[alsa-devel] [PATCH v3 0/1] Codec driver for TI WL1273 FM Radio.
Hello all.
Thank you for comments and thanks to Peter for advice on constraints and ALSA in general.
New in this version:
1. Now constraints are used in the start-up function. It wasn't as big a deal as I thought... Also now the codec is always in full duplex mode. But it still refuses to do the wrong thing with FM as real full duplex is then impossible.
2. Fixed the exported mode support function. The name in now wl1273_get_format and it returns the audio interface format.
3. Removed the remaining obsolete functions.
4. Not actually new but...
I didn't do this yet. But could you give me the git URL and then I'll resend?
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git
I tried the corresponding http URL
http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git but got
fatal: http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git/info/refs not found: did you run git update-server-info on the server?
Cheers, Matti
Matti J. Aaltonen (1): ASoC: TI WL1273 FM Radio Codec.
sound/soc/codecs/wl1273.c | 598 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 43 ++++ 2 files changed, 641 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h
This is an ALSA codec for the Texas Instruments WL1273 FM Radio.
Signed-off-by: Matti J. Aaltonen matti.j.aaltonen@nokia.com --- sound/soc/codecs/wl1273.c | 598 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wl1273.h | 43 ++++ 2 files changed, 641 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wl1273.c create mode 100644 sound/soc/codecs/wl1273.h
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c new file mode 100644 index 0000000..c9a9269 --- /dev/null +++ b/sound/soc/codecs/wl1273.c @@ -0,0 +1,598 @@ +/* + * ALSA SoC WL1273 codec driver + * + * Author: Matti Aaltonen, matti.j.aaltonen@nokia.com + * + * Copyright: (C) 2010 Nokia Corporation + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/mfd/wl1273-core.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wl1273.h" + +static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core, + int rate, int width) +{ + struct device *dev = &core->i2c_dev->dev; + int r = 0; + u16 mode; + + dev_dbg(dev, "rate: %d\n", rate); + dev_dbg(dev, "width: %d\n", width); + + mutex_lock(&core->lock); + + mode = core->i2s_mode & ~WL1273_IS2_WIDTH & ~WL1273_IS2_RATE; + + switch (rate) { + case 48000: + mode |= WL1273_IS2_RATE_48K; + break; + case 44100: + mode |= WL1273_IS2_RATE_44_1K; + break; + case 32000: + mode |= WL1273_IS2_RATE_32K; + break; + case 22050: + mode |= WL1273_IS2_RATE_22_05K; + break; + case 16000: + mode |= WL1273_IS2_RATE_16K; + break; + case 12000: + mode |= WL1273_IS2_RATE_12K; + break; + case 11025: + mode |= WL1273_IS2_RATE_11_025; + break; + case 8000: + mode |= WL1273_IS2_RATE_8K; + break; + default: + dev_err(dev, "Sampling rate: %d not supported\n", rate); + r = -EINVAL; + goto out; + } + + switch (width) { + case 16: + mode |= WL1273_IS2_WIDTH_32; + break; + case 20: + mode |= WL1273_IS2_WIDTH_40; + break; + case 24: + mode |= WL1273_IS2_WIDTH_48; + break; + case 25: + mode |= WL1273_IS2_WIDTH_50; + break; + case 30: + mode |= WL1273_IS2_WIDTH_60; + break; + case 32: + mode |= WL1273_IS2_WIDTH_64; + break; + case 40: + mode |= WL1273_IS2_WIDTH_80; + break; + case 48: + mode |= WL1273_IS2_WIDTH_96; + break; + case 64: + mode |= WL1273_IS2_WIDTH_128; + break; + default: + dev_err(dev, "Data width: %d not supported\n", width); + r = -EINVAL; + goto out; + } + + dev_dbg(dev, "WL1273_I2S_DEF_MODE: 0x%04x\n", WL1273_I2S_DEF_MODE); + dev_dbg(dev, "core->i2s_mode: 0x%04x\n", core->i2s_mode); + dev_dbg(dev, "mode: 0x%04x\n", mode); + + if (core->i2s_mode != mode) { + r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, + mode); + if (!r) + core->i2s_mode = mode; + + r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE, + WL1273_AUDIO_ENABLE_I2S); + if (r) + goto out; + } +out: + mutex_unlock(&core->lock); + + return r; +} + +static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core, + int channel_number) +{ + struct i2c_client *client = core->i2c_dev; + struct device *dev = &client->dev; + int r = 0; + + dev_dbg(dev, "%s\n", __func__); + + mutex_lock(&core->lock); + + if (core->channel_number == channel_number) + goto out; + + if (channel_number == 1 && core->mode == WL1273_MODE_RX) + r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, + WL1273_RX_MONO); + else if (channel_number == 1 && core->mode == WL1273_MODE_TX) + r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, + WL1273_TX_MONO); + else if (channel_number == 2 && core->mode == WL1273_MODE_RX) + r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET, + WL1273_RX_STEREO); + else if (channel_number == 2 && core->mode == WL1273_MODE_TX) + r = wl1273_fm_write_cmd(core, WL1273_MONO_SET, + WL1273_TX_STEREO); + else + r = -EINVAL; +out: + mutex_unlock(&core->lock); + + return r; +} + +static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = wl1273->mode; + + return 0; +} + +static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" }; + +static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + /* Do not allow changes while stream is running */ + if (codec->active) + return -EPERM; + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] >= ARRAY_SIZE(wl1273_audio_route)) + return -EINVAL; + + wl1273->mode = ucontrol->value.integer.value[0]; + + return 1; +} + +static const struct soc_enum wl1273_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route); + +static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + ucontrol->value.integer.value[0] = wl1273->core->audio_mode; + + return 0; +} + +static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + int val, r = 0; + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + val = ucontrol->value.integer.value[0]; + if (wl1273->core->audio_mode == val) + return 0; + + r = wl1273_fm_set_audio(wl1273->core, val); + if (r < 0) + return r; + + return 1; +} + +static const char *wl1273_audio_strings[] = { "Digital", "Analog" }; + +static const struct soc_enum wl1273_audio_enum = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), + wl1273_audio_strings); + +static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + ucontrol->value.integer.value[0] = wl1273->core->volume; + + return 0; +} + +static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + int r; + + dev_dbg(codec->dev, "%s: enter.\n", __func__); + + r = wl1273_fm_set_volume(wl1273->core, + ucontrol->value.integer.value[0]); + if (r) + return r; + + return 1; +} + +static const struct snd_kcontrol_new wl1273_controls[] = { + SOC_ENUM_EXT("Codec Mode", wl1273_enum, + snd_wl1273_get_audio_route, snd_wl1273_set_audio_route), + SOC_ENUM_EXT("Audio Switch", wl1273_audio_enum, + snd_wl1273_fm_audio_get, snd_wl1273_fm_audio_put), + SOC_SINGLE_EXT("Volume", 0, 0, WL1273_MAX_VOLUME, 0, + snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), +}; + +static int wl1273_add_controls(struct snd_soc_codec *codec) +{ + return snd_soc_add_controls(codec, wl1273_controls, + ARRAY_SIZE(wl1273_controls)); +} + +static int wl1273_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + switch (wl1273->mode) { + case WL1273_MODE_BT: + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 8000, 8000); + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1); + break; + case WL1273_MODE_FM_RX: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_err("Cannot play in RX mode.\n"); + return -EINVAL; + } + break; + case WL1273_MODE_FM_TX: + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + pr_err("Cannot capture in TX mode.\n"); + return -EINVAL; + } + break; + default: + return -EINVAL; + break; + } + + return 0; +} + +static int wl1273_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + struct wl1273_core *core = wl1273->core; + unsigned int rate, width, r; + + if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) { + pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n"); + return -EINVAL; + } + + rate = params_rate(params); + width = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + + if (wl1273->mode == WL1273_MODE_BT) { + if (rate != 8000) { + pr_err("Rate %d not supported.\n", params_rate(params)); + return -EINVAL; + } + + if (params_channels(params) != 1) { + pr_err("Only mono supported.\n"); + return -EINVAL; + } + + return 0; + } + + if (wl1273->mode == WL1273_MODE_FM_TX && + substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + pr_err("Only playback supported with TX.\n"); + return -EINVAL; + } + + if (wl1273->mode == WL1273_MODE_FM_RX && + substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + pr_err("Only capture supported with RX.\n"); + return -EINVAL; + } + + if (wl1273->mode != WL1273_MODE_FM_RX && + wl1273->mode != WL1273_MODE_FM_TX) { + pr_err("Unexpected mode: %d.\n", wl1273->mode); + return -EINVAL; + } + + r = snd_wl1273_fm_set_i2s_mode(core, rate, width); + if (r) + return r; + + wl1273->channels = params_channels(params); + r = snd_wl1273_fm_set_channel_number(core, wl1273->channels); + if (r) + return r; + + return 0; +} + +static struct snd_soc_dai_ops wl1273_dai_ops = { + .startup = wl1273_startup, + .hw_params = wl1273_hw_params, +}; + +struct snd_soc_dai wl1273_dai = { + .name = "WL1273 BT/FM codec", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE}, + .ops = &wl1273_dai_ops, +}; +EXPORT_SYMBOL_GPL(wl1273_dai); + +/* Audio interface format for the soc_card driver */ +int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt) +{ + struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec); + + switch (wl1273->channels) { + /* Stereo */ + case 2: + *fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + break; + /* Mono */ + case 1: + *fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + break; + default: + return -EINVAL; + } + + return 0; +} +EXPORT_SYMBOL_GPL(wl1273_get_format); + +static struct snd_soc_codec *wl1273_codec; + +/* + * initialize the driver + * register the mixer and dsp interfaces with the kernel + */ +static int wl1273_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct wl1273_priv *wl1273; + int r = 0; + + dev_dbg(&pdev->dev, "%s.\n", __func__); + + codec = wl1273_codec; + wl1273 = snd_soc_codec_get_drvdata(codec); + socdev->card->codec = codec; + + codec->name = "wl1273"; + codec->owner = THIS_MODULE; + codec->dai = &wl1273_dai; + codec->num_dai = 1; + + /* register pcms */ + r = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (r < 0) { + dev_err(&pdev->dev, "Wl1273: failed to create pcms.\n"); + goto err2; + } + + r = wl1273_add_controls(codec); + if (r < 0) { + dev_err(&pdev->dev, "Wl1273: failed to add contols.\n"); + goto err1; + } + + return r; +err1: + snd_soc_free_pcms(socdev); +err2: + return r; +} + +static int wl1273_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + + return 0; +} + +static int __devinit wl1273_codec_probe(struct platform_device *pdev) +{ + struct wl1273_core **pdata = pdev->dev.platform_data; + struct snd_soc_codec *codec; + struct wl1273_priv *wl1273; + int r; + + dev_dbg(&pdev->dev, "%s.\n", __func__); + + if (!pdata) { + dev_err(&pdev->dev, "Platform data is missing.\n"); + return -EINVAL; + } + + wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL); + if (wl1273 == NULL) { + dev_err(&pdev->dev, "Cannot allocate memory.\n"); + return -ENOMEM; + } + + wl1273->mode = WL1273_MODE_BT; + wl1273->core = *pdata; + + codec = &wl1273->codec; + snd_soc_codec_set_drvdata(codec, wl1273); + codec->dev = &pdev->dev; + wl1273_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "wl1273"; + codec->owner = THIS_MODULE; + codec->dai = &wl1273_dai; + codec->num_dai = 1; + + platform_set_drvdata(pdev, wl1273); + wl1273_codec = codec; + + codec->bias_level = SND_SOC_BIAS_OFF; + + r = snd_soc_register_codec(codec); + if (r != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", r); + goto err2; + } + + r = snd_soc_register_dai(&wl1273_dai); + if (r != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", r); + goto err1; + } + + return 0; + +err1: + snd_soc_unregister_codec(codec); +err2: + kfree(wl1273); + return r; +} + +static int __devexit wl1273_codec_remove(struct platform_device *pdev) +{ + struct wl1273_priv *wl1273 = platform_get_drvdata(pdev); + + dev_dbg(&pdev->dev, "%s\n", __func__); + + snd_soc_unregister_dai(&wl1273_dai); + snd_soc_unregister_codec(&wl1273->codec); + + kfree(wl1273); + wl1273_codec = NULL; + + return 0; +} + +MODULE_ALIAS("platform:wl1273_codec_audio"); + +static struct platform_driver wl1273_codec_driver = { + .probe = wl1273_codec_probe, + .remove = __devexit_p(wl1273_codec_remove), + .driver = { + .name = "wl1273_codec_audio", + .owner = THIS_MODULE, + }, +}; + +static int __init wl1273_modinit(void) +{ + return platform_driver_register(&wl1273_codec_driver); +} +module_init(wl1273_modinit); + +static void __exit wl1273_exit(void) +{ + platform_driver_unregister(&wl1273_codec_driver); +} +module_exit(wl1273_exit); + +struct snd_soc_codec_device soc_codec_dev_wl1273 = { + .probe = wl1273_soc_probe, + .remove = wl1273_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wl1273); + +MODULE_AUTHOR("Matti Aaltonen matti.j.aaltonen@nokia.com"); +MODULE_DESCRIPTION("ASoC WL1273 codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h new file mode 100644 index 0000000..087e587 --- /dev/null +++ b/sound/soc/codecs/wl1273.h @@ -0,0 +1,43 @@ +/* + * sound/soc/codec/wl1273.h + * + * ALSA SoC WL1273 codec driver + * + * Copyright (C) Nokia Corporation + * Author: Matti Aaltonen matti.j.aaltonen@nokia.com + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * 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. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __WL1273_CODEC_H__ +#define __WL1273_CODEC_H__ + +enum wl1273_mode { WL1273_MODE_BT, WL1273_MODE_FM_RX, WL1273_MODE_FM_TX }; + +/* codec private data */ +struct wl1273_priv { + struct snd_soc_codec codec; + enum wl1273_mode mode; + struct wl1273_core *core; + unsigned int channels; +}; + +extern struct snd_soc_dai wl1273_dai; +extern struct snd_soc_codec_device soc_codec_dev_wl1273; + +int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt); + +#endif /* End of __WL1273_CODEC_H__ */
On Fri, 2010-07-23 at 15:42 +0300, Matti J. Aaltonen wrote:
Hello all.
Thank you for comments and thanks to Peter for advice on constraints and ALSA in general.
New in this version:
- Now constraints are used in the start-up function. It wasn't as big
a deal as I thought... Also now the codec is always in full duplex mode. But it still refuses to do the wrong thing with FM as real full duplex is then impossible.
- Fixed the exported mode support function. The name in now
wl1273_get_format and it returns the audio interface format.
Removed the remaining obsolete functions.
Not actually new but...
I didn't do this yet. But could you give me the git URL and then I'll resend?
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git
I tried the corresponding http URL
http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git but got
fatal: http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git/info/refs not found: did you run git update-server-info on the server?
Oh, that's the wrong URL. The correct one is :-
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git
Can you re-do against multi-component. The changes all relate to probe() and remove().
Thanks
Liam
Hi.
On Fri, 2010-07-23 at 20:24 +0200, ext Liam Girdwood wrote:
On Fri, 2010-07-23 at 15:42 +0300, Matti J. Aaltonen wrote:
Hello all.
Thank you for comments and thanks to Peter for advice on constraints and ALSA in general.
New in this version:
- Now constraints are used in the start-up function. It wasn't as big
a deal as I thought... Also now the codec is always in full duplex mode. But it still refuses to do the wrong thing with FM as real full duplex is then impossible.
- Fixed the exported mode support function. The name in now
wl1273_get_format and it returns the audio interface format.
Removed the remaining obsolete functions.
Not actually new but...
I didn't do this yet. But could you give me the git URL and then I'll resend?
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git
I tried the corresponding http URL
http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git but got
fatal: http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git/info/refs not found: did you run git update-server-info on the server?
Oh, that's the wrong URL. The correct one is :-
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git
Can you re-do against multi-component. The changes all relate to probe() and remove().
Yes. No problem... I actually woke up last night and realized that the format function I wrote didn't have a NULL pointer check for the value parameter... So please ignore the v2. patch. I'll send version v2.B soon...
Cheers, Matti.
Thanks
Liam
On Sat, 2010-07-24 at 11:05 +0300, Matti J. Aaltonen wrote:
Hi.
On Fri, 2010-07-23 at 20:24 +0200, ext Liam Girdwood wrote:
On Fri, 2010-07-23 at 15:42 +0300, Matti J. Aaltonen wrote:
Hello all.
Thank you for comments and thanks to Peter for advice on constraints and ALSA in general.
New in this version:
- Now constraints are used in the start-up function. It wasn't as big
a deal as I thought... Also now the codec is always in full duplex mode. But it still refuses to do the wrong thing with FM as real full duplex is then impossible.
- Fixed the exported mode support function. The name in now
wl1273_get_format and it returns the audio interface format.
Removed the remaining obsolete functions.
Not actually new but...
I didn't do this yet. But could you give me the git URL and then I'll resend?
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git
I tried the corresponding http URL
http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git but got
fatal: http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git/info/refs not found: did you run git update-server-info on the server?
Oh, that's the wrong URL. The correct one is :-
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git
Can you re-do against multi-component. The changes all relate to probe() and remove().
Yes. No problem... I actually woke up last night and realized that the format function I wrote didn't have a NULL pointer check for the value parameter... So please ignore the v2. patch. I'll send version v2.B soon...
Ok, thanks.
Btw, now that we have a naming convention for components how about this having a "-radio" or "-fm" device suffix ?
Liam
On Sat, 2010-07-24 at 11:46 +0200, ext Liam Girdwood wrote:
On Sat, 2010-07-24 at 11:05 +0300, Matti J. Aaltonen wrote:
Hi.
On Fri, 2010-07-23 at 20:24 +0200, ext Liam Girdwood wrote:
On Fri, 2010-07-23 at 15:42 +0300, Matti J. Aaltonen wrote:
Hello all.
Thank you for comments and thanks to Peter for advice on constraints and ALSA in general.
New in this version:
- Now constraints are used in the start-up function. It wasn't as big
a deal as I thought... Also now the codec is always in full duplex mode. But it still refuses to do the wrong thing with FM as real full duplex is then impossible.
- Fixed the exported mode support function. The name in now
wl1273_get_format and it returns the audio interface format.
Removed the remaining obsolete functions.
Not actually new but...
I didn't do this yet. But could you give me the git URL and then I'll resend?
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git
I tried the corresponding http URL
http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git but got
fatal: http://www.kernel.org/pub/scm/linux/kernel/git/lrg/sound-2.6.git/info/refs not found: did you run git update-server-info on the server?
Oh, that's the wrong URL. The correct one is :-
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git
Can you re-do against multi-component. The changes all relate to probe() and remove().
Yes. No problem... I actually woke up last night and realized that the format function I wrote didn't have a NULL pointer check for the value parameter... So please ignore the v2. patch. I'll send version v2.B soon...
Ok, thanks.
Btw, now that we have a naming convention for components how about this having a "-radio" or "-fm" device suffix ?
That's not a problem either. I also noticed when comparing to other codecs that some function names could be better...
But I saw that there are quite a lot changes that need to be done... so this is something I can't do in a couple of hours. I'll go back to my vacation and do it properly when I come back... after one week.
B.R. Matti
Liam
participants (2)
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Liam Girdwood
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Matti J. Aaltonen