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[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix compile error when registering a DAI
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify some code which register a DAI with ASoC core according to modified registration API of DAI. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 3 +-- sound/soc/s3c24xx/s3c-i2s-v2.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index 69c41eb..007ed3a 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -268,8 +268,7 @@ static int __init aquila_init(void) return -ENOMEM; /* register voice DAI here */ - ret = snd_soc_register_dai(&aquila_snd_device->dev, - 0, &voice_dai); + ret = snd_soc_register_dai(&aquila_snd_device->dev, &voice_dai); if (ret) return ret; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index f4fbc0e..b3866d5 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -750,7 +750,7 @@ int s3c_i2sv2_register_dai(struct device *dev, int id, drv->suspend = s3c2412_i2s_suspend; drv->resume = s3c2412_i2s_resume; - return snd_soc_register_dai(dev, id, drv); + return snd_soc_register_dai(dev, drv); } EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai); -- 1.6.3.3

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[alsa-devel] multi-component: determine which DAI is active?
by Timur Tabi 22 Jul '10

22 Jul '10

Liam, In order to get fsl_dma.c to work with the WM8776, which has separate DAIs for playback and capture, I've had to do this: if (pcm->streams[0].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); if (ret) { dev_err(card->dev, "can't allocate playback dma buffer\n"); return ret; } } if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); dev_err(card->dev, "can't allocate capture dma buffer\n"); return ret; } } That is, I need to check if the 'substream' pointer has been initialized. This looks ugly to me. In an earlier version of ASoC, I would have done this: if (dai->playback.channels_min) { ... if (dai->capture.channels_min) { ... But dai->playback and dai->capture no longer exist. How should I handle this?

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[alsa-devel] [PATCH] ASoC: multi-component - Add Goni sound driver
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch add sound support for the Goni board based on S5PV210. The Goni board is based on Samsung SoC(S5PV210) and include WM8994 codec over I2S to support sound. The kind of jack is below states : * SND_JACK_HEADPHONE * SND_JACK_HEADSET * SND_JACK_MECHANICAL : When TV-OUT cable is inserted on Goni board, the TV-OUT cable isn't connected to television. * SND_JACK_AVOUT : When TV-OUT cable is inserted on Goni board, the TV-OUT cable is connected to television. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/Kconfig | 9 ++ sound/soc/s3c24xx/Makefile | 2 + sound/soc/s3c24xx/goni_wm8994.c | 298 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 309 insertions(+), 0 deletions(-) create mode 100644 sound/soc/s3c24xx/goni_wm8994.c diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index c09b402..1cdc37b 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -140,3 +140,12 @@ config SND_S5PC110_SOC_AQUILA_WM8994 help Say Y if you want to add support for SoC audio on aquila with the WM8994. + +config SND_S5PV210_SOC_GONI_WM8994 + tristate "SoC I2S Audio support for GONI - WM8994" + depends on SND_S3C24XX_SOC && MACH_GONI + select SND_S3C64XX_SOC_I2S_V4 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on goni + with the WM8994. diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 02dd12c..47ed6d7 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -31,6 +31,7 @@ snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-aquila-wm8994-objs := aquila_wm8994.o +snd-soc-goni-wm8994-objs := goni_wm8994.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -45,3 +46,4 @@ obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o obj-$(CONFIG_SND_SOC_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_S3C64XX_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_S5PC110_SOC_AQUILA_WM8994) += snd-soc-aquila-wm8994.o +obj-$(CONFIG_SND_S5PV210_SOC_GONI_WM8994) += snd-soc-goni-wm8994.o diff --git a/sound/soc/s3c24xx/goni_wm8994.c b/sound/soc/s3c24xx/goni_wm8994.c new file mode 100644 index 0000000..20bcaab --- /dev/null +++ b/sound/soc/s3c24xx/goni_wm8994.c @@ -0,0 +1,298 @@ +/* + * goni_wm8994.c + * + * Copyright (C) 2010 Samsung Electronics Co.Ltd + * Author: Chanwoo Choi <cw00.choi(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <asm/mach-types.h> +#include <mach/gpio.h> +#include <mach/regs-clock.h> + +#include <linux/mfd/wm8994/core.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +static struct snd_soc_card goni; +static struct platform_device *goni_snd_device; + +/* 3.5 pie jack */ +static struct snd_soc_jack jack; + +/* 3.5 pie jack detection DAPM pins */ +static struct snd_soc_jack_pin jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + }, +}; + +/* 3.5 pie jack detection gpios */ +static struct snd_soc_jack_gpio jack_gpios[] = { + { + .gpio = S5PV210_GPH0(6), + .name = "DET_3.5", + .report = SND_JACK_HEADSET | SND_JACK_MECHANICAL | + SND_JACK_AVOUT, + .debounce_time = 200, + }, +}; + +static const struct snd_soc_dapm_widget goni_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Left Spk", NULL), + SND_SOC_DAPM_SPK("Ext Right Spk", NULL), + SND_SOC_DAPM_SPK("Ext Rcv", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), + SND_SOC_DAPM_MIC("2nd Mic", NULL), + SND_SOC_DAPM_LINE("Radio In", NULL), +}; + +static const struct snd_soc_dapm_route goni_dapm_routes[] = { + {"Ext Left Spk", NULL, "SPKOUTLP"}, + {"Ext Left Spk", NULL, "SPKOUTLN"}, + + {"Ext Right Spk", NULL, "SPKOUTRP"}, + {"Ext Right Spk", NULL, "SPKOUTRN"}, + + {"Ext Rcv", NULL, "HPOUT2N"}, + {"Ext Rcv", NULL, "HPOUT2P"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + {"IN1RP", NULL, "Headset Mic"}, + + {"IN1RN", NULL, "2nd Mic"}, + {"IN1RP", NULL, "2nd Mic"}, + + {"IN1LN", NULL, "Main Mic"}, + {"IN1LP", NULL, "Main Mic"}, + + {"IN2LN", NULL, "Radio In"}, + {"IN2RN", NULL, "Radio In"}, +}; + +static int goni_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + /* add goni specific widgets */ + snd_soc_dapm_new_controls(codec, goni_dapm_widgets, + ARRAY_SIZE(goni_dapm_widgets)); + + /* set up goni specific audio routes */ + snd_soc_dapm_add_routes(codec, goni_dapm_routes, + ARRAY_SIZE(goni_dapm_routes)); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(codec, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(codec, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT1P"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2N"); + snd_soc_dapm_nc_pin(codec, "LINEOUT2P"); + + snd_soc_dapm_sync(codec); + + /* Headset jack detection */ + ret = snd_soc_jack_new(&goni, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_MECHANICAL | SND_JACK_AVOUT, + &jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&jack, ARRAY_SIZE(jack_pins), jack_pins); + if (ret) + return ret; + + ret = snd_soc_jack_add_gpios(&jack, ARRAY_SIZE(jack_gpios), jack_gpios); + if (ret) + return ret; + + return 0; +} + +static int goni_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + /* set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the cpu system clock */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops goni_hifi_ops = { + .hw_params = goni_hifi_hw_params, +}; + +static int goni_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int pll_out = 24000000; + int ret = 0; + + if (params_rate(params) != 8000) + return -EINVAL; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set the codec FLL */ + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL2, 0, pll_out, + params_rate(params) * 256); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL2, + params_rate(params) * 256, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver voice_dai = { + .name = "goni-voice-dai", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, +}; + +static struct snd_soc_ops goni_voice_ops = { + .hw_params = goni_voice_hw_params, +}; + +static struct snd_soc_dai_link goni_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "s3c64xx-i2s-v4-dai", + .codec_dai_name = "wm8994-hifi", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .init = goni_wm8994_init, + .ops = &goni_hifi_ops, +}, { + .name = "WM8994 Voice", + .stream_name = "Voice", + .cpu_dai_name = "goni-voice-dai", + .codec_dai_name = "wm8994-voice", + .platform_name = "s3c24xx-pcm-audio", + .codec_name = "wm8994-codec", + .ops = &goni_voice_ops, +}, +}; + +static struct snd_soc_card goni = { + .name = "goni", + .dai_link = goni_dai, + .num_links = ARRAY_SIZE(goni_dai), +}; + +static int __init goni_init(void) +{ + int ret; + + if (!machine_is_goni()) + return -ENODEV; + + goni_snd_device = platform_device_alloc("soc-audio", -1); + if (!goni_snd_device) + return -ENOMEM; + + /* register voice DAI here */ + ret = snd_soc_register_dai(&goni_snd_device->dev, &voice_dai); + if (ret) + return ret; + + platform_set_drvdata(goni_snd_device, &goni); + ret = platform_device_add(goni_snd_device); + + if (ret) + platform_device_put(goni_snd_device); + + return ret; +} + +static void __exit goni_exit(void) +{ + platform_device_unregister(goni_snd_device); +} + +module_init(goni_init); +module_exit(goni_exit); + +/* Module information */ +MODULE_DESCRIPTION("ALSA SoC WM8994 GONI(S5PV210)"); +MODULE_AUTHOR("Chanwoo Choi <cw00.choi(a)samsung.com>"); +MODULE_LICENSE("GPL"); -- 1.6.3.3

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[alsa-devel] [PATCH] ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
by Chanwoo Choi 22 Jul '10

22 Jul '10

This patch modify the wrong field name on Aquila board. Signed-off-by: Chanwoo Choi <cw00.choi(a)samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park(a)samsung.com> --- sound/soc/s3c24xx/aquila_wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/s3c24xx/aquila_wm8994.c b/sound/soc/s3c24xx/aquila_wm8994.c index f6f0e60..69c41eb 100644 --- a/sound/soc/s3c24xx/aquila_wm8994.c +++ b/sound/soc/s3c24xx/aquila_wm8994.c @@ -242,7 +242,7 @@ static struct snd_soc_dai_link aquila_dai[] = { }, { .name = "WM8994 Voice", .stream_name = "Voice", - .cpu_dai_drv = "aquila-voice-dai", + .cpu_dai_name = "aquila-voice-dai", .codec_dai_name = "wm8994-voice", .platform_name = "s3c24xx-pcm-audio", .codec_name = "wm8994-codec", -- 1.6.3.3

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[alsa-devel] problems about converting sample rates when using ALSA-LIB
by longer 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-lib-1.0.23), I found that : It mmap a address from my driver , but writing audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data . But I don't know why it don't write audio data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

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[alsa-devel] problems about converting sample rate when using ALSA lib
by longer zhou (周龙) 22 Jul '10

22 Jul '10

Hi all, I have a problem when using ALSA lib . The problem is : My ALSA driver only support a special sample rate , so ALSA lib must convert sample rates. But My driver don't get any audio data. When tracing the code of ALSA lib (alsa-driver-1.0.23), I found that : It mmap a address from my driver , but writting audio data to another address instead of that mmaped address. I think this is why my driver don't get any audio data .But I don't know why it don't write audo data to that mmaped address. Can you help me ? Any advices would help me a lot. Thank you, longer.zhou

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[alsa-devel] add support for Codec: Conexant ID 5068 (not supported by ALSA yet)
by ccamacho＠linuxmail.org 22 Jul '10

22 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

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[alsa-devel] Small memory leak in snd_pcm_ladspa_init()
by John Lindgren 22 Jul '10

22 Jul '10

Hello, Valgrind shows up a small memory leak in libasound when using a LADSPA plugin. Sorry, I don't have a patch this time. ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,976 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,977 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ... ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,980 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BEFD: snd_pcm_ladspa_init (pcm_ladspa.c:744) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ==30783== 16 bytes in 2 blocks are definitely lost in loss record 1,981 of 6,954 ==30783== at 0x4C2380C: calloc (vg_replace_malloc.c:467) ==30783== by 0x1B06BF25: snd_pcm_ladspa_init (pcm_ladspa.c:746) ==30783== by 0x1B04B325: snd_pcm_plugin_prepare (pcm_plugin.c:167) ==30783== by 0x1B04B2EE: snd_pcm_plugin_prepare (pcm_plugin.c:158) ==30783== by 0x1B03B2DC: snd_pcm_recover (pcm.c:7246) ... and so on. -- John

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[alsa-devel] add support for Codec: Conexant ID 5068 (not present in ALSA yet)
by ccamacho＠linuxmail.org 21 Jul '10

21 Jul '10

Hi all, The codec (Conexant ID 5068) required by my sound card (Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06)) is not present yet in alsa, I would highly appreciate if you guys could add support for it. You guys might find the information collected by the alsa-info.sh script on http://www.alsa-project.org/db/?f=8f8001d873d5c10c293eac1c94f2ede0b73cd865 Thanks a lot in advance for looking into this. Cheers, Cristopher

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[alsa-devel] Compiler errors in alsa-lib-1.0.22/test/*.c
by John Simpson 21 Jul '10

21 Jul '10

Howdy, I've been trying to use the alsa-lib API, but I haven't been able to get anything to compile. ================= ~/alsa/alsa-lib-1.0.22/test$ for f in *.c; do echo; cc $f 2>&1 | head -n2; done /tmp/ccNjLWLK.o: In function `dump_event_filter': client_event_filter.c:(.text+0x23): undefined reference to `snd_seq_client_info_event_filter_check' /tmp/ccNLDqdU.o: In function `main': control.c:(.text+0x1c): undefined reference to `snd_ctl_card_info_sizeof' /tmp/ccBehb51.o: In function `setparams_stream': latency.c:(.text+0x24): undefined reference to `snd_pcm_hw_params_any' /usr/lib/gcc/x86_64-linux-gnu/4.4.3/../../../../lib/crt1.o: In function `_start': (.text+0x20): undefined reference to `main' midiloop.c: In function ‘main’: midiloop.c:172: warning: format ‘%i’ expects type ‘int’, but argument 2 has type ‘size_t’ namehint.c: In function ‘main’: namehint.c:12: warning: passing argument 3 of ‘snd_device_name_hint’ from incompatible pointer type oldapi.c: In function ‘main’: oldapi.c:40: warning: assignment from incompatible pointer type /tmp/ccmB528p.o: In function `generate_sine': pcm.c:(.text+0x119): undefined reference to `snd_pcm_format_width' /tmp/ccChgWYC.o: In function `main': pcm_min.c:(.text+0x4d): undefined reference to `snd_pcm_open' /tmp/cc9FiONG.o: In function `write_ev': playmidi1.c:(.text+0x1b65): undefined reference to `snd_seq_event_output' /tmp/ccQkntBX.o: In function `main': queue_timer.c:(.text+0x1d0): undefined reference to `snd_seq_open' /tmp/ccmIMCZ4.o: In function `main': rawmidi.c:(.text+0x4e3): undefined reference to `snd_rawmidi_open' /tmp/cc3g8KGc.o: In function `event_decoder_start_timer': seq.c:(.text+0x4e2): undefined reference to `snd_seq_control_queue' seq-decoder.c:6: error: ‘SND_SEQ_EVENT_SYSTEM’ undeclared here (not in a function) seq-decoder.c:6: error: array index in initializer not of integer type seq-sender.c:68: error: expected ‘)’ before ‘*’ token seq-sender.c:100: error: expected ‘)’ before ‘*’ token /tmp/ccxyZfhg.o: In function `show_status': timer.c:(.text+0x1d): undefined reference to `snd_timer_status_sizeof' ================= Did I miss something? TIA -- John Wesley Simpson SwaJime's Cove℠ www.swajime.com

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