June 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: add missing MIX binding example
by Kuninori Morimoto 05 Jun '17

05 Jun '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/audio-graph-scu-card.txt | 47 +++++++++++++++++++++- 1 file changed, 46 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt index b2dd23f..ca494cf 100644 --- a/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt +++ b/Documentation/devicetree/bindings/sound/audio-graph-scu-card.txt @@ -35,7 +35,7 @@ Required properties: - compatible : "audio-graph-scu-card"; - dais : list of CPU DAI port{s} -Example +Example 1. Sampling Rate Covert sound_card { compatible = "audio-graph-scu-card"; @@ -70,3 +70,48 @@ Example }; }; }; + +Example 2. 2 CPU 1 Codec (Mixing) + + sound_card { + compatible = "audio-graph-scu-card"; + + label = "sound-card"; + prefix = "codec"; + routing = "codec Playback", "DAI0 Playback", + "codec Playback", "DAI1 Playback"; + convert-rate = <48000>; + + dais = <&cpu_port0 + &cpu_port1>; + }; + + audio-codec { + ... + + port { + codec_endpoint: endpoint { + remote-endpoint = <&cpu_endpoint0>; + }; + }; + }; + + dai-controller { + ... + ports { + cpu_port0: port { + cpu_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + ... + }; + }; + cpu_port1: port { + cpu_endpoint1: endpoint { + dai-format = "left_j"; + ... + }; + }; + }; + }; -- 1.9.1

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[alsa-devel] ALSA SoC + Hotplug ?
by Kuninori Morimoto 05 Jun '17

05 Jun '17

Hi ALSA ML Now, ALSA SoC can support HDMI, then, we want to support Hotplug for it. In my quick grep, it seems ALSA SoC doesn't have such support now (?) But, is it supported ? Thanks Kuninori Morimoto

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[alsa-devel] [PATCH v2 00/27] Revised full patchset for PCM in-kernel copy support
by Takashi Iwai 02 Jun '17

02 Jun '17

Hi, this is a full patchset of what I sent previously, containing the all changes instead of the snippet. The main purpose of this patchset is to eliminate the remaining usages of set_fs(). They are basically used for in-kernel PCM data transfer, and this patch provides the new API functions and replaces the hackish set_fs() calls with them. Unlike the first patchset with the unified copy_silence ops, this adds a new copy_kernel ops instead. At the same time, copy/silence are changed to receive the position and size in bytes instead of frames. This allows us to simplify the PCM core code. As a result, a good amount of code could be removed from pcm_lib.c. The difference from the previous patchset is that this is a full patchset, i.e. all relevant drivers have been covered, and also some small issues have been addressed, in addition, the documentation update is provided, too. I'm Cc'ing the media and the USB people since it touches solo6x10 and usb-gadget drivers. The previous ACK was dropped as each patch was rewritten again. Sorry for the doubly patch-review labours. thanks, Takashi === Takashi Iwai (26): ALSA: pcm: Introduce copy_user, copy_kernel and fill_silence ops ALSA: dummy: Convert to new PCM copy ops ALSA: es1938: Convert to the new PCM copy ops ALSA: nm256: Convert to new PCM copy ops ALSA: korg1212: Convert to the new PCM ops ALSA: rme32: Convert to the new PCM copy ops ALSA: rme96: Convert to the new PCM ops ALSA: rme9652: Convert to the new PCM ops ALSA: hdsp: Convert to the new PCM ops ALSA: gus: Convert to the new PCM ops ALSA: sb: Convert to the new PCM ops ALSA: sh: Convert to the new PCM ops ASoC: blackfin: Convert to the new PCM ops [media] solo6x10: Convert to the new PCM ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Check PCM state by a common helper function ALSA: pcm: Shuffle codes ALSA: pcm: Call directly the common read/write helpers ALSA: pcm: More unification of PCM transfer codes ALSA: pcm: Unify read/write loop ALSA: pcm: Simplify snd_pcm_playback_silence() ALSA: pcm: Direct in-kernel read/write support usb: gadget: u_uac1: Kill set_fs() usage ALSA: pcm: Kill set_fs() in PCM OSS layer ALSA: pcm: Build OSS writev/readv helpers conditionally ALSA: doc: Update copy_user, copy_kernel and fill_silence PCM ops .../sound/kernel-api/writing-an-alsa-driver.rst | 111 ++-- drivers/media/pci/solo6x10/solo6x10-g723.c | 32 +- drivers/usb/gadget/function/u_uac1.c | 7 +- include/sound/pcm.h | 80 ++- sound/core/oss/io.c | 4 +- sound/core/oss/pcm_oss.c | 81 +-- sound/core/oss/pcm_plugin.h | 6 +- sound/core/pcm_lib.c | 564 ++++++++------------- sound/drivers/dummy.c | 20 +- sound/isa/gus/gus_pcm.c | 97 ++-- sound/isa/sb/emu8000_pcm.c | 190 ++++--- sound/pci/es1938.c | 33 +- sound/pci/korg1212/korg1212.c | 112 ++-- sound/pci/nm256/nm256.c | 57 ++- sound/pci/rme32.c | 65 ++- sound/pci/rme96.c | 70 ++- sound/pci/rme9652/hdsp.c | 67 ++- sound/pci/rme9652/rme9652.c | 71 ++- sound/sh/sh_dac_audio.c | 54 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 27 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 36 +- sound/soc/soc-pcm.c | 5 +- 22 files changed, 977 insertions(+), 812 deletions(-) -- 2.13.0

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[alsa-devel] [PATCH 1/4] ASoC: cs35l35: Add Boost Inductor Calculation
by Charles Keepax 02 Jun '17

02 Jun '17

From: Brian Austin <brian.austin(a)cirrus.com> Add the Boost Inductor parameters based off the size of the inductor on the HW setup Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/sound/cs35l35.h | 2 ++ sound/soc/codecs/cs35l35.c | 82 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l35.h | 6 ++++ 3 files changed, 90 insertions(+) diff --git a/include/sound/cs35l35.h b/include/sound/cs35l35.h index 29da899..d69cd78 100644 --- a/include/sound/cs35l35.h +++ b/include/sound/cs35l35.h @@ -99,6 +99,8 @@ struct cs35l35_platform_data { bool shared_bst; /* Specifies this amp is using an external boost supply */ bool ext_bst; + /* Inductor Value */ + int boost_ind; /* ClassH Algorithm */ struct classh_cfg classh_algo; /* Monitor Config */ diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index f8aef58..c6eabb8 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -756,6 +756,76 @@ static int cs35l35_codec_set_sysclk(struct snd_soc_codec *codec, return ret; } +static int cs35l35_boost_inductor(struct cs35l35_private *cs35l35, + int inductor) +{ + struct regmap *regmap = cs35l35->regmap; + unsigned int bst_ipk = 0; + + /* + * Digital Boost Converter Configuration for feedback, + * ramping, switching frequency, and estimation block seeding. + */ + + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_SWFREQ_MASK, 0x00); + + regmap_read(regmap, CS35L35_BST_PEAK_I, &bst_ipk); + bst_ipk &= CS35L35_BST_IPK_MASK; + + switch (inductor) { + case 1000: /* 1 uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x24); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x24); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x00); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x4E); + break; + case 1200: /* 1.2 uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x20); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x20); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x01); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x47); + break; + case 1500: /* 1.5uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x20); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x20); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x02); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x3C); + break; + case 2200: /* 2.2uH */ + regmap_write(regmap, CS35L35_BST_CONV_COEF_1, 0x19); + regmap_write(regmap, CS35L35_BST_CONV_COEF_2, 0x25); + regmap_update_bits(regmap, CS35L35_BST_CONV_SW_FREQ, + CS35L35_BST_CONV_LBST_MASK, 0x03); + + if (bst_ipk < 0x04) + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x1B); + else + regmap_write(regmap, CS35L35_BST_CONV_SLOPE_COMP, 0x23); + break; + default: + dev_err(cs35l35->dev, "Invalid Inductor Value %d uH\n", + inductor); + return -EINVAL; + } + return 0; +} + static int cs35l35_codec_probe(struct snd_soc_codec *codec) { struct cs35l35_private *cs35l35 = snd_soc_codec_get_drvdata(codec); @@ -775,6 +845,10 @@ static int cs35l35_codec_probe(struct snd_soc_codec *codec) cs35l35->pdata.bst_ipk << CS35L35_BST_IPK_SHIFT); + ret = cs35l35_boost_inductor(cs35l35, cs35l35->pdata.boost_ind); + if (ret) + return ret; + if (cs35l35->pdata.gain_zc) regmap_update_bits(cs35l35->regmap, CS35L35_PROTECT_CTL, CS35L35_AMP_GAIN_ZC_MASK, @@ -1198,6 +1272,14 @@ static int cs35l35_handle_of_data(struct i2c_client *i2c_client, pdata->bst_ipk = (val32 - 1680) / 110; } + ret = of_property_read_u32(np, "cirrus,boost-ind-nanohenry", &val32); + if (ret >= 0) { + pdata->boost_ind = val32; + } else { + dev_err(&i2c_client->dev, "Inductor not specified.\n"); + return -EINVAL; + } + if (of_property_read_u32(np, "cirrus,sp-drv-strength", &val32) >= 0) pdata->sp_drv_str = val32; if (of_property_read_u32(np, "cirrus,sp-drv-unused", &val32) >= 0) diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index 5a6e43a..621bfef 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -200,6 +200,12 @@ #define CS35L35_SP_I2S_DRV_MASK 0x03 #define CS35L35_SP_I2S_DRV_SHIFT 0 +/* Boost Converter Config */ +#define CS35L35_BST_CONV_COEFF_MASK 0xFF +#define CS35L35_BST_CONV_SLOPE_MASK 0xFF +#define CS35L35_BST_CONV_LBST_MASK 0x03 +#define CS35L35_BST_CONV_SWFREQ_MASK 0xF0 + /* Class H Algorithm Control */ #define CS35L35_CH_STEREO_MASK 0x40 #define CS35L35_CH_STEREO_SHIFT 6 -- 2.1.4

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[alsa-devel] [PATCH v2] topology: Add parsing for rates from conf
by guneshwor.o.singh＠intel.com 02 Jun '17

02 Jun '17

From: Guneshwor Singh <guneshwor.o.singh(a)intel.com> rate_min and rate_max are parsed currently, some drivers may also use rates to specify PCM stream capability. So add support to parse it. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- include/topology.h | 1 + src/topology/pcm.c | 71 +++++++++++++++++++++++++++++++++++++++++++++++ src/topology/tplg_local.h | 20 +++++++++++++ 3 files changed, 92 insertions(+) diff --git a/include/topology.h b/include/topology.h index ccb3a004..42d23762 100644 --- a/include/topology.h +++ b/include/topology.h @@ -534,6 +534,7 @@ extern "C" { * SectionPCMCapabilities."name" { * * formats "S24_LE,S16_LE" # Supported formats + * rates "48000" # Supported rates * rate_min "48000" # Max supported sample rate * rate_max "48000" # Min supported sample rate * channels_min "2" # Min number of channels diff --git a/src/topology/pcm.c b/src/topology/pcm.c index daef20e4..9a27a13a 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -20,6 +20,26 @@ #include "list.h" #include "tplg_local.h" +#define RATE(v) [SND_PCM_RATE_##v] = #v + +static const char *const snd_pcm_rate_names[] = { + RATE(5512), + RATE(8000), + RATE(11025), + RATE(16000), + RATE(22050), + RATE(32000), + RATE(44100), + RATE(48000), + RATE(64000), + RATE(88200), + RATE(96000), + RATE(176400), + RATE(192000), + RATE(CONTINUOUS), + RATE(KNOT), +}; + struct tplg_elem *lookup_pcm_dai_stream(struct list_head *base, const char* id) { struct list_head *pos; @@ -309,6 +329,42 @@ static int split_format(struct snd_soc_tplg_stream_caps *caps, char *str) return 0; } +static int get_rate_value(const char* name) +{ + int rate; + for (rate = 0; rate <= SND_PCM_RATE_LAST; rate++) { + if (snd_pcm_rate_names[rate] && + strcasecmp(name, snd_pcm_rate_names[rate]) == 0) { + return rate; + } + } + + return SND_PCM_RATE_UNKNOWN; +} + +static int split_rate(struct snd_soc_tplg_stream_caps *caps, char *str) +{ + char *s = NULL; + snd_pcm_rates_t rate; + int i = 0; + + s = strtok(str, ","); + while (s) { + rate = get_rate_value(s); + + if (rate == SND_PCM_RATE_UNKNOWN) { + SNDERR("error: unsupported stream rate %s\n", s); + return -EINVAL; + } + + caps->rates |= 1 << rate; + s = strtok(NULL, ", "); + i++; + } + + return 0; +} + /* Parse pcm stream capabilities */ int tplg_parse_stream_caps(snd_tplg_t *tplg, snd_config_t *cfg, void *private ATTRIBUTE_UNUSED) @@ -360,6 +416,21 @@ int tplg_parse_stream_caps(snd_tplg_t *tplg, continue; } + if (strcmp(id, "rates") == 0) { + s = strdup(val); + if (!s) + return -ENOMEM; + + err = split_rate(sc, s); + free(s); + + if (err < 0) + return err; + + tplg_dbg("\t\t%s: %s\n", id, val); + continue; + } + if (strcmp(id, "rate_min") == 0) { sc->rate_min = atoi(val); tplg_dbg("\t\t%s: %d\n", id, sc->rate_min); diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h index 42e3201d..127842f4 100644 --- a/src/topology/tplg_local.h +++ b/src/topology/tplg_local.h @@ -38,6 +38,26 @@ struct tplg_ref; struct tplg_elem; +typedef enum _snd_pcm_rates { + SND_PCM_RATE_UNKNOWN = -1, + SND_PCM_RATE_5512 = 0, + SND_PCM_RATE_8000, + SND_PCM_RATE_11025, + SND_PCM_RATE_16000, + SND_PCM_RATE_22050, + SND_PCM_RATE_32000, + SND_PCM_RATE_44100, + SND_PCM_RATE_48000, + SND_PCM_RATE_64000, + SND_PCM_RATE_88200, + SND_PCM_RATE_96000, + SND_PCM_RATE_176400, + SND_PCM_RATE_192000, + SND_PCM_RATE_CONTINUOUS = 30, + SND_PCM_RATE_KNOT = 31, + SND_PCM_RATE_LAST = SND_PCM_RATE_KNOT, +} snd_pcm_rates_t; + struct snd_tplg { /* opaque vendor data */ -- 2.13.0

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[alsa-devel] cht_bsw_rt5645 mostly silent
by James Cameron 01 Jun '17

01 Jun '17

G'day, I've a laptop on my lab bench with a Realtek ALC5645 codec attached to an Intel(R) Atom(TM) x5-Z8350. Built-in speakers work properly. Not working are; - headphones; only ever silent playback, - built-in microphone; only ever silent recording, - headset microphone; only ever silent recording, - there is no response to headset jack insertion or removal, It has never worked. http://www.alsa-project.org/db/?f=f14914c88544d57f736ed6623000957c93bc4612 http://dev.laptop.org/~quozl/z/1dCw32.txt dmesg http://dev.laptop.org/~quozl/z/1dD2qx.txt dmidecode http://dev.laptop.org/~quozl/z/1dGJdK.txt dsdt.dsl I'm testing with 4.11.1 kernel on Ubuntu 17.04. I've tried latest UCM/chtrt5645/HiFi.conf c8b344c. I've tried using alsamixer to experiment with some of the 253 or so controls. I don't have a datasheet. (My previous work was with ALC5631, for which I do have a datasheet, and it was very helpful!) Also in the box was an NTC thermistor cabled to a TRRS plug, which may suggest a design with switchable decoupling capacitors. -- James Cameron http://quozl.netrek.org/

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[alsa-devel] [PATCH] oss: Fix a sleep-in-atomic bug in midi_outc
by Jia-Ju Bai 01 Jun '17

01 Jun '17

The driver may sleep under a spin lock, and the function call path is: midi_outc (acquire the lock by spin_lock_irqsave) oss_broken_sleep_on schedule_timeout --> may sleep To fix it, the lock is released before oss_broken_sleep_on, and the lock is acquired again after this function. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)163.com> --- sound/oss/sequencer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index f19da4b..3d95d752 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -1211,7 +1211,9 @@ static void midi_outc(int dev, unsigned char data) spin_lock_irqsave(&lock,flags); while (n && !midi_devs[dev]->outputc(dev, data)) { + spin_unlock_irqrestore(&lock, flags); oss_broken_sleep_on(&seq_sleeper, HZ/25); + spin_lock_irqsave(&lock, flags); n--; } spin_unlock_irqrestore(&lock,flags); -- 1.7.9.5

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[alsa-devel] [PATCH] ALSA: hda - Fix applying MSI dual-codec mobo quirk
by Takashi Iwai 01 Jun '17

01 Jun '17

The previous commit [63691587f7b0: ALSA: hda - Apply dual-codec quirk for MSI Z270-Gaming mobo] attempted to apply the existing dual-codec quirk for a MSI mobo. But it turned out that this isn't applied properly due to the MSI-vendor quirk before this entry. I overlooked such two MSI entries just because they were put in the wrong position, although we have a list ordered by PCI SSID numbers. This patch fixes it by rearranging the unordered entries. Fixes: 63691587f7b0 ("ALSA: hda - Apply dual-codec quirk for MSI Z270-Gaming mobo") Reported-by: Rudolf Schmidt <info(a)rudolfschmidt.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 918e45268915..a57988d617e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2324,11 +2324,11 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), - SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), - SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), -- 2.13.0

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[alsa-devel] Quick fix-update for alsa-lib 1.1.4?
by Takashi Iwai 01 Jun '17

01 Jun '17

Hi Jaroslav, it seems that alsa-lib 1.1.4 got a few regression (dmix binary incompatibility and the build failure for old glibc). Shall we release a quick fix version, 1.1.4a or 1.1.5, before the 1.1.4 is widely deployed? thanks, Takashi

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[alsa-devel] [PATCH 1/3] pcm: add support to get PCM sample rate from name
by guneshwor.o.singh＠intel.com 01 Jun '17

01 Jun '17

From: Guneshwor Singh <guneshwor.o.singh(a)intel.com> Get PCM sample rate from rate name similar to snd_pcm_format_value() which retrieves format from format name. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- include/pcm.h | 38 ++++++++++++++++++++++++++++++++++++++ src/pcm/pcm.c | 37 +++++++++++++++++++++++++++++++++++++ 2 files changed, 75 insertions(+) diff --git a/include/pcm.h b/include/pcm.h index 0be1a321..81bc55b1 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -119,6 +119,43 @@ typedef enum _snd_pcm_access { SND_PCM_ACCESS_LAST = SND_PCM_ACCESS_RW_NONINTERLEAVED } snd_pcm_access_t; +/** PCM sample rate */ +typedef enum _snd_pcm_rates { + /** Unknown */ + SND_PCM_RATE_UNKNOWN = -1, + /** 5512 Hz sample rate */ + SND_PCM_RATE_5512 = 0, + /** 8000 Hz sample rate */ + SND_PCM_RATE_8000, + /** 11025 Hz sample rate */ + SND_PCM_RATE_11025, + /** 16000 Hz sample rate */ + SND_PCM_RATE_16000, + /** 22050 Hz sample rate */ + SND_PCM_RATE_22050, + /** 32000 Hz sample rate */ + SND_PCM_RATE_32000, + /** 44100 Hz sample rate */ + SND_PCM_RATE_44100, + /** 48000 Hz sample rate */ + SND_PCM_RATE_48000, + /** 64000 Hz sample rate */ + SND_PCM_RATE_64000, + /** 88200 Hz sample rate */ + SND_PCM_RATE_88200, + /** 96000 Hz sample rate */ + SND_PCM_RATE_96000, + /** 176400 Hz sample rate */ + SND_PCM_RATE_176400, + /** 192000 Hz sample rate */ + SND_PCM_RATE_192000, + /** continuous range within rate_min and rate_max */ + SND_PCM_RATE_CONTINUOUS = 30, + /** more continuous range */ + SND_PCM_RATE_KNOT = 31, + SND_PCM_RATE_LAST = SND_PCM_RATE_KNOT, +} snd_pcm_rates_t; + /** PCM sample format */ typedef enum _snd_pcm_format { /** Unknown */ @@ -1047,6 +1084,7 @@ const char *snd_pcm_format_description(const snd_pcm_format_t format); const char *snd_pcm_subformat_name(const snd_pcm_subformat_t subformat); const char *snd_pcm_subformat_description(const snd_pcm_subformat_t subformat); snd_pcm_format_t snd_pcm_format_value(const char* name); +snd_pcm_rates_t snd_pcm_rate_value(const char* name); const char *snd_pcm_tstamp_mode_name(const snd_pcm_tstamp_t mode); const char *snd_pcm_state_name(const snd_pcm_state_t state); diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 200b10c2..bff1b949 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1722,6 +1722,7 @@ static int __snd_pcm_poll_revents(snd_pcm_t *pcm, struct pollfd *pfds, #define START(v) [SND_PCM_START_##v] = #v #define HW_PARAM(v) [SND_PCM_HW_PARAM_##v] = #v #define SW_PARAM(v) [SND_PCM_SW_PARAM_##v] = #v +#define RATE(v) [SND_PCM_RATE_##v] = #v #define FORMAT(v) [SND_PCM_FORMAT_##v] = #v #define SUBFORMAT(v) [SND_PCM_SUBFORMAT_##v] = #v @@ -1754,6 +1755,24 @@ static const char *const snd_pcm_access_names[] = { ACCESS(RW_NONINTERLEAVED), }; +static const char *const snd_pcm_rate_names[] = { + RATE(5512), + RATE(8000), + RATE(11025), + RATE(16000), + RATE(22050), + RATE(32000), + RATE(44100), + RATE(48000), + RATE(64000), + RATE(88200), + RATE(96000), + RATE(176400), + RATE(192000), + RATE(CONTINUOUS), + RATE(KNOT), +}; + static const char *const snd_pcm_format_names[] = { FORMAT(S8), FORMAT(U8), @@ -1978,6 +1997,24 @@ const char *snd_pcm_format_description(snd_pcm_format_t format) } /** + * \brief get PCM sample rate from name + * \param name PCM sample rate name (case insensitive) + * \return PCM sample rate + */ +snd_pcm_rates_t snd_pcm_rate_value(const char* name) +{ + snd_pcm_rates_t rate; + for (rate = 0; rate <= SND_PCM_RATE_LAST; rate++) { + if (snd_pcm_rate_names[rate] && + strcasecmp(name, snd_pcm_rate_names[rate]) == 0) { + return rate; + } + } + + return SND_PCM_RATE_UNKNOWN; +} + +/** * \brief get PCM sample format from name * \param name PCM sample format name (case insensitive) * \return PCM sample format -- 2.13.0

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