April 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/8] ASoC: generic-dmaengine-pcm: Add support for half-duplex
by Lars-Peter Clausen 13 May '13

13 May '13

Some platforms which are half-duplex share the same DMA channel between the playback and capture stream. Add support for this to the generic dmaengine PCM driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/dmaengine_pcm.h | 5 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 43 ++++++++++++++++++++++++----------- 2 files changed, 35 insertions(+), 13 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b1d1150..f11c35c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -91,6 +91,11 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * bytes that are still left to transfer. */ #define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2) +/* + * The PCM is half duplex and the DMA channel is shared between capture and + * playback. + */ +#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ae0c37e..5fd5ed4 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -29,7 +29,7 @@ struct dmaengine_pcm { struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; - bool compat; + unsigned int flags; }; static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) @@ -128,6 +128,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) + return pcm->chan[0]; + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); @@ -148,7 +151,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - if (!pcm->chan[i] && pcm->compat) { + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); } @@ -215,6 +218,25 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; +static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, + struct device_node *of_node) +{ + unsigned int i; + + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + return; + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { + pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[1] = pcm->chan[0]; + } else { + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + pcm->chan[i] = of_dma_request_slave_channel(of_node, + dmaengine_pcm_dma_channel_names[i]); + } + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -225,23 +247,15 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; - unsigned int i; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; pcm->config = config; + pcm->flags = flags; - if (flags & SND_DMAENGINE_PCM_FLAG_COMPAT) - pcm->compat = true; - - if (!(flags & SND_DMAENGINE_PCM_FLAG_NO_DT) && dev->of_node) { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(dev->of_node, - dmaengine_pcm_dma_channel_names[i]); - } - } + dmaengine_pcm_request_chan_of(pcm, dev->of_node); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, @@ -272,8 +286,11 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) + if (pcm->chan[i]) { dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } } snd_soc_remove_platform(platform); -- 1.8.0

3 19

[alsa-devel] [PATCH 1/2] ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable
by Fabio Estevam 06 May '13

06 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Variable 'ret' is not needed here, so just remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b1d9b5e..4f74b05 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -116,7 +116,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i, ret = 0; + int i; if (!np) return 1; /* no device tree */ @@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return ret; + return 0; } static int mxs_sgtl5000_probe(struct platform_device *pdev) -- 1.7.9.5

3 5

[alsa-devel] [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
by mengdong.lin＠intel.com 06 May '13

06 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. This patch - adds a codec flag to delay resuming a codec. System resume will skip the codecs if this flag is set, and these codecs will be resumed on later codec access. - adds a set_power_state ops for Haswell HDMI codec. In a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..bcb7205 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5508,11 +5508,15 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca172..5b5e5f4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -886,6 +886,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..d116908 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -98,6 +98,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -977,6 +985,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(&spec->pins[pin_idx], 1); snd_hda_jack_report_sync(codec); } @@ -1846,6 +1861,63 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + spec->ready_to_resume = 0; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = &spec->pins[pin_idx]; + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1868,6 +1940,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

4 51

[alsa-devel] [RFC] Sound/HID: wiimote: add speaker support
by David Herrmann 05 May '13

05 May '13

Classic Wii Remotes provide a speaker on the device. We can stream PCM data, mute and change volume via special protocol requests and remote audio registers. The device supports several different formats, but fully understood are only signed 8-bit PCM and something that looks like 4-bit Yamaha ADPCM. Theoretically, we can set data-rates from 183Hz up to 12MHz, but a realistic range for Bluetooth l2cap is 1500-4000 Hz. Data is streamed as 20bytes blocks and must be sent in a constant rate. There is no way to read the current position. It would be way too slow, anyway. Signed-off-by: David Herrmann <dh.herrmann(a)gmail.com> --- Hi I'm reworking large parts of the HID Nintendo Wii Remote driver and wanted to add speaker support. I have never worked with the sound subsystem until now and need some help. The Wii Remote is a Bluetooth gamepad that features a small speaker on the remote. All information was gathered via reverse-engineering so I cannot tell you any specifics about the hardware, sorry. What I figured out so far is how to set up the speaker, stream data, set volume and mute the speaker. Supported formats are: signed 8-bit PCM 4-bit Yamaha ADPCM I implemented only the 8-bit PCM as I couldn't figure out whether the kernel supports the other format? The wiiproto_* helpers in hid-wiimote-core.c are used to send the requests. Data is streamed as 20-bytes packets and must be sent at a constant rate. I would really appreciate if someone can have a look at hid-wiimote-speaker.c and help me fill the gaps. I figured out how to write the basic snd_card management and PCM setup, but I am stuck with buffer management now. I have a function called wiiproto_req_audio() which sends up to 20bytes of PCM data to the device. However, I cannot figure out how to integrate that into the snd_pcm object. The problem is that all other drivers I found have large DMA buffers which are filled whenever an interrupt signals more data. However, I don't have this setup but instead I need a timer (does the sound-core provide that?) which causes the "copy" callback to be called with 20bytes of data at a constant rate. Any help welcome! And if this whole idea is stupid, please tell me! ;) Thanks David drivers/hid/Kconfig | 3 + drivers/hid/Makefile | 3 + drivers/hid/hid-wiimote-core.c | 54 +++- drivers/hid/hid-wiimote-modules.c | 5 + drivers/hid/hid-wiimote-speaker.c | 539 ++++++++++++++++++++++++++++++++++++++ drivers/hid/hid-wiimote.h | 17 ++ 6 files changed, 619 insertions(+), 2 deletions(-) create mode 100644 drivers/hid/hid-wiimote-speaker.c diff --git a/drivers/hid/Kconfig b/drivers/hid/Kconfig index e8ef86c..0b4d7e2 100644 --- a/drivers/hid/Kconfig +++ b/drivers/hid/Kconfig @@ -715,6 +715,9 @@ config HID_WIIMOTE the Wii U Gamepad) might be supported in the future. But currently support is limited to Bluetooth based devices. + If Alsa sound support (CONFIG_SND) is enabled, this driver also provides + a sound card interface for Wii Remote speakers. + If unsure, say N. To compile this driver as a module, choose M here: the diff --git a/drivers/hid/Makefile b/drivers/hid/Makefile index 8b7106b..d149e10 100644 --- a/drivers/hid/Makefile +++ b/drivers/hid/Makefile @@ -32,6 +32,9 @@ hid-wiimote-y := hid-wiimote-core.o hid-wiimote-modules.o ifdef CONFIG_DEBUG_FS hid-wiimote-y += hid-wiimote-debug.o endif +ifdef CONFIG_SND + hid-wiimote-y += hid-wiimote-speaker.o +endif obj-$(CONFIG_HID_A4TECH) += hid-a4tech.o obj-$(CONFIG_HID_ACRUX) += hid-axff.o diff --git a/drivers/hid/hid-wiimote-core.c b/drivers/hid/hid-wiimote-core.c index 906c146..53f1e88 100644 --- a/drivers/hid/hid-wiimote-core.c +++ b/drivers/hid/hid-wiimote-core.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/mutex.h> #include <linux/spinlock.h> +#include <linux/uaccess.h> #include "hid-ids.h" #include "hid-wiimote.h" @@ -309,8 +310,8 @@ void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags) #define wiiproto_req_weeprom(wdata, os, buf, sz) \ wiiproto_req_wmem((wdata), true, (os), (buf), (sz)) -static void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, - __u32 offset, const __u8 *buf, __u8 size) +void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size) { __u8 cmd[22]; @@ -359,6 +360,52 @@ void wiiproto_req_rmem(struct wiimote_data *wdata, bool eeprom, __u32 offset, wiimote_queue(wdata, cmd, sizeof(cmd)); } +void wiiproto_req_speaker(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_SPEAKER; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +void wiiproto_req_mute(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_MUTE; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +/* must not be called from atomic contexts */ +int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len) +{ + unsigned long flags; + __u8 cmd[22]; + + if (len > 20) + len = 20; + + cmd[0] = WIIPROTO_REQ_AUDIO; + cmd[1] = (len & 0xff) << 3; + + if (copy_from_user(&cmd[2], buf, len)) + return -EFAULT; + + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + /* requries the cmd-mutex to be held */ int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size) @@ -570,6 +617,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN10] = (const __u8[]){ @@ -582,6 +630,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN20] = (const __u8[]){ @@ -595,6 +644,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_ACCEL, WIIMOD_IR, WIIMOD_BUILTIN_MP, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_BALANCE_BOARD] = (const __u8[]) { diff --git a/drivers/hid/hid-wiimote-modules.c b/drivers/hid/hid-wiimote-modules.c index 4ce0e02..7acf721 100644 --- a/drivers/hid/hid-wiimote-modules.c +++ b/drivers/hid/hid-wiimote-modules.c @@ -2072,6 +2072,11 @@ const struct wiimod_ops *wiimod_table[WIIMOD_NUM] = { [WIIMOD_IR] = &wiimod_ir, [WIIMOD_BUILTIN_MP] = &wiimod_builtin_mp, [WIIMOD_NO_MP] = &wiimod_no_mp, +#ifdef CONFIG_SND + [WIIMOD_SPEAKER] = &wiimod_speaker, +#else + [WIIMOD_SPEAKER] = &wiimod_dummy, +#endif }; const struct wiimod_ops *wiimod_ext_table[WIIMOTE_EXT_NUM] = { diff --git a/drivers/hid/hid-wiimote-speaker.c b/drivers/hid/hid-wiimote-speaker.c new file mode 100644 index 0000000..ecec5d9 --- /dev/null +++ b/drivers/hid/hid-wiimote-speaker.c @@ -0,0 +1,539 @@ +/* + * Driver for audio speakers of Nintendo Wii / Wii U peripherals + * Copyright (c) 2012-2013 David Herrmann <dh.herrmann(a)gmail.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. + */ + +/* + * Audio Speakers + * Some Wii peripherals provide an audio speaker that supports 8bit PCM and + * some other mostly unknown formats. Not all setup options are known, but we + * know how to setup an 8bit PCM or 4bit ADPCM stream and adjust volume. Data + * is sent as 20bytes chunks and needs to be streamed at a constant rate. + */ + +#include <linux/device.h> +#include <linux/hid.h> +#include <linux/mutex.h> +#include <linux/spinlock.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "hid-wiimote.h" + +struct wiimote_speaker { + struct snd_card *card; + unsigned int online : 1; + unsigned int mute : 1; + __u8 volume; +}; + +enum wiimod_speaker_mode { + WIIMOD_SPEAKER_MODE_PCM8, + WIIMOD_SPEAKER_MODE_ADPCM4, +}; + +static int wiimod_speaker_enable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (!speaker->online) { + speaker->online = 1; + wiiproto_req_speaker(wdata, true); + wiiproto_req_mute(wdata, speaker->mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static void wiimod_speaker_disable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->online) { + speaker->online = 0; + wiiproto_req_speaker(wdata, false); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_mute(struct wiimote_data *wdata, bool mute) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->mute != mute) { + speaker->mute = mute; + if (speaker->online) + wiiproto_req_mute(wdata, mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_volume(struct wiimote_data *wdata, __u8 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->volume != volume) { + speaker->volume = volume; + if (speaker->online) + wiiproto_req_wmem(wdata, false, 0xa20005, &volume, + sizeof(volume)); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +/* Change speaker configuration. \mode can be one of WIIMOD_SPEAKER_MODE_*, + * \rate is the PCM sample rate and \volume is the requested volume. */ +static int wiimod_speaker_setup(struct wiimote_data *wdata, + __u8 mode, __u16 rate, __s16 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + __u8 config[7], m, wmem; + __u16 r; + int ret; + + if (!rate) + return -EINVAL; + + switch (mode) { + case WIIMOD_SPEAKER_MODE_PCM8: + r = 12000000ULL / rate; + m = 0x40; + break; + case WIIMOD_SPEAKER_MODE_ADPCM4: + r = 6000000ULL / rate; + m = 0x00; + break; + default: + return -EINVAL; + } + + config[0] = 0x00; + config[1] = m; + config[2] = r & 0x00ff; + config[3] = (r & 0xff00) >> 8; + config[4] = volume & 0xff; + config[5] = 0x00; + config[6] = 0x00; + + wiimote_cmd_acquire_noint(wdata); + + /* mute speaker during setup and read/write volume field */ + spin_lock_irqsave(&wdata->state.lock, flags); + + wiiproto_req_mute(wdata, true); + if (volume < 0) + config[4] = speaker->volume; + else + speaker->volume = volume; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* power speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20009, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* prepare setup */ + wmem = 0x08; + ret = wiimote_cmd_write(wdata, 0xa20001, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* write configuration */ + ret = wiimote_cmd_write(wdata, 0xa20001, config, sizeof(config)); + if (ret) + goto out_unlock; + + /* enable speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20008, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* unmute speaker after setup if not muted */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + +out_unlock: + wiimote_cmd_release(wdata); + return ret; +} + +/* PCM layer */ + +static const struct snd_pcm_hardware wiimod_speaker_playback_hw = { + .info = SNDRV_PCM_INFO_NONINTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 1500, + .rate_max = 4000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 20, + .period_bytes_min = 20, + .period_bytes_max = 20, + .periods_min = 1, + .periods_max = 1, +}; + +static int wiimod_speaker_playback_open(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = wiimod_speaker_playback_hw; + runtime->private_data = wdata; + + return wiimod_speaker_enable(wdata); +} + +static int wiimod_speaker_playback_close(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + + wiimod_speaker_disable(wdata); + + return 0; +} + +static int wiimod_speaker_playback_hw_params(struct snd_pcm_substream *subs, + struct snd_pcm_hw_params *hw) +{ + /* TODO: anything to do here? */ + + return 0; +} + +static int wiimod_speaker_playback_hw_free(struct snd_pcm_substream *subs) +{ + return 0; +} + +static int wiimod_speaker_playback_prepare(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + size_t buf_size, period_size, periods; + int ret; + + /* TODO: If I have set the hw-params to a fixed 20 bytes buffer, do + * I actually need to do any computations here? */ + buf_size = frames_to_bytes(runtime, runtime->buffer_size); + period_size = frames_to_bytes(runtime, runtime->period_size); + periods = runtime->periods; + + switch (runtime->format) { + case SNDRV_PCM_FMTBIT_S8: + ret = wiimod_speaker_setup(wdata, WIIMOD_SPEAKER_MODE_PCM8, + runtime->rate, -1); + break; + default: + return -EINVAL; + } + + return ret; +} + +static int wiimod_speaker_playback_trigger(struct snd_pcm_substream *subs, + int cmd) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + /* unmute device on start if not muted by user-space */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + /* mute device when stopping transmission */ + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_req_mute(wdata, true); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + } + + return 0; +} + +static snd_pcm_uframes_t wiimod_speaker_playback_pointer(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + unsigned long pos; + + /* There is no way to read the current position. Even if there was a + * way to do that, it would be so slow that it would be useless. + * TODO: Should we interpolate via a timer? Is this function actually + * needed? When is it called? */ + pos = 0; + + pos %= 20; + + return bytes_to_frames(subs->runtime, pos); +} + +static int wiimod_speaker_playback_copy(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + void __user *buf, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: copy from "buf" "count" bytes to "protobuf + pos" + * With the given hw-params, can pos be non-zero? Can count != 20? + * If not, should I simply send a 20byte frame to the device here? */ + + return 0; +} + +static int wiimod_speaker_playback_silence(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: set "count" bytes of "protobuf + pos" to zero + * Can "pos" actually be non-zero with the given hw-params? If not, + * can I simply send a 20byte zeroed frame to the remote device? */ + + return 0; +} + +/* TODO: is there a reason this cannot be "const"? */ +static struct snd_pcm_ops wiimod_speaker_playback_ops = { + .open = wiimod_speaker_playback_open, + .close = wiimod_speaker_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = wiimod_speaker_playback_hw_params, + .hw_free = wiimod_speaker_playback_hw_free, + .prepare = wiimod_speaker_playback_prepare, + .trigger = wiimod_speaker_playback_trigger, + .pointer = wiimod_speaker_playback_pointer, + .copy = wiimod_speaker_playback_copy, + .silence = wiimod_speaker_playback_silence, +}; + +/* volume control */ + +static int wiimod_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = 0; + info->value.integer.max = 0xff; + + return 0; +} + +static int wiimod_speaker_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = speaker->volume; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + unsigned long value; + + value = val->value.integer.value[0]; + if (value > 0xff) + value = 0xff; + + wiimod_speaker_set_volume(wdata, value); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_volume = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = wiimod_speaker_volume_info, + .get = wiimod_speaker_volume_get, + .put = wiimod_speaker_volume_put, +}; + +/* mute control */ + +static int wiimod_speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = !!speaker->mute; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + + wiimod_speaker_set_mute(wdata, val->value.integer.value[0]); + + return 0; +} + +/* TODO: Is *_IFACE_CARD the right interface? */ +static const struct snd_kcontrol_new wiimod_speaker_mute = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = wiimod_speaker_mute_get, + .put = wiimod_speaker_mute_put, +}; + +/* initialization and setup */ + +static int wiimod_speaker_probe(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + int ret; + struct wiimote_speaker *speaker; + struct snd_card *card; + struct snd_kcontrol *kcontrol; + struct snd_pcm *pcm; + + /* create sound card device */ + ret = snd_card_create(-1, NULL, THIS_MODULE, + sizeof(struct wiimote_speaker), &card); + if (ret) + return ret; + speaker = card->private_data; + + wdata->speaker = speaker; + speaker->card = card; + speaker->mute = 1; + speaker->volume = 0xff; + strcpy(card->driver, "hid-wiimote"); + strcpy(card->shortname, "wiimote"); + strcpy(card->longname, "Nintendo Wii Remote speaker"); + + /* create volume control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_volume, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create mute control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_mute, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create PCM sub-device for playback */ + ret = snd_pcm_new(card, "Speaker", 0, 1, 0, &pcm); + if (ret) + goto err_free; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &wiimod_speaker_playback_ops); + pcm->private_data = wdata; + + /* register sound card */ + snd_card_set_dev(card, &wdata->hdev->dev); + ret = snd_card_register(card); + if (ret) + goto err_free; + + return 0; + +err_free: + snd_card_free(card); + wdata->speaker = NULL; + return ret; +} + +static void wiimod_speaker_remove(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + + if (!speaker) + return; + + snd_card_free_when_closed(speaker->card); + wdata->speaker = NULL; +} + +const struct wiimod_ops wiimod_speaker = { + .flags = 0, + .arg = 0, + .probe = wiimod_speaker_probe, + .remove = wiimod_speaker_remove, +}; diff --git a/drivers/hid/hid-wiimote.h b/drivers/hid/hid-wiimote.h index 30caef5..4907bc7 100644 --- a/drivers/hid/hid-wiimote.h +++ b/drivers/hid/hid-wiimote.h @@ -148,6 +148,7 @@ struct wiimote_data { struct timer_list timer; struct wiimote_ext *ext; struct wiimote_debug *debug; + struct wiimote_speaker *speaker; union { struct input_dev *input; @@ -172,6 +173,7 @@ enum wiimod_module { WIIMOD_IR, WIIMOD_BUILTIN_MP, WIIMOD_NO_MP, + WIIMOD_SPEAKER, WIIMOD_NUM, WIIMOD_NULL = WIIMOD_NUM, }; @@ -208,9 +210,12 @@ enum wiiproto_reqs { WIIPROTO_REQ_LED = 0x11, WIIPROTO_REQ_DRM = 0x12, WIIPROTO_REQ_IR1 = 0x13, + WIIPROTO_REQ_SPEAKER = 0x14, WIIPROTO_REQ_SREQ = 0x15, WIIPROTO_REQ_WMEM = 0x16, WIIPROTO_REQ_RMEM = 0x17, + WIIPROTO_REQ_AUDIO = 0x18, + WIIPROTO_REQ_MUTE = 0x19, WIIPROTO_REQ_IR2 = 0x1a, WIIPROTO_REQ_STATUS = 0x20, WIIPROTO_REQ_DATA = 0x21, @@ -241,6 +246,12 @@ extern void wiiproto_req_status(struct wiimote_data *wdata); extern void wiiproto_req_accel(struct wiimote_data *wdata, __u8 accel); extern void wiiproto_req_ir1(struct wiimote_data *wdata, __u8 flags); extern void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags); +extern void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size); +extern void wiiproto_req_speaker(struct wiimote_data *wdata, bool on); +extern void wiiproto_req_mute(struct wiimote_data *wdata, bool on); +extern int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len); extern int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size); extern ssize_t wiimote_cmd_read(struct wiimote_data *wdata, __u32 offset, @@ -283,6 +294,12 @@ static inline void wiidebug_deinit(void *u) { } #endif +#ifdef CONFIG_SND + +extern const struct wiimod_ops wiimod_speaker; + +#endif + /* requires the state.lock spinlock to be held */ static inline bool wiimote_cmd_pending(struct wiimote_data *wdata, int cmd, __u32 opt) -- 1.8.2.1

2 6

[alsa-devel] [PATCH] drm/i915: Add private api for power well usage -- alignment between graphic team and audio team
by Li, Jocelyn 03 May '13

03 May '13

Hi Guys, I apologize that I may have misunderstood on the agreement we have made on power well work. I assumed that audio team is expected to create private gfx driver API to control the power well between gfx and audio for internal releases. That's why I asked Xingchao work on that. So the correct implementation approach between gfx and audio should be, #1, Daniel will work on the existing private API in gfx driver for power well control that can be used by the audio driver to enable/disable the power well. #2, Once #1 is done, Audio team will integrate Daniel's patch into audio driver and add/improve on anything that is missing Please correct me if my understanding is not correct. Thanks, Jocelyn -----Original Message----- From: Daniel Vetter [mailto:daniel.vetter@ffwll.ch] Sent: Thursday, April 25, 2013 3:52 PM To: Wang, Xingchao Cc: Zanoni, Paulo R; ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; Wang Xingchao; Takashi Iwai; Barnes, Jesse; Wysocki, Rafael J Subject: Re: [PATCH] drm/i915: Add private api for power well usage On Thu, Apr 25, 2013 at 4:36 AM, Wang, Xingchao <xingchao.wang(a)intel.com> wrote: > Hi Daniel/Paulo, > > Any comments on this? > Add Jesse and Rafael in loop. Well I've figured we want to get the power well integration working before we waste time reviewing an interface which might or might not quite work. But if you want, I can drop two quick comments: - I think the functions needs some prefix like i915_ - Imo the power well should just be reference counted (the current interface name suggests it's generally useable, but actually can only cope with one request from the audio driver). - Is the spin_lock (irq-disabling even) really required for the audio driver to work? power well enabling can take up to 20ms, so this won't work as-is. Can't we just use a mutex? - I'd need to see the audio driver side of this to check how this works for coping with arbitrary module load ordering. Cheers, Daniel > > thanks > --xingchao > > >> -----Original Message----- >> From: Wang, Xingchao >> Sent: Wednesday, April 24, 2013 3:29 PM >> To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R; 'Takashi Iwai' >> Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; 'Wang Xingchao' >> Subject: RE: [PATCH] drm/i915: Add private api for power well usage >> >> Hi Guys, >> >> Sorry I should Add Takashi and alsa maillist in loop. >> >> I tested this patch against Daniel's "drm-intel-next-queued" Branch, >> on Haswell Mobile C3 stepping machines. >> >> Test steps: >> 1. only connect one monitor through HDMI cable 2. echo 1 > >> /sys/module/i915/parameters/disable_power_well ==> I think this could >> be set as 1 by default if applied my patch. >> 3. plug out HDMI cable >> 4. cat /proc/asound/card0/codec#0 >> >> At step 3, i915 will try to disable power well but rejected as >> display audio is using power well. >> Without the patch, display audio would crash at step 4. >> >> I did not list suspend/resume test here, as during suspend sequence, >> i915 will _ENABLE_ power well, that's a BUG? >> >> For me above test case is the only one to reproduce the issue, if you >> have more ideas and need more test, please let me know. >> >> I'm afraid there's some risk because the dependency between alsa >> driver and drm driver. For example, if i915 module was not loaded, >> display audio Will call >> request_power_well() and meet symbol error. Even if i915 module >> loaded after snd-hda-intel module, there's such risk too. Any comment on this? >> >> Another risk is the pointer "power_well_device", it should be set >> NULL when >> i915 module unloaded, I added the caller but have no idea where to >> call it, if anyone has suggestion, that would be fine. >> >> thanks >> --xingchao >> >> >> > -----Original Message----- >> > From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] >> > Sent: Thursday, April 24, 2014 10:20 PM >> > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R >> > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> > Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; Wang, Xingchao; Wang >> > Xingchao >> > Subject: [PATCH] drm/i915: Add private api for power well usage >> > >> > Display audio depends on power well in graphic side, it should >> > request power well before use it and release power well after use. >> > I915 will not shutdown power well if it detects audio is using. >> > This patch protects display audio crash for Intel Haswell mobile >> > C3 stepping board. >> > >> > Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> >> > --- >> > drivers/gpu/drm/i915/intel_pm.c | 73 >> > +++++++++++++++++++++++++++++++++++---- >> > include/drm/audio_drm.h | 39 +++++++++++++++++++++ >> > sound/pci/hda/hda_intel.c | 8 +++++ >> > 3 files changed, 113 insertions(+), 7 deletions(-) create mode >> > 100644 include/drm/audio_drm.h >> > >> > diff --git a/drivers/gpu/drm/i915/intel_pm.c >> > b/drivers/gpu/drm/i915/intel_pm.c index 2557926..9983f56 100644 >> > --- a/drivers/gpu/drm/i915/intel_pm.c >> > +++ b/drivers/gpu/drm/i915/intel_pm.c >> > @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct >> > drm_device >> > *dev) >> > return true; >> > } >> > >> > -void intel_set_power_well(struct drm_device *dev, bool enable) >> > +void set_power_well(struct drm_device *dev, bool enable) >> > { >> > struct drm_i915_private *dev_priv = dev->dev_private; >> > bool is_enabled, enable_requested; >> > uint32_t tmp; >> > >> > - if (!HAS_POWER_WELL(dev)) >> > - return; >> > - >> > - if (!i915_disable_power_well && !enable) >> > - return; >> > - >> > tmp = I915_READ(HSW_PWR_WELL_DRIVER); >> > is_enabled = tmp & HSW_PWR_WELL_STATE; >> > enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 >> > +4326,71 @@ void intel_set_power_well(struct drm_device *dev, bool >> > +enable) >> > } >> > } >> > >> > +/* Global drm_device for display audio drvier usage */ static >> > +struct drm_device *power_well_device; static bool >> > +audio_power_well_using, i915_power_well_using; >> > +/* Lock protecting power well setting */ >> > +DEFINE_SPINLOCK(powerwell_lock); >> > + >> > +void request_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio requesting power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, true); >> > + audio_power_well_using = true; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(request_power_well); >> > + >> > +void release_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio releasing power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, false); >> > + audio_power_well_using = false; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(release_power_well); >> > + >> > +/* TODO: Call this when i915 module unload */ void >> > +remove_power_well(void) { >> > + spin_lock_irq(&powerwell_lock); >> > + audio_power_well_using = false; >> > + i915_power_well_using = false; >> > + power_well_device = NULL; >> > + spin_unlock_irq(&powerwell_lock); } >> > + >> > +void intel_set_power_well(struct drm_device *dev, bool enable) { >> > + if (!HAS_POWER_WELL(dev)) >> > + return; >> > + >> > + spin_lock_irq(&powerwell_lock); >> > + power_well_device = dev; >> > + i915_power_well_using = enable; >> > + if (!enable && audio_power_well_using) >> > + goto out; >> > + >> > + if (!i915_disable_power_well && !enable) >> > + goto out; >> > + set_power_well(dev, enable); >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > + >> > /* >> > * Starting with Haswell, we have a "Power Down Well" that can be >> > turned >> off >> > * when not needed anymore. We have 4 registers that can request >> > the power well diff --git a/include/drm/audio_drm.h >> > b/include/drm/audio_drm.h new file mode 100644 index >> > 0000000..4412b36 >> > --- /dev/null >> > +++ b/include/drm/audio_drm.h >> > @@ -0,0 +1,39 @@ >> > >> +/************************************************************** >> > ******** >> > +**** >> > + * >> > + * Copyright 2013 Intel Inc. >> > + * All Rights Reserved. >> > + * >> > + * Permission is hereby granted, free of charge, to any person >> > +obtaining a >> > + * copy of this software and associated documentation files (the >> > + * "Software"), to deal in the Software without restriction, >> > +including >> > + * without limitation the rights to use, copy, modify, merge, >> > +publish, >> > + * distribute, sub license, and/or sell copies of the Software, >> > +and to >> > + * permit persons to whom the Software is furnished to do so, >> > +subject to >> > + * the following conditions: >> > + * >> > + * The above copyright notice and this permission notice >> > +(including the >> > + * next paragraph) shall be included in all copies or substantial >> > +portions >> > + * of the Software. >> > + * >> > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY >> KIND, >> > +EXPRESS OR >> > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF >> > +MERCHANTABILITY, >> > + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO >> > EVENT >> > +SHALL >> > + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE >> > FOR >> > +ANY CLAIM, >> > + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, >> > TORT >> > +OR >> > + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE >> > SOFTWARE >> > +OR THE >> > + * USE OR OTHER DEALINGS IN THE SOFTWARE. >> > + * >> > + * >> > + >> > >> +*************************************************************** >> > ******** >> > +***/ >> > +/* >> > + * Authors: >> > + * Wang Xingchao <xingchao.wang(a)intel.com> */ >> > + >> > +#ifndef _AUDIO_DRM_H_ >> > +#define _AUDIO_DRM_H_ >> > + >> > +extern void request_power_well(void); extern void >> > +release_power_well(void); >> > + >> > +#endif /* _AUDIO_DRM_H_ */ >> > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c >> > index 418bfc0..59a85de 100644 >> > --- a/sound/pci/hda/hda_intel.c >> > +++ b/sound/pci/hda/hda_intel.c >> > @@ -50,6 +50,7 @@ >> > #include <linux/clocksource.h> >> > #include <linux/time.h> >> > #include <linux/completion.h> >> > +#include <drm/audio_drm.h> >> > >> > #ifdef CONFIG_X86 >> > /* for snoop control */ >> > @@ -2869,6 +2870,7 @@ static int azx_suspend(struct device *dev) >> > pci_disable_device(pci); >> > pci_save_state(pci); >> > pci_set_power_state(pci, PCI_D3hot); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2881,6 +2883,7 @@ static int azx_resume(struct device *dev) >> > if (chip->disabled) >> > return 0; >> > >> > + request_power_well(); >> > pci_set_power_state(pci, PCI_D0); >> > pci_restore_state(pci); >> > if (pci_enable_device(pci) < 0) { @@ -2913,6 +2916,7 @@ static >> > int azx_runtime_suspend(struct device >> > *dev) >> > >> > azx_stop_chip(chip); >> > azx_clear_irq_pending(chip); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2921,6 +2925,7 @@ static int azx_runtime_resume(struct device *dev) >> > struct snd_card *card = dev_get_drvdata(dev); >> > struct azx *chip = card->private_data; >> > >> > + request_power_well(); >> > azx_init_pci(chip); >> > azx_init_chip(chip, 1); >> > return 0; >> > @@ -3671,6 +3676,8 @@ static int azx_probe(struct pci_dev *pci, >> > return -ENOENT; >> > } >> > >> > + request_power_well(); >> > + >> > err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); >> > if (err < 0) { >> > snd_printk(KERN_ERR "hda-intel: Error creating >> > card!\n"); @@ >> > -3806,6 +3813,7 @@ static void azx_remove(struct pci_dev *pci) >> > if (card) >> > snd_card_free(card); >> > pci_set_drvdata(pci, NULL); >> > + release_power_well(); >> > } >> > >> > /* PCI IDs */ >> > -- >> > 1.7.9.5 > -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

9 23

[alsa-devel] [PATCH -next 0/3] mop500_ab8500 fixes
by Fabio Baltieri 03 May '13

03 May '13

Hi, these are small fixes to the mop500_ab8500 asoc driver, developed on top of today's ASoC-next. The first patch is actually non ab8500 specific, but as ab8500 seems to be the only codec to use the SND_SOC_DAPM_CLOCK_SUPPLY widget I'm posting it here anyway. The other two patches sort out a mixup of card and codec controls in mop500_ab8500. Thanks, Fabio Fabio Baltieri (3): ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare ASoC: ux500: move clock controls to ab8500-codec ASoC: ux500: register controls to card instead of codec sound/soc/codecs/ab8500-codec.c | 10 ++++++++++ sound/soc/soc-dapm.c | 4 ++-- sound/soc/ux500/mop500_ab8500.c | 20 +++++--------------- 3 files changed, 17 insertions(+), 17 deletions(-) -- 1.8.2

3 11

[alsa-devel] [PATCH] ASoC: Added the condition for the channels_min of cpu_dai
by JongHo Kim 02 May '13

02 May '13

Only playback device, the pcm substream for capture does not need to be generated. If codec dai driver has both playback.channels_min and capture.channels_min, and cpu dai driver has only playback.channels_min, soc_new_pcm() will make the pcm substream for capture although DataIN is not connected. So added the condition for the channels_min of cpu_dai in soc_new_pcm(). Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6..c405e39 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2018,9 +2018,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } -- 1.8.1.1

2 4

[alsa-devel] [PATCH v2 1/4] ASoC: imx-sgtl5000: Do not enter the error path on success
by Fabio Estevam 02 May '13

02 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Return on success instead of entering the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Keep of_node_put in the success path. sound/soc/fsl/imx-sgtl5000.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78..5a6aaa3 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -174,6 +174,11 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + clk_fail: clk_put(data->codec_clk); fail: -- 1.7.9.5

3 12

[alsa-devel] [PATCH] MAINTAINERS: Add linux-samsung-soc list to all related entries
by Tomasz Figa 01 May '13

01 May '13

Several entries in MAINTAINERS file related to Samsung SoCs do not point to linux-samsung-soc mailing list, which is supposed to collect all Samsung SoC-related threads, to ease following of Samsung SoC-related work. This leads to a problem with people skipping this mailing list in their posts, even though they are related to Samsung SoCs. This patch adds pointers to linux-samsung-soc mailing list to affected entries of MAINTAINERS file. Signed-off-by: Tomasz Figa <tomasz.figa(a)gmail.com> --- MAINTAINERS | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 6c0d68b..07cb8da 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1139,6 +1139,7 @@ F: arch/arm/mach-exynos*/ ARM/SAMSUNG MOBILE MACHINE SUPPORT M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/mach-s5pv210/mach-aquila.c F: arch/arm/mach-s5pv210/mach-goni.c @@ -1150,6 +1151,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Kamil Debski <k.debski(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-g2d/ @@ -1158,6 +1160,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/include/plat/*fimc* F: drivers/media/platform/s5p-fimc/ @@ -1168,6 +1171,7 @@ M: Kamil Debski <k.debski(a)samsung.com> M: Jeongtae Park <jtp.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/s5p-dev-mfc.c F: drivers/media/platform/s5p-mfc/ @@ -1177,6 +1181,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Tomasz Stanislawski <t.stanislaws(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-tv/ @@ -2679,6 +2684,7 @@ M: Joonyoung Shim <jy0922.shim(a)samsung.com> M: Seung-Woo Kim <sw0312.kim(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: dri-devel(a)lists.freedesktop.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/daeinki/drm-exynos.git S: Supported F: drivers/gpu/drm/exynos @@ -3142,6 +3148,7 @@ F: Documentation/extcon/ EXYNOS DP DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_dp* F: include/video/exynos_dp* @@ -3151,6 +3158,7 @@ M: Inki Dae <inki.dae(a)samsung.com> M: Donghwa Lee <dh09.lee(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_mipi* F: include/video/exynos_mipi* @@ -6892,12 +6900,14 @@ F: drivers/platform/x86/samsung-laptop.c SAMSUNG AUDIO (ASoC) DRIVERS M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung SAMSUNG FRAMEBUFFER DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/s3c-fb.c @@ -7087,6 +7097,7 @@ F: drivers/mmc/host/sdhci-pltfm.[ch] SECURE DIGITAL HOST CONTROLLER INTERFACE (SDHCI) SAMSUNG DRIVER M: Ben Dooks <ben-linux(a)fluff.org> L: linux-mmc(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/mmc/host/sdhci-s3c.c -- 1.8.2.1

4 3

[alsa-devel] [PATCH] ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format.
by Marek Belisko 30 Apr '13

30 Apr '13

When McASP is bit clock and frame clock master enable pin output for rx clocks. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d54e0d9..2953c55 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -505,7 +505,10 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, ACLKX | AFSX); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | ACLKR); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: Add devicetree support for spdif dummy codecs.
by Marek Belisko 30 Apr '13

30 Apr '13

changes from v1 (for patch 2 and 3): - add prefix linux for compatible property (for both tx and rx) - rename spdif_transceiver.c to spdif_transmitter.c - add bindings documentation Marek Belisko (3): ASoC: spdif_transceiver: Change driver filename to spdif_transmitter.c. ASoC: spdif_transmitter: Add DT support. ASoC: spdif_receiver: Add DT support. .../devicetree/bindings/sound/spdif-receiver.txt | 10 ++++++++++ .../bindings/sound/spdif-transmitter.txt | 10 ++++++++++ sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/spdif_receiver.c | 10 ++++++++++ .../{spdif_transciever.c => spdif_transmitter.c} | 10 ++++++++++ 5 files changed, 41 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) -- 1.7.9.5

4 7

Re: [alsa-devel] linux-next: Tree for Apr 30 (sound)
by Randy Dunlap 30 Apr '13

30 Apr '13

On 04/29/13 23:57, Stephen Rothwell wrote: > Hi all, > > Please do not add any v3.11 destined work to your linux-next included > branches until after v3.10-rc1 is released. > > Changes since 20130429: > when CONFIG_PROC_FS is not enabled: sound/core/memalloc.c: In function 'snd_mem_exit': sound/core/memalloc.c:525:20: error: 'SND_MEM_PROC_FILE' undeclared (first use in this function) -- ~Randy

1 0

[alsa-devel] [patch] ASoC: wm8994: missing break in wm8994_aif3_hw_params()
by Dan Carpenter 30 Apr '13

30 Apr '13

The missing break here means that we always return early and the function is a no-op. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 14094f5..1eb152c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2882,6 +2882,7 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, default: return 0; } + break; default: return 0; }

3 3

Re: [alsa-devel] what is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE
by Clemens Ladisch 30 Apr '13

30 Apr '13

(please do not drop the mailing list from replies.) Zhang wei wrote: > I do not really understand. For example,an codec is wm8985 ... The SNDRV_PCM_* symbols specify the format in which samples are stored in memory. I do not know how ASoC maps these symbols to on-the-wire formats. >>Most devices that support 24 bits use SNDRV_PCM_FORMAT_S32_LE, which >>stores the sample's bits in the upper bits of a 32-bit word. > > But how to distinguish the SNDRV_PCM_FORMAT_S24_LE and the REAL > SNDRV_PCM_FORMAT_S32_LE? If a 24-bit sample with the value 0x123456 is stored in memory, it looks like this: SNDRV_PCM_FORMAT_S24_LE: 56 34 12 00 SNDRV_PCM_FORMAT_S32_LE: 00 56 34 12 Regards, Clemens

3 2

[alsa-devel] [PATCH, for-next] ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
by Eldad Zack 30 Apr '13

30 Apr '13

Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively (neglecting the index): be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Daniel Mack confirmed that the driver works on little endian CPUs, leading to the conclusion that the device-side structure is actually little endian. This changes the code to use le16_to_cpu(). Caught by sparse. Acked-by: Daniel Mack <zonque(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..4b3fb91 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,13 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), + le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

3 2

[alsa-devel] ASoC : playback and capture counts in soc_new_pcm
by JongHo Kim 29 Apr '13

29 Apr '13

Hi I am developing the ALSA ASoC part. I has some problem. Now my situation must only use the cpu_dai(AP I2S IP) to half duplex. So, I want to allocate either a capture substream or a playback substream. But ASoC(soc-pcm.c) referred to the playback(capture)_channel_min of codec_dai at the calling snd_pcm_new in soc_new_pcm. I do not want to modify the codec_dai codes. As a result, hope to see below /dev/snd/pcmC0D0p /dev/snd/pcmC0D1c attach the ASCII schematic diagram. Slaves Master +--------------+ +-----------------+ |I2S.0 data_out|---------------|data_in(AIF1) | |I2S.0 bclk |-------+-------|bclk | |I2S.0 lrclk |-------|--+----|lrclk | +--------------+ | | | | +--------------+ | | | AUDIO CODEC | |I2S.1 bclk |-------+ | | | |I2S.1 lrclk |----------+ | | |I2S.1 data_in |---------------|data_out(AIF1) | +--------------+ +-----------------+ attach the i2s and dai_link codes. ----------------------------------------------------------------------------- static struct snd_soc_dai_driver board_i2s_dai_hf_playback_template = { .probe = board_i2s_probe, .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_driver board_i2s_dai_hf_capture_template = { .probe = board_i2s_probe, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_link board_dai[] = { { .name = "audcodec_l0", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.0", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, { .name = "audcodec_l1", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.1", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, }; ----------------------------------------------------------------------------- So, I suggest the next diff. --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2014,9 +2014,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) capture = 1; } Thanks.

2 2

[alsa-devel] ALSA SOC Driver (mxs-wm8753)
by John Graham 29 Apr '13

29 Apr '13

Hi there, I'm completely new to both kernel/ALSA driver programming (at least at a serious level) and was looking for some help on a project I'm working on - I'm part way there but I've kind of hit a brick wall and don't know where to turn. I'm trying to get a codec working on a new platform (new for the codec - the platform itself is not new). Here are the details: The audio codec is the Wolfson WM8753, which already has a codec driver in the kernel mainline tree. We also have a vendor-supplied (but GPL) patch to make the wm8753 work on a CC9M2443 (based on the Samsung S3C2443). This includes the glue code ("platform" code?) in sound/soc/s3c24xx/cc9m2443js_wm8753.c. We have a product with this combination fully working. The platform is the Karo TX28 (based on the Freescale i.MX28), and there are already drivers in the kernel tree to make the TX28 work with an SGTL5000 codec in sound/soc/mxs/mxs-sgtl5000.c. We also have a product with this combination fully working. Now we want to move the WM8753 over from the CC9M2443 to work on the TX28. To simplify, what I *have* is: CC9M2443 <----> WM8753 TX28 <----> SGTL5000 and what I *want* is: /-> WM8753 TX28 <-/ What I was hoping I could do was just "mash" the existing drivers together - i.e. copy sound/soc/mxs/mxs-sgtl5000.c into sound/soc/mxs/mxs-wm8753.c, change the parts in it that were specific for the SGTL5000 to the parts that drive the wm8753 in sound/soc/s3c24xx/cc9m2443js_wm8753.c, tell the TX28 to probe for it and that it would "magically work" (hey, crazier things have happened...). Obviously, it does not work, but I've got it a little way down the line. The WM8752 is probed for by the TX28 and recognized - the I2C control works, and you can change the mixer settings with amixer. When I try to record (using arecord) I can see with an oscilloscope the I2S clock lines look correct (in that they look the same as when the CC9M2443 talks to the WM8753 at the same audio rate). The problem now is that the I2S data lines don't look right - the line from codec to TX28 looks like a (more-or-less) square wave with the same period as the I2S clock, but only going from ~3.1 - ~3.5 V, as opposed to the 0 - ~3.3 V I expect. Furthermore, it's a pretty regular square wave - it doesn't have all the breaks you see on a working I2S data line. I know I probably haven't provided the right detail to get to the bottom of this problem, but part of my problem is that I'm so new to this that I'm not entirely sure where the problem might lie, or what extra detail to include to ask for more help. Is anyone able to point me in the right direction/suggest some ideas as to what might be wrong/anything else at all? Any help would be much appreciated - many thanks, John Graham

2 1

[alsa-devel] Configure a CODEC which has I2C connected to CPU, but no PCM bus connection
by Craig McQueen 29 Apr '13

29 Apr '13

I have an embedded device with somewhat unusual configuration: there is a second audio CODEC (Wolfson WM8940) which is connected to the PCM interface of another device (a radio module, which is acting as PCM bus master), so that radio device can do analog audio I/O. So there is no CPU<-->CODEC PCM link, but the CODEC _is_ connected to the CPU's I2C, and the CPU needs to configure the CODEC registers to interface to the radio module. See the attached diagram which shows a simplified view of the configuration. This configuration came about because both the CPU and the radio module can _only_ operate as PCM bus masters. I am not sure how I could fit this into the ALSA ASoC architecture. Is the ALSA ASoC driver architecture able to accommodate this scenario? Can I use the WM8940 driver to configure the second CODEC through I2C? Could I simply set up a machine driver that defines a DAI including a pointer to the WM8940 CODEC DAI, but NULL pointer for CPU DAI? Can I set it up so the Wolfson CODEC gains could appear as ALSA mixer widgets? Regards, Craig McQueen

2 1

[alsa-devel] [PATCH] ASoC: wm8962: Convert to devm_input_allocate_device()
by Mark Brown 29 Apr '13

29 Apr '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 50b0983..1cee9b3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3189,7 +3189,7 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int ret; - wm8962->beep = input_allocate_device(); + wm8962->beep = devm_input_allocate_device(codec->dev); if (!wm8962->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -3210,7 +3210,6 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) ret = input_register_device(wm8962->beep); if (ret != 0) { - input_free_device(wm8962->beep); wm8962->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -3227,7 +3226,6 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(wm8962->beep); cancel_work_sync(&wm8962->beep_work); wm8962->beep = NULL; -- 1.7.10.4

4 11

[alsa-devel] ALSA: usb-audio: caiaq: endianness bug
by Eldad Zack 29 Apr '13

29 Apr '13

Hi Daniel, I think I found a endian usage issue in the caiaq driver. I have a patch to fix it (below), but it assumes that the driver works currently on little endian CPUs - I don't have a device to test it with. Cheers, Eldad -- >8 -- Subject: [PATCH] ALSA: usb-audio: caiaq: fix endianness bug Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively: be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Assuming the device-side structure is actually little endian, change the code to use le16_to_cpu(). Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..2f4ae4d 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,12 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

2 2

Re: [alsa-devel] linux-next: Tree for Apr 29 (sound/soc)
by Randy Dunlap 29 Apr '13

29 Apr '13

On 04/29/13 02:17, Stephen Rothwell wrote: > Hi all, > > Changes since 20130426: > on i386: CONFIG_I2C=m sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x1ebe4): undefined reference to `regmap_init_i2c' make[1]: *** [vmlinux] Error 1 -- ~Randy

1 0

[alsa-devel] [PATCH 0/3] snd-firewire-lib: add handling CMP output connection
by Takashi Sakamoto 29 Apr '13

29 Apr '13

Current implementation of snd-firewire-lib can handle only CMP input connection. This series of patch enable it to handle both CMP output and input connection according to IEC 61883-1. Note that this series of patch still supports only a point-to-point connection without overlaying. The overlaying and any broadcast connection are not supported. Each connections gain its own isochronous resource. Takashi Sakamoto (3): rename macros, variables and functions related to CMP plug Add "direction" member to cmp_connection structure Add handling CMP output connection sound/firewire/cmp.c | 192 +++++++++++++++++++++++++++++++++------------ sound/firewire/cmp.h | 13 ++- sound/firewire/speakers.c | 2 +- 3 files changed, 152 insertions(+), 55 deletions(-) -- 1.7.10.4

2 12

[alsa-devel] Missing patches for 3.10 merge?
by Takashi Iwai 29 Apr '13

29 Apr '13

Hi, if anyone posted fix patches that should be applied to 3.10, but still unapplied, please let me know ASAP. I got a complete home (and system) FS crash on Friday (due to too much experiment with btrfs) and might have lost some incoming mails. thanks, Takashi

2 2

[alsa-devel] [PATCH] ALSA: compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..5389b9a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,14 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transistion to after stop, so if user + * wants to read data after stop we allow that + */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING || + stream->runtime->state != SNDRV_PCM_STATE_DRAINING || + stream->runtime->state != SNDRV_PCM_STATE_PAUSED || + stream->runtime->state != SNDRV_PCM_STATE_SETUP) { retval = -EBADFD; goto out; } -- 1.7.0.4

4 8

[alsa-devel] [PATCH] compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 12 ++++++++++-- 1 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..7b774b7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,16 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transition to after stop, so if user + * wants to read data after stop we allow that + */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_XRUN: + case SNDRV_PCM_STATE_SUSPENDED: + case SNDRV_PCM_STATE_DISCONNECTED: retval = -EBADFD; goto out; } -- 1.7.0.4

2 1