 1, 1), + RT5645_HWEQ("Speaker HWEQ"), }; /** @@ -631,6 +720,22 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } +static int rt5645_enable_hweq(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < RT5645_HWEQ_NUM; i++) { + if (rt5645_validate_hweq(rt5645->eq_param[i].reg)) + regmap_write(rt5645->regmap, rt5645->eq_param[i].reg, + rt5645->eq_param[i].val); + else + break; + } + + return 0; +} + /** * rt5645_sel_asrc_clk_src - select ASRC clock source for a set of filters * @codec: SoC audio codec device. @@ -1535,6 +1640,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + rt5645_enable_hweq(codec); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, @@ -1543,6 +1649,7 @@ static int rt5645_spk_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT5645_EQ_CTRL2, 0); snd_soc_update_bits(codec, RT5645_PWR_DIG1, RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | RT5645_PWR_CLS_D_L, 0); @@ -3060,6 +3167,9 @@ static int rt5645_probe(struct snd_soc_codec *codec) snd_soc_dapm_sync(dapm); } + rt5645->eq_param = devm_kzalloc(codec->dev, + RT5645_HWEQ_NUM * sizeof(struct rt5645_eq_param_s), GFP_KERNEL); + return 0; } -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 1/2] BAT: Add missing locale.h header
by Caleb Crome 23 Oct '15

23 Oct '15

Missing include became apparent when -O3 was not specified. Added the header to remove error messages during debug compile. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- bat/bat.c | 1 + 1 file changed, 1 insertion(+) diff --git a/bat/bat.c b/bat/bat.c index 56cfaf6..086b9fa 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -23,6 +23,7 @@ #include <getopt.h> #include <math.h> #include <limits.h> +#include <locale.h> #include "aconfig.h" #include "gettext.h" -- 2.6.1

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[alsa-devel] [PATCH v4 0/2] ASoC: nau8825: Add FLL config and device property debug
by Ben Zhang 23 Oct '15

23 Oct '15

This patch series adds the codec FLL configuration code to allow different input MCLK frequencies, and prints out the received device properties to make debugging easier with multiple versions of ACPI _DSD blob. Changes in v2: - Changed function/struct prefix from nau88xx_ to nau8825_ to match with the rest of the driver. Changes in v3: - Changed the implicit unsigned 64-bit divide to div_u64() to fix build errors on 32-bit arch. Changes in v4: - v2 and v3 updated only patch 1/2, not 2/2. The posted partial patch series makes code review confusing. v4 updates both patches in this series to the latest version. Ben Zhang (2): ASoC: nau8825: Add FLL configuration ASoC: nau8825: Show device properties sound/soc/codecs/nau8825.c | 193 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 28 +++++-- 2 files changed, 216 insertions(+), 5 deletions(-) -- 2.6.0.rc2.230.g3dd15c0

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[alsa-devel] I/O errors, snd_pcm_hw_params_get_ functions don't deliver values and system freezes on drop/drain/close
by Bommes, Michael 23 Oct '15

23 Oct '15

Hello, I am trying to develop an audio capture application for an SoC (ASC8852A with a TW2866 Audio Codec). I have SSH access to the SoC and the operating system was set up by someone else. Libraries and drivers were precompiled by the manufacturer, as well as a demo system for a video capturing server. Unfortunately I'm not that familiar with operating systems, driver programming and linux in special (although I know some basics regularly used in high-level development). I'm not sure whether I installed ALSA libraries and drivers correctly. Currently I have: - loaded in kernel (output printed to terminal): Advanced Linux Sound Architecture Driver Version 1.0.18rc3. ASoC version 0.13.2 ALSA device list: No soundcards found. - loaded by /etc/modules and modprobe.conf TW2866_AUDIO CodecNum=2 i2c-gpio bus_num=2 scl0=6 sda0=7 scl1=12 sda1=13 - performed after system startup: snddevices script delivered in the SDK of the SoC. sh snddevices The script is placed in the same folder as the TW2866_AUDIO.ko - alsa.conf was placed in /etc/ALSA/alsa.conf because that place is used in the reference OS and some install script, too. After that I have performed some cat and ls commands: /dev/snd/ is created and holds entries (in the demo system there is another timer entry which is missing in my OS): controlC0 pcmC0D0c pcmC0D0p pcmC0D1c pcmC0D2c pcmC0D3c pcmC0D4c cat /proc/asound/devices 0: [ 0] : control 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture 33: : timer and cat /proc/asound/cards 0 [VPL_AUDIO ]: VPL AUDIO - VPL Audio TW2866 Driver VPL Audio Codec Driver, TW2866. Now I have no idea whether the drivers and stuff were installed correctly. I tried to program a little sample program, but once I get pcm device to state SND_PCM_STATE_RUNNING the system freezes if I try to drain, drop or close the application, including all other ssh connections to the SoC so that a hard reset is necessary. Errors on readi occur before (10 seconds between each readi call and the error message), but none of the errors mentioned in the documentation (-EPIPE etc) but a -EIO error (Input/Output). But I'm not 100% sure whether the number of periods and buffer sizes were chosen correctly, so it might be possible that it's another user error (wrong buffer sizes delivered to driver could lead to freezes?). So I've reduced the code to setting hardware parameters and found out that snd_pcm_hw_params_get_ calls don't return errors but some of them always return 0 or don't change the output parameter. Here's sample source code and terminal output: ---------------------------------------------------------------------------------------------------------------- /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <alsa/asoundlib.h> void testError(const char* task, int rc) { if (rc < 0) { fprintf(stderr,"ERROR: %s: %s\n", task, snd_strerror(rc)); exit(1); } } int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0); testError("snd_pcm_open",rc); /* Allocate a hardware parameters object. */ // snd_pcm_hw_params_alloca(params); snd_pcm_hw_params_malloc(&params); /* Fill it in with default values. */ rc = snd_pcm_hw_params_any(handle, params); testError("snd_pcm_hw_params_any", rc); /* Set the desired hardware parameters. */ /* Interleaved mode */ rc = snd_pcm_hw_params_set_access(handle, params,SND_PCM_ACCESS_RW_INTERLEAVED); testError("snd_pcm_hw_params_set_access", rc); // test whether _set_access was applied snd_pcm_access_t access = SND_PCM_ACCESS_RW_NONINTERLEAVED; snd_pcm_hw_params_get_access (params, &access); if(access != SND_PCM_ACCESS_RW_INTERLEAVED) printf("ERROR! not SND_PCM_ACCESS_RW_INTERLEAVED\n"); /* Signed 16-bit little-endian format */ rc = snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_S16_LE); // non-16 bit formats fail to be set testError("snd_pcm_hw_params_set_format", rc); snd_pcm_format_t gottenFormat = SND_PCM_FORMAT_S8; snd_pcm_hw_params_get_format (params, &gottenFormat); if(gottenFormat != SND_PCM_FORMAT_S16_LE) { printf ("ERROR! not SND_PCM_FORMAT_S16_LE ... %i instead\n", gottenFormat); switch(gottenFormat) { case SND_PCM_FORMAT_S16_LE: printf("SND_PCM_FORMAT_S16_LE\n"); break; case SND_PCM_FORMAT_UNKNOWN: printf("SND_PCM_FORMAT_UNKNOWN\n"); break; default: break; //printf("%ld\n",gottenFormat); } } /* Two channels (stereo) */ rc = snd_pcm_hw_params_set_channels(handle, params, 2); testError("snd_pcm_hw_params_set_channels", rc); /* 44100 bits/second sampling rate (CD quality) */ unsigned int rate = 8000; //snd_pcm_hw_params_set_rate_near(handle, params,&val, &dir); rc = snd_pcm_hw_params_set_rate_near(handle, params,&rate, &dir); testError("snd_pcm_hw_params_set_rate_near", rc); printf("applied rate (return value): %i\n", (int)rate); unsigned int usedRate; rc = snd_pcm_hw_params_get_rate (params, &usedRate, &dir); testError("snd_pcm_hw_params_get_rate", rc); printf("used rate (getter value): %i\n", usedRate); unsigned int usedPeriods; snd_pcm_hw_params_get_periods(params, &usedPeriods, &dir); testError("", rc); printf("default periods: %i\n", usedPeriods); unsigned int periods = 8; // Set number of periods. Periods used to be called fragments. rc = snd_pcm_hw_params_set_periods(handle, params, periods, 0); testError("snd_pcm_hw_params_set_periods", rc); printf("Chosen number of periods (return value): %i\n", periods); rc = snd_pcm_hw_params_get_periods(params, &usedPeriods, &dir); testError("snd_pcm_hw_params_get_periods", rc); printf("used periods (getter value): %i\n", usedPeriods); /* Set period size to 32 frames. */ frames = 32; // snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir); snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir); testError("snd_pcm_hw_params_set_period_size_near", rc); printf("frames per period tried to set (return value): %ld\n", frames); printf("write to driver\n"); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); testError("snd_pcm_hw_params", rc); printf("written to driver\n"); snd_pcm_uframes_t maxSize; snd_pcm_hw_params_get_period_size_max(params, &maxSize, &dir); printf("max period size: %i\n", (int)maxSize); /* Use a buffer large enough to hold one period */ rc = snd_pcm_hw_params_get_period_size(params, &frames, &dir); testError("snd_pcm_hw_params_get_period_size", rc); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); /* We want to loop for 5 seconds */ rc = snd_pcm_hw_params_get_period_time(params, &val, &dir); testError("snd_pcm_hw_params_get_period_time", rc); loops = 5000000 / val; printf("frames per period: %ld\n", frames); // ERROR: here: size will be 0, so loops will be performed but nothing is copied... while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); printf("finished\n"); return 0; } ------------------------------------------------------------------------------------------------------------------ with output: ERROR! not SND_PCM_FORMAT_S16_LE ... 0 instead applied rate (return value): 8000 used rate (getter value): 8000 default periods: 0 Chosen number of periods (return value): 8 (comment: output parameter meant here) used periods (getter value): 0 frames per period tried to set (return value): 32 (comment: output parameter meant here) write to driver hello, alsa~. written to driver max period size: 0 frames per period: 0 finished This code doesn't freeze but getter functions "return" (where I mean the output parameter) 0 for periods and framesPerPeriod and format-getter doesn't return the expected, while sample-rate getter works. I'm a but confused about the "hello, alsa~." output, is that normal behaviour for the driver or might it be some kind of debug output in library or driver? System freeze looks like driver issue, or might this only be a wrong usage of the driver? Is there a way to check which parts of the ALSA installation (library, driver, codecs) do or don't work? How can I be certain that snddevices script and .conf files are sufficient? Thank you very much in advance, Michael

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[alsa-devel] [PATCH] BAT: fix missing header
by Caleb Crome 23 Oct '15

23 Oct '15

There was no warning when compiling with -O3, but gave a warning without it. commit a19fa1f7dff40e56dc4327c5e5fdde9e15bd8a55 Author: Caleb Crome <ccrome(a)gmail.com> Date: Tue Oct 20 20:59:50 2015 +0000 Add missing locale.h header. diff --git a/bat/bat.c b/bat/bat.c index 57c9db8..14666b2 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -23,6 +23,7 @@ #include <getopt.h> #include <math.h> #include <limits.h> +#include <locale.h> #include "aconfig.h" #include "gettext.h"

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[alsa-devel] [PATCH] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 23 Oct '15

23 Oct '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c737f1..6a947dc 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -504,6 +504,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); } break; default: -- 1.9.1

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[alsa-devel] [GIT PULL] ASoC updates for v4.3
by Mark Brown 23 Oct '15

23 Oct '15

The following changes since commit 7379047d5585187d1288486d4627873170d0005a: Linux 4.3-rc6 (2015-10-18 16:08:42 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.3-rc6 for you to fetch changes up to f69eccc4b248ce53b1b830034babd0a795d5763c: Merge remote-tracking branches 'asoc/fix/rt298', 'asoc/fix/sx', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus (2015-10-23 08:44:14 +0900) ---------------------------------------------------------------- ASoC: Fixes for v4.3 A bunch of driver fixes plus one core fix which fixes problems with misreporting values from _SX controls following a recent refactoring. This had gone unnoticed as such controls are quite rare. ---------------------------------------------------------------- Bard Liao (2): ASoC: rt298: correct index default value ASoC: rt298: fix wrong setting of gpio2_en Charles Keepax (2): ASoC: Add info callback for SX_TLV controls ASoC: wm8904: Correct number of EQ registers Jiada Wang (1): ASoC: wm8962: mark cache_dirty flag after software reset in pm_resume Mark Brown (3): Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-fix-rt5645 Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt298', 'asoc/fix/sx', 'asoc/fix/wm8904' and 'asoc/fix/wm8962' into asoc-linus include/sound/soc.h | 6 ++++-- include/sound/wm8904.h | 2 +- sound/soc/codecs/rt298.c | 26 +++++++++++++------------- sound/soc/codecs/wm8962.c | 2 ++ sound/soc/soc-ops.c | 28 ++++++++++++++++++++++++++++ 5 files changed, 48 insertions(+), 16 deletions(-)

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