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[alsa-devel] [PATCH - Set EAPD for Realtek ALC665 (Vendor Id: 0x10eSet EAPD for Realtek ALC665 (Vendor Id: 0x10ec0665).
by Andres Mejia 20 Mar '11

20 Mar '11

Signed-off-by: Andres Mejia <mcitadel(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50855cf..be910fa6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1264,6 +1264,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 1); @@ -4236,6 +4237,7 @@ static void alc_power_eapd(struct hda_codec *codec) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 0); -- 1.7.4.1

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[alsa-devel] snd_pcm_writei returns -EAGAIN after poll
by Ivan Sorokin 20 Mar '11

20 Mar '11

Hello! I use slightly modified test/pcm.c in write_and_poll mode. My program looks like this: snd_pcm_open with SND_PCM_NONBLOCK snd_pcm_set_params infinite loop poll for snd_pcm_poll_descriptors snd_pcm_poll_descriptors_revents if POLLOUT flag is set snd_pcm_writei Everything works, but snd_pcm_writei sometimes returns -EAGAIN. The probability of when snd_pcm_writei returns -EAGAIN depends on buffer size that I pass to it. ExperimentallyI found out that when I call snd_pcm_writei with 940 bytes buffers (maybe something related to period size?) it never returns -EAGAIN. Why poll returns when there is no place for new data? Is it ok to ignore EAGAIN and just start another poll? Is there any recommended buffer size for snd_pcm_writei? My test program: https://github.com/sorokin/asoundpp/blob/master/sine_async.cpp (94 LOC)

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[alsa-devel] ALSA/pulseaudio problem workaround
by Craig Bourne 19 Mar '11

19 Mar '11

Here is a workaround for the problem I described in my email to you yesterday: Background My past experience with ALSA, JACK and many of the Linux audio tools that depend on them is that often some few applications in the tool-chain of a complex task will not work properly (or at all) unless invoked with *root* privileges. One is now frustrated in getting such things to work Fedora 14 as they have for the past decade or so in that the most basic tools (e.g., alsactl, QjackCtl) no longer work for root. Indeed, audio also does not work for unprivileged users on Fedora 14 either (though this evidently fails at a different level and will be taken up in a subsequent communication). Symptoms alsactl fails carping about pulseaudio, audacity can be started on its own but no longer wants to cooperate with JACK and ALSA resulting in horribly distorted sound from something horked together in the background as a "favor" to the user, QjackCtl runs, produces messages, and blinks its cute little meter display but evidently does this off on the sidelines reporting 0 XRUNS even as every little jot and tiddle of computation produces a surprising audio result. Strategy Based on my observation that PULSEAUDIO is evidently being allowed to interpose itself between the user (even the superuser) and ALSA, and since this user has been kept largely ignorant of the workings of all this and does not know the secret handshake, I will attempt to at least minimize the damage to audio and regain some measure of control while not attempting to disable the PULSEAUDIO/ALSA cabal. Workaround Install the following if they are not already on your system: - avahi - padevchooser - pulseaudio-libs-zeroconf - pulseaudio-module-zeroconf Enable and start the Avahi Daemon Create these two shell scripts (that will be used by QjackCtl), make them executable, and place them in a directory where they will stay to be invoked each time QjackCtl is started: PAjack_startup #!/bin/bash # Name: PAjack_startup pulseaudio -D --system #perhaps more fastidious than the above line is #pulseaudio -D --system --disallow-exit --disallow-module-loading padevchooser& PAjack_shutdown #!/bin/bash #Name: PAjack_shutdown pulseaudio --kill killall padevchooser Invoke QjackCtl click Setup select the Options tab tick the box "Execute script after Startup" and enter ~/bin/PAjack_startup [Note: using, of course the path to your script] tick the box "Execute script on Shutdown" and enter ~/bin/PAjack_shutdown [Note: as before, using your path to the file] Quit QjackCtl -- your scripts will now be invoked the next time it is used. A partial workaround for the unprivileged user, at least to the point of getting QjackCtl and audacious working places a slightly modified PAjack_startup script in a place where QjackCtl can find it when it starts with the changes used above for root (e.g., insure that both acounts have a subdirectory named "bin" in their home directory and have QjackCtl look for ~/bin/PAjack_startup when it is invoked). Here is the script for unprivileged users: #!/bin/bash #Name: PAjack_startup pulseaudio -D padevchooser& Limitations This is not a general solution to the ALSA/PULSEAUDIO problems I am experiencing but it is a workaround that at least allows me to have some minimal functionality back from basic audio tools that I have come to rely on in my work as a composer. -- Comradely, Craig Bourne

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[alsa-devel] syslog requested reporting this to ALSA developers (ubuntu 10.04 inside virtualbox)
by Jason E. Aten 19 Mar '11

19 Mar '11

from my syslog, it requests a report to ALSA developers. Let me know if I can provide further assistance. -Jason $ tail /var/log/syslog Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:58:29 virtub gdm-session-worker[1030]: GLib-GObject-CRITICAL: g_value_get_boolean: assertion `G_VALUE_HOLDS_BOOLEAN (value)' failed Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1256 of process 1256 (n/a) owned by '1000' high priority at nice level -11. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 4 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1257 of process 1256 (n/a) owned by '1000' RT at priority 5. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 5 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1258 of process 1256 (n/a) owned by '1000' RT at priority 5. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 6 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:59:40 virtub AptDaemon: INFO: Initializing daemon root@virtub:/var/log# uname -a Linux virtub 2.6.32-29-generic #58-Ubuntu SMP Fri Feb 11 20:52:10 UTC 2011 x86_64 GNU/Linux root@virtub:/var/log# dpkg -s pulseaudio Package: pulseaudio Status: install ok installed Priority: optional Section: sound Installed-Size: 4540 Maintainer: Ubuntu Core Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Architecture: amd64 Version: 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14 Replaces: pulseaudio-module-hal, pulseaudio-module-rygel-media-server (<< 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14), pulseaudio-module-udev (<< 1:0.9.22~0.9.21+stable-qu\ eue-32-g8478-0ubuntu14) Provides: pulseaudio-module-hal, pulseaudio-module-rygel-media-server, pulseaudio-module-udev Depends: libasound2 (>> 1.0.22), libc6 (>= 2.9), libdbus-1-3 (>= 1.0.2), libice6 (>= 1:1.0.0), libltdl7 (>= 2.2.6b), libpulse0 (= 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu1\ 4), libsamplerate0, libsm6, libsndfile1 (>= 1.0.20), libspeexdsp1 (>= 1.2~beta3.2-1), libtdb1 (>= 1.1.2~git20080520), libudev0 (>= 147), libx11-6 (>= 0), libxtst6, adduser, lsb\ -base (>= 3), consolekit, libasound2-plugins, pulseaudio-utils Recommends: pulseaudio-module-x11, gstreamer0.10-pulseaudio, pulseaudio-esound-compat, rtkit Suggests: pavumeter, paman, paprefs Conflicts: libltdl3 (<< 1.5.24-1), pulseaudio-module-rygel-media-server (<< 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14), pulseaudio-module-udev (<< 1:0.9.22~0.9.21+stable-\ queue-32-g8478-0ubuntu14) Conffiles: /etc/default/pulseaudio 5ac1e3004508adbb816e9b1af352617e /etc/init.d/pulseaudio bd6ccc065d494d8baac1d823521833b4 /etc/pulse/default.pa d2d623c97aef9f81f5b55f2d7e736c87 /etc/pulse/system.pa 182fbc3ba3449027c4d1e39a150c8396 /etc/pulse/daemon.conf ed3746b9cd3ba486184128fd611b73c9 /etc/dbus-1/system.d/pulseaudio-system.conf 69d56dc5342bf589a2e199541709a452 /etc/xdg/autostart/pulseaudio.desktop 1310d377ad0d534f699e864037f66bd2 Description: PulseAudio sound server PulseAudio, previously known as Polypaudio, is a sound server for POSIX and WIN32 systems. It is a drop in replacement for the ESD sound server with much better latency, mixing/re-sampling quality and overall architecture. . These are some of PulseAudio's features: . * High quality software mixing of multiple audio streams with support for more than one sink/source. May be used to combine multiple sound cards into one (with sample rate adjustment). . * Wide range of supported client libraries. ESD, ALSA, oss, libao and GStreamer client applications are supported as-is. Native PulseAudio plug-ins are also available for xmms and mplayer. . * Good low latency behaviour and very accurate latency measurement for playback and recording. Ability to fully synchronize multiple playback streams. . * Network transparency, allowing an application to play back or record audio on a different machine than the one it is running on. . * Extensible plug-in architecture with plug-ins for jackd, multicast-rtp lirc and avahi, just to name a few. . This package contains the daemon and basic module set. Homepage: http://www.pulseaudio.org Original-Maintainer: Pulseaudio maintenance team < pkg-pulseaudio-devel(a)lists.alioth.debian.org> root@virtub:/var/log#

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Re: [alsa-devel] Microphone not working on PCI ID 8086:3b56
by Jeff Chua 18 Mar '11

18 Mar '11

On Tue, Sep 28, 2010 at 9:42 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > Enable all CONFIG_SND_HDA_*. Done that. Still no mic. On Tue, Sep 28, 2010 at 9:42 PM, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > It's easier to enable all of them. They are small modules. Tried just about everything, still no mic. Sounds works in all cases. I found that with SND_HDA_CODEC_INTELHDMI selected, SND_DYNAMIC_MINORS will be enabled by default which won't work in my case as I'm not using udev. Here's a patch to take away the default so non-udev will still works. Do I really need udev to get mic to work? Thanks, Jeff --- lx/sound/pci/hda/Kconfig.org 2010-09-14 16:34:52.000000000 +0800 +++ lx/sound/pci/hda/Kconfig 2010-09-14 16:35:09.000000000 +0800 @@ -145,7 +145,6 @@ config SND_HDA_CODEC_INTELHDMI bool "Build INTEL HDMI HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include INTEL HDMI HD-audio codec support in

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[alsa-devel] [PATCHv4] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> Acked-by: Liam Girdwood <lrg(a)ti.com> --- src/ucm/main.c | 21 ++++++++++++++------- 1 files changed, 14 insertions(+), 7 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..88f8bf6 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1059,24 +1059,31 @@ static int get_value3(const char **value, * \param uc_mgr Use case manager * \param identifier Value identifier (string) * \param value Returned value string - * \param modifier modifier name (string) + * \param item Modifier or Device name (string) * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >= 0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

2 2

[alsa-devel] [PATCHv3] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> Acked-by: Liam Girdwood <lrg(a)ti.com> --- src/ucm/main.c | 21 ++++++++++++++------- 1 files changed, 14 insertions(+), 7 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..1246aa8 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1059,24 +1059,31 @@ static int get_value3(const char **value, * \param uc_mgr Use case manager * \param identifier Value identifier (string) * \param value Returned value string - * \param modifier modifier name (string) + * \param item Modifier of Device name (string) * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >= 0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

2 1

[alsa-devel] [PATCHv2] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> Acked-by: Liam Girdwood <lrg(a)ti.com> --- src/ucm/main.c | 21 ++++++++++++++------- 1 files changed, 14 insertions(+), 7 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..0393601 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1059,24 +1059,31 @@ static int get_value3(const char **value, * \param uc_mgr Use case manager * \param identifier Value identifier (string) * \param value Returned value string - * \param modifier modifier name (string) + * \param item item name (string) * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >= 0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

2 1

[alsa-devel] [PATCH] alsa-lib: ucm: allow values to be read from devices
by Margarita Olaya 18 Mar '11

18 Mar '11

This allows devices value to be read using the get_value(). Also fix a little formatting. Signed-off-by: Margarita Olaya Cabrera <magi(a)slimlogic.co.uk> --- src/ucm/main.c | 19 +++++++++++++------ 1 files changed, 13 insertions(+), 6 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 42fd373..fb11e61 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -1063,20 +1063,27 @@ static int get_value3(const char **value, * \return Zero on success (value is filled), otherwise a negative error code */ static int get_value(snd_use_case_mgr_t *uc_mgr, - const char *identifier, - const char **value, - const char *modifier) + const char *identifier, + const char **value, + const char *item) { - struct use_case_modifier *mod; + struct use_case_modifier *mod; + struct use_case_device *dev; int err; - if (modifier != NULL) { - mod = find_modifier(uc_mgr, modifier); + if (item != NULL) { + mod = find_modifier(uc_mgr, item); if (mod != NULL) { err = get_value1(value, &mod->value_list, identifier); if (err >= 0 || err != -ENOENT) return err; } + dev = find_device(uc_mgr->active_verb, item); + if (dev != NULL) { + err = get_value1(value, &dev->value_list, identifier); + if (err >=0 || err != -ENOENT) + return err; + } } err = get_value1(value, &uc_mgr->active_verb->value_list, identifier); if (err >= 0 || err != -ENOENT) -- 1.7.0.4

3 4

[alsa-devel] Keyclick latency
by Marc-André Hébert 18 Mar '11

18 Mar '11

Hello, I am working on an embedded device which does audio playback. Now when the user presses a key, we want to play a tone to provide a keyclick feedback to the user. What we always did (we were using OSS before) was pretty much the simplest thing possible which is for the application to: -stop sending playback samples -send the keyclick samples -resume sending the playback samples With our new hardware this approach does not produce good results because of the size of the device's ringbuffer. The larger buffer introduces a significant latency issue for the user feedback. Reducing the ring buffer size is not a viable option to ensure no underruns are generated on the system. Now I was wondering if there a good approach for this using alsa. Checking the api, a simple solution would be to: -call snd_pcm_drop -send the keyclick samples -resume sending the playback samples I've tried this and the latency is good, unfortunately there is the drawback that some samples were dropped. So I guess the application could : -keep a copy of the samples sent, -check the amount of samples left in the ringbuffer before calling snd_pcm_drop, -send the keyclick -resend dropped samples -resume sending the playback samples Seems a little overcomplicated so I was wondering if I was missing something. Would there be a simpler approach? Something like pausing the stream or reading the contents of the ringbuffer before dropping. Thanks Marc

2 1

[alsa-devel] [PATCH] ASoC: Mark WM8958 DSP2 registers readable
by Mark Brown 18 Mar '11

18 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 328f328..bdd1ac7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -55,6 +55,9 @@ static int wm8994_retune_mobile_base[] = { static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->control_data; + switch (reg) { case WM8994_GPIO_1: case WM8994_GPIO_2: @@ -71,6 +74,15 @@ static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) case WM8994_INTERRUPT_STATUS_2: case WM8994_INTERRUPT_RAW_STATUS_2: return 1; + + case WM8958_DSP2_PROGRAM: + case WM8958_DSP2_CONFIG: + case WM8958_DSP2_EXECCONTROL: + if (control->type == WM8958) + return 1; + else + return 0; + default: break; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Remove bogus check for register validity in debugfs write
by Mark Brown 18 Mar '11

18 Mar '11

Since not all registers need to be cached and the cache is entirely optional anyway we shouldn't be checking that a register is in the cached range. If the register is invalid then the actual I/O code can determine that and report an error. Similarly, the step size can and should be enforced by the lower level code if it's important. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 17efacd..4dda589 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -259,8 +259,6 @@ static ssize_t codec_reg_write_file(struct file *file, while (*start == ' ') start++; reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->driver->reg_cache_size) || (reg % step)) - return -EINVAL; while (*start == ' ') start++; if (strict_strtoul(start, 16, &value)) -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: Treat WM8958 revision A as WM8994 revision D
by Mark Brown 18 Mar '11

18 Mar '11

The first WM8958 revision requires similar treatment. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 36 ++++++++++++++++++++++++++---------- 1 files changed, 26 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3dc64c8..9458289 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3327,14 +3327,23 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8958: snd_soc_add_controls(codec, wm8958_snd_controls, ARRAY_SIZE(wm8958_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, - ARRAY_SIZE(wm8994_lateclk_widgets)); - snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, - ARRAY_SIZE(wm8994_adc_widgets)); - snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, - ARRAY_SIZE(wm8994_dac_widgets)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); + if (wm8994->revision < 1) { + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, + ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, + ARRAY_SIZE(wm8994_adc_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets, + ARRAY_SIZE(wm8994_dac_revd_widgets)); + } else { + snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, + ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, + ARRAY_SIZE(wm8994_dac_widgets)); + } break; } @@ -3358,10 +3367,17 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: - snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, - ARRAY_SIZE(wm8994_lateclk_intercon)); - snd_soc_dapm_add_routes(dapm, wm8958_intercon, - ARRAY_SIZE(wm8958_intercon)); + if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, + ARRAY_SIZE(wm8994_revd_intercon)); + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, + ARRAY_SIZE(wm8994_lateclk_revd_intercon)); + } else { + snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon, + ARRAY_SIZE(wm8994_lateclk_intercon)); + snd_soc_dapm_add_routes(dapm, wm8958_intercon, + ARRAY_SIZE(wm8958_intercon)); + } break; } -- 1.7.4.1

2 3

[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "RESOURCE_LEAK" checker.
by sudarshan.bisht＠nokia.com 18 Mar '11

18 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- modules/mixer/simple/sbase.c | 1 + modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 4 ++++ src/control/control_hw.c | 4 +++- src/pcm/pcm_file.c | 4 +++- src/pcm/pcm_ladspa.c | 20 ++++++++++++++------ src/pcm/pcm_plug.c | 9 +++++++-- src/pcm/pcm_rate.c | 4 ++++ src/rawmidi/rawmidi.c | 4 +++- 9 files changed, 40 insertions(+), 12 deletions(-) diff --git a/modules/mixer/simple/sbase.c b/modules/mixer/simple/sbase.c index 97feee8..bb2f59d 100644 --- a/modules/mixer/simple/sbase.c +++ b/modules/mixer/simple/sbase.c @@ -377,6 +377,7 @@ static int simple_event_add1(snd_mixer_class_t *class, if (ctype != SND_CTL_ELEM_TYPE_BOOLEAN) { __invalid_type: snd_mixer_selem_id_free(id); + free(hsimple); return -EINVAL; } break; diff --git a/modules/mixer/simple/sbasedl.c b/modules/mixer/simple/sbasedl.c index 0137586..494802f 100644 --- a/modules/mixer/simple/sbasedl.c +++ b/modules/mixer/simple/sbasedl.c @@ -99,7 +99,7 @@ int mixer_simple_basic_dlopen(snd_mixer_class_t *class, __error: if (initflag) free(priv); - if (h == NULL) + if (h) snd_dlclose(h); free(xlib); return -ENXIO; diff --git a/src/conf.c b/src/conf.c index 8939d62..c4daae7 100644 --- a/src/conf.c +++ b/src/conf.c @@ -638,6 +638,7 @@ static int get_char_skip_comments(input_t *input) fd = malloc(sizeof(*fd)); if (!fd) { free(str); + snd_input_close(in); return -ENOMEM; } fd->name = str; @@ -1847,6 +1848,7 @@ int snd_config_delete(snd_config_t *config) list_del(&config->list); free(config->id); free(config); + config = NULL; return 0; } @@ -4589,6 +4591,8 @@ static int parse_args(snd_config_t *subs, const char *str, snd_config_t *defs) if (err < 0) { _err: free(val); + if (sub) + snd_config_delete(sub); return err; } free(val); diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..29cbbc6 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -230,8 +230,10 @@ static int snd_ctl_hw_elem_tlv(snd_ctl_t *handle, int op_flag, return -errno; } if (op_flag == 0) { - if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) + if (xtlv->tlv[1] + 2 * sizeof(unsigned int) > tlv_size) { + free(xtlv); return -EFAULT; + } memcpy(tlv, xtlv->tlv, xtlv->tlv[1] + 2 * sizeof(unsigned int)); } free(xtlv); diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index bfa1cc8..adba7c0 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -226,7 +226,9 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) file->final_fname); return -errno; } - fd = fileno(pipe); + fd = dup(fileno(pipe)); + if (fd >= 0) + fclose(pipe); } else { if (file->trunc) fd = open(file->final_fname, O_WRONLY|O_CREAT|O_TRUNC, diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index c413c10..41e1d38 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -750,8 +750,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads if (instance->input.data == NULL || instance->input.m_data == NULL || instance->output.data == NULL || - instance->output.m_data == NULL) + instance->output.m_data == NULL) { + free(pchannels); return -ENOMEM; + } for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->output.channels.array[idx]; if (pchannels[chn] == NULL && chn < ichannels) { @@ -761,8 +763,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->input.data[idx] = pchannels[chn]; if (instance->input.data[idx] == NULL) { instance->input.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 0); - if (instance->input.data[idx] == NULL) - return -ENOMEM; + if (instance->input.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } } } for (idx = 0; idx < instance->output.channels.size; idx++) { @@ -770,8 +774,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads /* FIXME/OPTIMIZE: check if we can remove double alloc */ /* if LADSPA plugin has no broken inplace */ instance->output.data[idx] = malloc(sizeof(LADSPA_Data) * ladspa->allocated); - if (instance->output.data[idx] == NULL) - return -ENOMEM; + if (instance->output.data[idx] == NULL) { + free(pchannels); + return -ENOMEM; + } pchannels[chn] = instance->output.m_data[idx] = instance->output.data[idx]; } } @@ -793,8 +799,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->output.data[idx] = NULL; } else { instance->output.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 1); - if (instance->output.data[idx] == NULL) + if (instance->output.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } } } } diff --git a/src/pcm/pcm_plug.c b/src/pcm/pcm_plug.c index e9d2923..e80bd3e 100644 --- a/src/pcm/pcm_plug.c +++ b/src/pcm/pcm_plug.c @@ -1298,6 +1298,7 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_route_load_ttable(tt, ttable, csize, ssize, &cused, &sused, -1); if (err < 0) { snd_config_delete(sconf); + free(ttable); return err; } } @@ -1310,12 +1311,16 @@ int _snd_pcm_plug_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_open_slave(&spcm, root, sconf, stream, mode, conf); snd_config_delete(sconf); - if (err < 0) + if (err < 0) { + free(ttable); return err; + } err = snd_pcm_plug_open(pcmp, name, sformat, schannels, srate, rate_converter, route_policy, ttable, ssize, cused, sused, spcm, 1); - if (err < 0) + if (err < 0) { + free(ttable); snd_pcm_close(spcm); + } return err; } #ifndef DOC_HIDDEN diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 70e30e5..cf218ae 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1392,11 +1392,13 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, } else { SNDERR("Invalid type for rate converter"); snd_pcm_close(pcm); + free(rate); return -EINVAL; } if (err < 0) { SNDERR("Cannot find rate converter"); snd_pcm_close(pcm); + free(rate); return -ENOENT; } #else @@ -1405,6 +1407,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, err = open_func(SND_PCM_RATE_PLUGIN_VERSION, &rate->obj, &rate->ops); if (err < 0) { snd_pcm_close(pcm); + free(rate); return err; } #endif @@ -1413,6 +1416,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, ! rate->ops.input_frames || ! rate->ops.output_frames) { SNDERR("Inproper rate plugin %s initialization", type); snd_pcm_close(pcm); + free(rate); return err; } diff --git a/src/rawmidi/rawmidi.c b/src/rawmidi/rawmidi.c index b28488a..25732d2 100644 --- a/src/rawmidi/rawmidi.c +++ b/src/rawmidi/rawmidi.c @@ -255,8 +255,10 @@ static int snd_rawmidi_open_conf(snd_rawmidi_t **inputp, snd_rawmidi_t **outputp snd_config_delete(type_conf); if (err >= 0) err = open_func(inputp, outputp, name, rawmidi_root, rawmidi_conf, mode); - if (err < 0) + if (err < 0) { + snd_dlclose(h); return err; + } if (inputp) { (*inputp)->dl_handle = h; h = NULL; snd_rawmidi_params_default(*inputp, &params); -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/1] Fixed issues/defects reported by Coverity tool
by sudarshan.bisht＠nokia.com 18 Mar '11

18 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has got fixes for many potential memory leaks. Sudarshan (1): alsa-lib: fixed coverity reported issues under "RESOURCE_LEAK" checker. modules/mixer/simple/sbase.c | 1 + modules/mixer/simple/sbasedl.c | 2 +- src/conf.c | 2 ++ src/control/control_hw.c | 4 +++- src/pcm/pcm_file.c | 5 +++-- src/pcm/pcm_ladspa.c | 20 ++++++++++++++------ src/pcm/pcm_plug.c | 5 ++++- src/pcm/pcm_rate.c | 1 + src/rawmidi/rawmidi.c | 4 +++- 9 files changed, 32 insertions(+), 12 deletions(-)

3 3

[alsa-devel] Destruction of the sound after prolonged output
by Максим 18 Mar '11

18 Mar '11

Hi, My name is Max. I've got some problems using ALSA, a certain type of destruction of sound appears after a prolonged sound output. The destruction is like "bit crush", stutterings, interruptions, etc. Once started, it no longer stops. It starts almost always but only after a prolonged sound output, nearby 24 hours. To reproduce it shurely I use my test wav file and standard speaker-test utility with following options: speaker-test -t wav -w test.wav -r 16000 I've attached test.wav to this mail. To reproduce this I use a fresh minimal installation of Ubuntu 10.10, command line only, with standard Ubuntu 2.6.35 generic kernel. Minimal installation is made by installing Ubuntu from an "alternate" version of the installation CD. It provide option "command line minimal install" that allow install only base system with command line. So there isn't any desktop applications like Gnome, Pulseaudio, etc. I've used standard Ubuntu package of ALSA with version 1.0.23, and also with compiled 1.0.24. I've tried to reproduce this behavior on Centos and Fedora (both in single mode, command line only) with Intel HDA chipset, and on Ubuntu with a Realtek chipset. The results was the same, sound destruction after a day of sound output. Above is the output of speaker-test with parameters: speaker-test -d -c 2 -t wav Before this command my sound test was used during about 6 hours to reproduce the sound destruction: speaker-test -t wav -w test.wav -r 16000 Plug PCM: Linear conversion PCM (S32_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Soft volume PCM Control: PCM Playback Volume min_dB: -51 max_dB: 0 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 16384 stop_threshold : 16384 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 16384 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 appl_ptr : 0 hw_ptr : 926558209

2 1

Re: [alsa-devel] Question about your DSP topic branch
by Patrick Lai 18 Mar '11

18 Mar '11

Hi Liam, I have two more questions about your DSP topic branch 7. I see in sdp4430.c, SDP4430 MODEM front-end dailink no_host_mode is set to SND_SOC_DAI_LINK_NO_HOST. What is the purpose of no_host_mode? Is it for use case that two physical devices can exchange audio data without host-processor intervention? If so, if user-space application tries to write to PCM buffer, will framework reject the buffer? 8. I see there is dmic codec(dmic.c) under sound/soc/codec which is pretty much just a dummy codec driver. I supposed the configuration of DMIC is done in other driver. Would it be better if we could have something like fixed voltage regulator? So, there is no need to duplicate the effort. Look forward to seeing your reply soon Thanks On 1/6/2011 3:39 PM, Patrick Lai wrote: > Hi Liam, > > I sync to your kernel DSP topic branch two days back in attempt to > understand the up-coming ASOC DSP design. I have few questions to get > clarification from you. > > 1. In the sdp4430.c, there are both FE dai link and BE dai link have > omap-aess-audio as platform driver. How is omap-aess-audio platform > driver used in both front-end and backend? > > 2. Front-end dai with stream name "Multimedia" has platform driver as > omap-pcm-audio which is the DMA based. This front-end dai is mapped to > backend (i.e PDM-DL1) with platform driver as omap-aess-audio. > Omap-aess-audio looks to me is DSP platform driver. If a stream is DMA > based, it seems strange to have DSP-based backend. > > 3. To best of my knowledge, I see in omap-abe.c which is the front end > implementation. Playback_trigger() seems to go ahead enable all > back-ends linked to the given front-end. However, front end may not be > routed to all back-ends. Where in the code to make sure BE is only > activated when a stream is being routed to that particular back-end? > > 4. omap-abe.c manages activation of BE DAIS and omap-abe-dsp.c manages > routing of front-end to back-end DAPM widgets and routing map. Am I > correct? This leads to previous question. How are two drivers working > together to make sure BEs are only activated if Front-end has been > routed to them? > > 5. Is there mechanism for front-end to switch between DMA and DSP > platform driver? It looks to me that mapping of frond-end and platform > driver is predetermined based on use case. For example, HDMI goes > through DMA as legacy dailink. > > 6. struct abe_data has array member called dapm. It looks to me that > this array simply tracks dapm components status but I don't see it's > really being used in meaningful way in omap-abe-adsp.c. > > Thanks > -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 13

[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "REVERSE_INULL" checker.
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- src/control/setup.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/src/control/setup.c b/src/control/setup.c index eecda45..3606e5c 100644 --- a/src/control/setup.c +++ b/src/control/setup.c @@ -398,7 +398,7 @@ static int snd_config_get_ctl_elem_value(snd_config_t *conf, static int add_elem(snd_sctl_t *h, snd_config_t *_conf, snd_config_t *private_data) { - snd_config_t *conf; + snd_config_t *conf = NULL ; snd_config_iterator_t i, next; char *tmp; int iface = SND_CTL_ELEM_IFACE_MIXER; @@ -412,6 +412,9 @@ static int add_elem(snd_sctl_t *h, snd_config_t *_conf, snd_config_t *private_da snd_config_t *value = NULL, *mask = NULL; snd_sctl_elem_t *elem = NULL; int err; + /* The last parameter of snd_config_expand() is an output parameter, and + * it is guaranteed that this parameter is set when the function succeeds. + */ err = snd_config_expand(_conf, _conf, NULL, private_data, &conf); if (err < 0) return err; -- 1.7.0.4

1 0

[alsa-devel] [PATCH 0/1] Fixed issues/defects reported by Coverity tool
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has a fix for a situation where variable can become NULL after certain function call, but later not been checked before passing to the next function call. Sudarshan (1): alsa-lib: fixed coverity reported issues under "REVERSE_INULL" checker. src/control/setup.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

3 3

[alsa-devel] [PATCH] alsa-lib: fixed coverity reported issues under "USE_AFTER_FREE" checker.
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> --- modules/mixer/simple/python.c | 1 + src/control/control_hw.c | 1 + src/mixer/mixer.c | 1 - src/mixer/simple_abst.c | 2 ++ 4 files changed, 4 insertions(+), 1 deletions(-) diff --git a/modules/mixer/simple/python.c b/modules/mixer/simple/python.c index c822c52..784d9a0 100644 --- a/modules/mixer/simple/python.c +++ b/modules/mixer/simple/python.c @@ -663,6 +663,7 @@ pymixer_attach_hctl(struct pymixer *pymixer, PyObject *args) return NULL; err = snd_mixer_attach_hctl(pymixer->mixer, hctl); if (err < 0) { + snd_hctl_close(hctl); PyErr_Format(PyExc_RuntimeError, "Cannot attach hctl: %s", snd_strerror(err)); return NULL; } diff --git a/src/control/control_hw.c b/src/control/control_hw.c index cf258b4..16c4987 100644 --- a/src/control/control_hw.c +++ b/src/control/control_hw.c @@ -414,6 +414,7 @@ int snd_ctl_hw_open(snd_ctl_t **handle, const char *name, int card, int mode) if (err < 0) { close(fd); free(hw); + return err; } ctl->ops = &snd_ctl_hw_ops; ctl->private_data = hw; diff --git a/src/mixer/mixer.c b/src/mixer/mixer.c index 85d843f..5e5789a 100644 --- a/src/mixer/mixer.c +++ b/src/mixer/mixer.c @@ -228,7 +228,6 @@ int snd_mixer_attach_hctl(snd_mixer_t *mixer, snd_hctl_t *hctl) return -ENOMEM; err = snd_hctl_nonblock(hctl, 1); if (err < 0) { - snd_hctl_close(hctl); free(slave); return err; } diff --git a/src/mixer/simple_abst.c b/src/mixer/simple_abst.c index 9e9aaf5..127d8d2 100644 --- a/src/mixer/simple_abst.c +++ b/src/mixer/simple_abst.c @@ -336,6 +336,8 @@ int snd_mixer_simple_basic_register(snd_mixer_t *mixer, err = find_module(class, top); if (err >= 0) err = snd_mixer_attach_hctl(mixer, priv->hctl); + if(err < 0) + goto __error; if (err >= 0) { priv->attach_flag = 1; err = snd_mixer_class_register(class, mixer); -- 1.7.0.4

1 0

[alsa-devel] [PATCH] *** Fixed issues/defects reported by Coverity tool ***
by sudarshan.bisht＠nokia.com 17 Mar '11

17 Mar '11

From: Sudarshan <sudarshan.bisht(a)nokia.com> Coverity Static Analysis helps developers find hard-to-spot, yet potentially crash-causing defects early in the development phase, reducing the cost,time, and risk of software errors. This patch has a fix for conditions where 1) variable is used evern after getting freed thus causing junk assignment and 2) double free of memory. Sudarshan (1): alsa-lib: fixed coverity reported issues under "USE_AFTER_FREE" checker. src/control/control_hw.c | 1 + src/mixer/mixer.c | 2 +- 2 files changed, 2 insertions(+), 1 deletions(-)

3 3

[alsa-devel] some questions about sound system
by loody 17 Mar '11

17 Mar '11

Dear all: I just install Ubuntu9.10 on my laptop and I try to update my sound system and recompile the utility. there are some questions I met: 1. when configure the utility it says checking panel.h presence... no checking for panel.h... no configure: error: required curses helper header not found and config.log is "conftest.c:10:42: error: CoreFoundation/CFPreferences.h: No such file or directory conftest.c: In function 'main': conftest.c:14: error: 'NULL' undeclared (first use in this function) conftest.c:14: error: (Each undeclared identifier is repor" the above error seems I miss install some package before. I google the name but I have no improvement. "conftest.c:20:28: error: ac_nonexistent.h: No such file or directory configure:7397: $? = 1" the above error seems the header located where the conftest.c doesn't expect. how could solve them all? 2. there are some configure file in share when I compile advance linux sound architecture. like alsa.conf. have anyone know where I can find the naming rule? appreciate your help, miloody

4 7

[alsa-devel] ALC663 on ASUS G72GX Laptop
by nibs 16 Mar '11

16 Mar '11

I was advised to inquire about this here. Should the driver create two digital devices? _From the ALC663 datasheet: "In addition to the standard (primary) S/PDIF output function, the ALC663 features another independent (secondary) S/PDIF-OUT output and converters that transport digital audio output to a High Definition Media Interface (HDMI) transmitter (becoming more common in high-end PCs." More information here. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5310

1 0

[alsa-devel] UCM list 'modifiers' Seg-Fault when list is empty
by Alejandro Gonzalez 16 Mar '11

16 Mar '11

Hi, I've been testing the Alsa UCM and find that when listing 'modifiers' there is a segmentation fault if the list is empty. $ alsaucm -c Panda set _verb HiFi list _modifiers Im setting defaults list is empty Segmentation fault (core dumped) Using 'gdb' I see that the issue is trying to free a not NULL pointer. Program received signal SIGSEGV, Segmentation fault. 0x402a096c in free () from /lib/libc.so.6 (gdb) bt #0 0x402a096c in free () from /lib/libc.so.6 #1 0x40221224 in snd_use_case_free_list () from /usr/lib/libasound.so.2 #2 0x00008d8a in do_one (context=0x13008) at usecase.c:256 #3 do_commands (context=0x13008) at usecase.c:329 #4 0x000091aa in main (argc=<value optimized out>, argv=0xbe9986c4) at usecase.c:440 (gdb) quit But this can be easily resolved with the correct initialization of the '**list' pointer on the 'do_one' function in 'alsaucm/usecase.c'. You can find a patch attached to the mail. Regards, Alejandro

1 0

[alsa-devel] [PATCH] sound/core: merge list_del()/list_add_tail() to list_move_tail()
by Nicolas Kaiser 16 Mar '11

16 Mar '11

Merge list_del() + list_add_tail() to list_move_tail(). Signed-off-by: Nicolas Kaiser <nikai(a)nikai.net> --- sound/core/timer.c | 8 +++----- 1 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index ed01632..7c1cbf0 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -186,9 +186,8 @@ static void snd_timer_check_slave(struct snd_timer_instance *slave) list_for_each_entry(master, &timer->open_list_head, open_list) { if (slave->slave_class == master->slave_class && slave->slave_id == master->slave_id) { - list_del(&slave->open_list); - list_add_tail(&slave->open_list, - &master->slave_list_head); + list_move_tail(&slave->open_list, + &master->slave_list_head); spin_lock_irq(&slave_active_lock); slave->master = master; slave->timer = master->timer; @@ -414,8 +413,7 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) static int snd_timer_start1(struct snd_timer *timer, struct snd_timer_instance *timeri, unsigned long sticks) { - list_del(&timeri->active_list); - list_add_tail(&timeri->active_list, &timer->active_list_head); + list_move_tail(&timeri->active_list, &timer->active_list_head); if (timer->running) { if (timer->hw.flags & SNDRV_TIMER_HW_SLAVE) goto __start_now; -- 1.7.3.4

1 0

[alsa-devel] [PATCH] ASoC: Add National Semiconductor LM49352 Audio Codec support
by Reddy, MR Swami 16 Mar '11

16 Mar '11

Hello, Patch for adding National Semiconductor LM49352 Audio Codec (http://www.national.com/ds/LM/LM49352.pdf) support along with Samsung_6410 (i.e. SMDK6410) platform support for LM49352. [Tested this patch on SMDK6410 platform with Kernel- 2.6.24 and ALSA version is - 1.0.15]. Please review and let me know the comments/suggestion on this patch. And also let me know the forward-porting (to the latest ALSA version APIs) steps/process. Thanks in advance. Thanks Swami

2 6

[alsa-devel] aseqnet ipv6 and embedded clients
by errordeveloper＠gmail.com 16 Mar '11

16 Mar '11

Hi, I am looking at implementing wireless MIDI network using Contiki OS. I would run it on relativelly powerfull microcontroller (Freescale MC13224). I looked at sourcode of aseqnet command and it does look fairly small. Contiki has uIP 6loWPAN for networking and it connects to linux using SLIP-like 'tun' virtual userspace interface. I wonder if there could be any potential issues in terms of IPv6 compatibility, would aseqnet on Linux work with IPv6 or I will certainly need a tunnel ? Would be greate to hear _ANY_ concerns, of course I do imagine there will be a latency (hopefully not too huge) and I hope to work on this further. I also though of MIDI RTP, but that seems quite a bit bigger thing .. Could anyone give a little insight on what aseqnet really does? I would imagine it is quite plain, am I wrong ? Well, I could perhaps implement some very simplistic netcat/telnet kind of data dumping, but would be greate to get working smothly end- to-end, i.e. all the way from my wireless node (which will have a MIDI din5 connector) to an app on Linux, even amidi --dump ..hm Also would be greate to get some idea how clocking works w/ aseqnet. Cheers, -- Ilya

1 0

[alsa-devel] Wrong email in author-tag.
by Peter Rosin 16 Mar '11

16 Mar '11

Hi Takashi, This is not really important to /me/, on /this/ occation. But I thought I'd ask anyway as it seems strange that the author tag ended up wrong. If it is a symptom of more errors, others may care more. Personally I'm mainly wondering how it could happen, and would like for it to not happen again (especially if I ever send more substantial patches). The commit e2e9566230 ALSA: AT73C213: Rectify misleading comment. was first sent as a git patch [1] and then again amended with a pair of other (seemingly) trivial comment fixes [2] (the amended fixes were not commited for some reason). Both patches were sent from my axentia account and both carried signed-off-by tags using the axentia account. Yet the commit is made naming my lysator account as author (the sign-off is still intact). The patch that seems to have ended up being commited is [1], and what I think has happened is that I used the lysator account in the discussion following the 2nd patch, but I fail to understand how the sender of that one message (or was it a few?) could "leak" into the author tag. I'm assuming it was just a silly mistake, I'm mearly pointing out the mishap and understand that there is little one can do to fix this buglet this late. Cheers, Peter [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033550.h… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033645.h…

2 1

[alsa-devel] [GIT PULL] Updates for 2.6.39
by Liam Girdwood 16 Mar '11

16 Mar '11

These should be the last updates for 2.6.39. Thanks Liam --- The following changes since commit 273de37655369e57919f19e157fafb2b4b1e4472: Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.39 (2011-02-01 14:55:10 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.39 Jarkko Nikula (5): ASoC: omap: rx51: Use gpio_request_one to configure tvout_sel gpio ASoC: omap: rx51: Add headset support ASoC: omap: rx51: Report headset insertion instead of video out cable ASoC: omap: rx51: Add FM transmitter support ASoC: omap: rx51: Enable McBSP2 sidetone sound/soc/omap/rx51.c | 42 ++++++++++++++++++++++++++++++++++++------ 1 files changed, 36 insertions(+), 6 deletions(-)

2 1

[alsa-devel] [PATCH 1/2 v3] ALSA: Add snd_ctl_update() to dynamically update a control
by Dimitris Papastamos 16 Mar '11

16 Mar '11

Add a function to dynamically update a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or update existing controls. A separate patch needs to be made to allow ALSA Mixer to update the controls dynamically. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- Hi, Any potentially locking related issues have been fixed now. Any comments on whether it would be advisable to use the same `numid' when I update an existing control? Thanks, Dimitris include/sound/control.h | 1 + sound/core/control.c | 65 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 66 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index e67db28..2a2fd82 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -113,6 +113,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, v void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_update(struct snd_card * card, struct snd_kcontrol * kcontrol, bool add_on_update); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, diff --git a/sound/core/control.c b/sound/core/control.c index db51e4e..1d2b524 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -368,6 +368,71 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_add); /** + * snd_ctl_update - update the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to update + * @add_on_update: add the control if not already added + * + * Updates the given control. If the given control does not exist + * and the add_on_update flag is set, the control is added. If the + * control exists, it is destroyed first. + * + * Returns zero if successful, or a negative error code on failure. + * + * It frees automatically the control which cannot be added or updated. + */ +int snd_ctl_update(struct snd_card *card, struct snd_kcontrol *kcontrol, + bool add_on_update) +{ + struct snd_ctl_elem_id id; + unsigned int idx; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) { + ret = -EINVAL; + goto error; + } + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + if (add_on_update) + goto add; + up_write(&card->controls_rwsem); + ret = -EINVAL; + goto error; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) { + up_write(&card->controls_rwsem); + goto error; + } +add: + if (snd_ctl_find_hole(card, kcontrol->count) < 0) { + up_write(&card->controls_rwsem); + ret = -ENOMEM; + goto error; + } + list_add_tail(&kcontrol->list, &card->controls); + card->controls_count += kcontrol->count; + kcontrol->id.numid = card->last_numid + 1; + card->last_numid += kcontrol->count; + up_write(&card->controls_rwsem); + for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); + return 0; + +error: + snd_ctl_free_one(kcontrol); + return ret; +} + +EXPORT_SYMBOL(snd_ctl_update); + +/** * snd_ctl_remove - remove the control from the card and release it * @card: the card instance * @kcontrol: the control instance to remove -- 1.7.4.1

2 2

[alsa-devel] [PATCH] ALSA: HDA: Correct address for SSYNC register
by David Henningsson 16 Mar '11

16 Mar '11

Here's something I stumbled upon, while debugging something probably unrelated. I don't exactly know the implications of failing multiple stream synchronisation (is that something we commonly use?), or the implications of writing to a register that is "Reserved" in the HDA Spec (can there be crazy vendor magic in there we're unwillingly messing with?). Perhaps someone else knows more and can tell if this is something that is actually important. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] RFC: ST specific ASoC sound driver
by rajeev 16 Mar '11

16 Mar '11

Hi, I have developed a ASoC sound driver for a ST specific platform (SPEAr). This module actually uses the Designware (Synopsys) "dw_apb_i2s_db" IP + some ST specific glue logic on top of it. Now I am confused as to whether I should submit a single patch for the SPEAr ASoC driver or should I try to break the driver such that the Designware and glue logic parts are handled separately (which given the current shape of design seems improbable) Any pointers on the same will be highly appreciated. Best Regards Rajeev

2 1

[PATCH] ALSA: HDA: Correct address for SYNC register
by David Henningsson 16 Mar '11

16 Mar '11

The SSYNC register is located at address 0x38 according to the HDA specification. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 70a9d32..ab3f654 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -176,7 +176,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #define ICH6_REG_INTCTL 0x20 #define ICH6_REG_INTSTS 0x24 #define ICH6_REG_WALLCLK 0x30 /* 24Mhz source */ -#define ICH6_REG_SYNC 0x34 +#define ICH6_REG_SYNC 0x38 #define ICH6_REG_CORBLBASE 0x40 #define ICH6_REG_CORBUBASE 0x44 #define ICH6_REG_CORBWP 0x48 -- 1.7.1 --------------050206000100040505070002--

1 0

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 16 Mar '11

16 Mar '11

The following changes since commit 80e1b84ffe1354f314fd85c939900b816f394e5a: Merge branch 'for-2.6.38' into for-2.6.39 (2011-03-10 14:16:35 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 A final batch of updates for 2.6.39 - these have been sitting in my -next tree for several days now. Christian Glindkamp (1): ASoC: Add MAX9850 codec driver Mark Brown (1): ASoC: Fix spacing in MAX8950 Takashi Iwai (1): ALSA: Add snd_ctl_activate_id() Vasily Khoruzhick (2): ASoC: PXA: z2: Mute internal speaker when headphones are connected ASoC: PXA: Z2: Fix codec pin name include/sound/control.h | 2 + sound/core/control.c | 46 +++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9850.c | 389 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9850.h | 38 +++++ sound/soc/pxa/z2.c | 7 +- 7 files changed, 487 insertions(+), 1 deletions(-) create mode 100644 sound/soc/codecs/max9850.c create mode 100644 sound/soc/codecs/max9850.h

3 3

[alsa-devel] [PATCHv2] omap3: alsa-soc: Remove restrictive checks for cpu type
by Sanjeev Premi 16 Mar '11

16 Mar '11

Current checks for cpu type were too restrictive leading to failures for other silicons in same family. The problem was found while testing audio playback on AM37x and AM35x processors. But should exist on OMAP36xx as well. Signed-off-by: Sanjeev Premi <premi(a)ti.com> --- Changes in this version: 1) Added one more check to the changes. 2) Updated commit message to use plurals. sound/soc/omap/omap-mcbsp.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2175f09..a88a0bf 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x() || cpu_is_omap44xx()) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - if (cpu_is_omap343x()) { + if (cpu_is_omap34xx()) { dma_data->set_threshold = omap_mcbsp_set_threshold; /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */ if (omap_mcbsp_get_dma_op_mode(bus_id) == -- 1.7.2.2

1 0

Re: [alsa-devel] [PATCHv2] omap3: alsa-soc: Remove restrictive checks for cpu type
by Premi, Sanjeev 16 Mar '11

16 Mar '11

> -----Original Message----- > From: Premi, Sanjeev > Sent: Monday, March 14, 2011 8:53 PM > To: linux-omap(a)vger.kernel.org; alsa-devel(a)alsa-project.org; > linux-arm-kernel(a)lists.infradead.org > Cc: Premi, Sanjeev > Subject: [PATCHv2] omap3: alsa-soc: Remove restrictive checks > for cpu type > > Current checks for cpu type were too restrictive leading > to failures for other silicons in same family. > > The problem was found while testing audio playback on > AM37x and AM35x processors. But should exist on OMAP36xx > as well. > > Signed-off-by: Sanjeev Premi <premi(a)ti.com> > --- > Changes in this version: > 1) Added one more check to the changes. > 2) Updated commit message to use plurals. > > sound/soc/omap/omap-mcbsp.c | 4 ++-- > 1 files changed, 2 insertions(+), 2 deletions(-) [sp] This patch didn't seem to be lost in transit! Sending it again. Did anyone else ever face this issue? ~sanjeev

1 0

[alsa-devel] [PATCH 1/2 v2] ALSA: Add snd_ctl_update() to dynamically update a control
by Dimitris Papastamos 16 Mar '11

16 Mar '11

Add a function to dynamically update a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or update existing controls. A separate patch needs to be made to make ALSA Mixer cope with this. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/control.h | 1 + sound/core/control.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index e67db28..12e4de7 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -113,6 +113,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, v void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_update(struct snd_card * card, struct snd_kcontrol * kcontrol, int add_on_update); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, diff --git a/sound/core/control.c b/sound/core/control.c index db51e4e..9c5d7a9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -397,6 +397,52 @@ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_remove); /** + * snd_ctl_update - update the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to update + * @add_on_update: add the control if not already added + * + * Updates the control instance created via snd_ctl_new() or + * snd_ctl_new1() of the given card. + * + * Returns zero if successful, or a negative error code on failure. + * + */ +int snd_ctl_update(struct snd_card *card, struct snd_kcontrol *kcontrol, + int add_on_update) +{ + struct snd_ctl_elem_id id; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) + return -EINVAL; + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + up_write(&card->controls_rwsem); + if (add_on_update) + return snd_ctl_add(card, kcontrol); + return -EINVAL; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) { + up_write(&card->controls_rwsem); + return ret; + } + up_write(&card->controls_rwsem); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) + return ret; + return 0; +} + +EXPORT_SYMBOL(snd_ctl_update); + +/** * snd_ctl_remove_id - remove the control of the given id and release it * @card: the card instance * @id: the control id to remove -- 1.7.4.1

4 7

[alsa-devel] [PATCH 12/13] alsa: use list_move() instead of list_del()/list_add() combination
by Kirill A. Shutemov 16 Mar '11

16 Mar '11

Signed-off-by: Kirill A. Shutemov <kirill(a)shutemov.name> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- sound/pci/ctxfi/ctvmem.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 65da6e4..b78f3fc 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -52,8 +52,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size) if (entry->size == size) { /* Move the vm node from unused list to used list directly */ - list_del(&entry->list); - list_add(&entry->list, &vm->used); + list_move(&entry->list, &vm->used); vm->size -= size; block = entry; goto out; -- 1.7.4.1

2 1

[alsa-devel] [PATCH -next] soc: sgtl5000 uses regulator interfaces
by Randy Dunlap 15 Mar '11

15 Mar '11

From: Randy Dunlap <randy.dunlap(a)oracle.com> sgtl5000 uses regulator interfaces, so make selecting it depend on REGULATOR. sgtl5000.c:(.text+0x4ae33): undefined reference to `regulator_unregister' sgtl5000.c:(.text+0x4af41): undefined reference to `rdev_get_drvdata' sgtl5000.c:(.text+0x4b191): undefined reference to `regulator_register' Signed-off-by: Randy Dunlap <randy.dunlap(a)oracle.com> Cc: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20110304.orig/sound/soc/codecs/Kconfig +++ linux-next-20110304/sound/soc/codecs/Kconfig @@ -33,7 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 - select SND_SOC_SGTL5000 if I2C + select SND_SOC_SGTL5000 if I2C && REGULATOR select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C

3 3

[alsa-devel] Linux Silicon Image HDMI (SiI9135) Audio Driver
by He Yuan -X (heyuan - Strataserv , Inc at Cisco) 15 Mar '11

15 Mar '11

Hi All, I'm looking for Linux Silicon Image HDMI (SiI9135) audio driver. I'd greatly appreciate it if anyone could share with me or point me where I can find it. Thank you very much in advance for your kind help. Best regards, River

2 1

[alsa-devel] Error in configuring playback device
by Irfan Shaikh 15 Mar '11

15 Mar '11

Hello all Problem : 1) I have an playback thread which i am closing and creating based on certain conditions Some times the my thread does not exit grace fully due to which my snd_pcm_close() does not gets executed. 2) Next time when i create thread i get error in configuring playback device My idea to solve above problem : Can i alwayse close the device handle before opening it if pointer to playback handle exists For eg : if(playback_handle) snd_pcm_close(playback_handle); snd_pcm_open(playback_handle); Please suugest me some better method to solve the problem. Regards SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

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[alsa-devel] [PATCH 1/2] ALSA: Add snd_ctl_update() to dynamically update a control
by Dimitris Papastamos 15 Mar '11

15 Mar '11

Add a function to dynamically update a given control. If the control does not already exist, a third parameter is used to determine whether to actually add that control. This is useful in cases where downloadable firmware at runtime can add or update existing controls. A separate patch needs to be made to make ALSA Mixer cope with this. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/control.h | 1 + sound/core/control.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 45 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index e67db28..12e4de7 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -113,6 +113,7 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, v void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_update(struct snd_card * card, struct snd_kcontrol * kcontrol, int add_on_update); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, diff --git a/sound/core/control.c b/sound/core/control.c index db51e4e..db23745 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -397,6 +397,50 @@ int snd_ctl_remove(struct snd_card *card, struct snd_kcontrol *kcontrol) EXPORT_SYMBOL(snd_ctl_remove); /** + * snd_ctl_update - update the control instance of the card + * @card: the card instance + * @kcontrol: the control instance to update + * @add_on_update: add the control if not already added + * + * Updates the control instance created via snd_ctl_new() or + * snd_ctl_new1() of the given card. + * + * Returns zero if successful, or a negative error code on failure. + * + */ +int snd_ctl_update(struct snd_card *card, struct snd_kcontrol *kcontrol, + int add_on_update) +{ + struct snd_ctl_elem_id id; + struct snd_kcontrol *old; + int ret; + + if (!kcontrol) + return -EINVAL; + if (snd_BUG_ON(!card || !kcontrol->info)) + return -EINVAL; + id = kcontrol->id; + down_write(&card->controls_rwsem); + old = snd_ctl_find_id(card, &id); + if (!old) { + up_write(&card->controls_rwsem); + if (add_on_update) + return snd_ctl_add(card, kcontrol); + return -EINVAL; + } + ret = snd_ctl_remove(card, old); + if (ret < 0) + return ret; + up_write(&card->controls_rwsem); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) + return ret; + return 0; +} + +EXPORT_SYMBOL(snd_ctl_update); + +/** * snd_ctl_remove_id - remove the control of the given id and release it * @card: the card instance * @id: the control id to remove -- 1.7.4.1

2 2

[alsa-devel] codec power question
by Koul, Vinod 15 Mar '11

15 Mar '11

Hi, In some cases the audio stream is kept ON and idle by application. In that case codec is also ON. Is there a mechanism where DAPM can detect this idle condition and turn off the idle widgets or turn of the codec BIAS -- ~Vinod

2 4

[alsa-devel] Dsnoop-plugin buffer sizes
by Magnus Olsson 15 Mar '11

15 Mar '11

Hi, I'm having issues with garbage data in my audio stream, which I believe is caused by incorrect buffer sizes somewhere in the audio pipeline. The garbage occurs periodically and is always constant in length. This problem only triggers when I use the dsnoo-plugin. If running directly towards the device, the audio stream is fine (no garbage). I've not tested if this occurs with any other plugin. I'm using Dsnoop to capture a mono stream from a stereo device using arecord, arecord -v -r 16000 -c1 -Daudiosource0 -fS16_LE /tmp/in.wav (where audiosource0 is my dsnoop-device, bound to channel 0) The verbose info from arecord shows me that ALSA has correctly created the Dsnoop device, with a "buffer_size" of 8000. It also shows that it has opened my Hardware device in MMAP-mode, with a "buffer_size" of 8192. Here comes the kicker: The length of the garbage seen in my audio stream is always equal to the difference between the two buffer sizes. So in the above case, I'm seeing 8000 valid samples, followed by 192 "garbage" samples. Furthermore, I've added debugprints to the PCM driver hw_params() function, and in the above configuration, and it shows that the PCM is asked to allocate 4 periods of 8000 bytes each, NOT 8192 bytes as one would expect for the Hardware device. I added some poisoning to the pre-allocated PCM buffer, and it clearly shows that the garbage comes from ALSA reading data from places in the buffer it has not allocated: ALSA thinks the ring buffer contains periods of size 8192, but the PCM driver is only asked to alloc size 8000. Here comes the second kicker: I originally noticed this issue as I was adjusting the periods_min parameter of the PCM driver (PCM hardware has some special needs). It seems like for some periods_min, ALSA will "default" to a buffer_size so that the buffer_size of my Dsnoop device AND my hardware device is identical. In this case, everything works. For example, if I set period_min to 4, ALSA will default to Dsnoop buffer_size 8000 and Hardware buffer_size 8192. If I set periods_min to 3, ALSA will instead default Dsnoop buffer_size to 6000 and Hardware buffer_size to 6000. In the latter case, no garbage appears, since there is no buffer size diff. Now my question is if it's possible that I have some misconfiguration and/or missing constraint in my system? .. or maybe this is just a bug. My first thought was that 8000 was rounded to the nearest page 8192, however, for periods_min=3, both buffers stay at 6000 which is not a page multiple. Best regards, Magnus

2 4

[alsa-devel] Need Help For submitting A SOC drivers
by rajeev 15 Mar '11

15 Mar '11

Hi, I have developed a SOC based sound card driver for a ST specific SOC and also a codec driver. I wish to submit the same to mainline. Please let me know the procedure for the same. Best Regards Rajeev Kumar ST Microelectronics

3 3

[alsa-devel] QasMixer 0.9.0 released
by Sebastian H. 14 Mar '11

14 Mar '11

Hello Alsa and Qt users QasMixer version 0.9.0 is now available. This version fixes some issues with different desktop themes and brings new scaling check button widgets. New features * New scaling check button widget for the sliders pad * New notification balloon timeout option Visible changes * Fix for the widget spacing in the switches area on some desktop themes * Use of QCheckBoxes instead of custom check box widgets - better desktop theme compatibility * Reduced manual styling that may interfere with desktop themes * Playback/Capture sliders labels colors are back * Overhauled settings dialog * Application icon tweaks * Layout tweaks Core changes * Optimizations and fixes Homepage with more informations and screenshots http://xwmw.org/qasmixer Files are hosted at sourceforge http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] [PATCH] alsa-driver git repo: Build problems when copying symlink to itself
by Colin Guthrie 14 Mar '11

14 Mar '11

Hi, After doing a "simple": git clone alsa-driver git clone alsa-kernel cd alsa-driver ./gitcompile --with-debug=full --with-isapnp=yes --with-sequencer=yes --with-moddir=updates/alsa: The build bombed out pretty quickly with the following error. The attached patch fixes it. Hacking autoconf.h... make[1]: Entering directory `/home/colin/Development/Personal/ALSA/alsa-driver/include' make -C sound prepare make[2]: Entering directory `/home/colin/Development/Personal/ALSA/alsa-driver/include/sound' make prepare2 make[3]: Entering directory `/home/colin/Development/Personal/ALSA/alsa-driver/include/sound' cp ../../alsa-kernel/include/ac97_codec.h . patch -p0 -i ac97_codec.patch ac97_codec.h patching file ac97_codec.h Hunk #1 succeeded at 650 (offset 5 lines). cp ../../alsa-kernel/include/core.h . patch -p0 -i core.patch core.h patching file core.h cp ../../alsa-kernel/include/info.h . cp: `../../alsa-kernel/include/info.h' and `./info.h' are the same file make[3]: *** [info.h] Error 1 make[3]: Leaving directory `/home/colin/Development/Personal/ALSA/alsa-driver/include/sound' make[2]: *** [prepare] Error 2 make[2]: Leaving directory `/home/colin/Development/Personal/ALSA/alsa-driver/include/sound' make[1]: *** [prepare] Error 2 make[1]: Leaving directory `/home/colin/Development/Personal/ALSA/alsa-driver/include' make: *** [dep] Error 1 -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

3 10

[alsa-devel] The sound-2.6.git tree is gone?
by David Henningsson 14 Mar '11

14 Mar '11

$ git clone git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git Initialized empty Git repository in /tmp/sound-2.6/.git/ fatal: The remote end hung up unexpectedly Where has the tree gone? I see no sign of it at http://git.kernel.org. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH] omap3: alsa-soc: Remove restrictive check for cpu type
by Sanjeev Premi 14 Mar '11

14 Mar '11

Current check for cpu type is too restrictive leading to failures for other silicons in same family. The problem was found while testing audio playback on AM37x and AM35x processors. But should exist on OMAP36xx as well. Signed-off-by: Sanjeev Premi <premi(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2175f09..84d434a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x() || cpu_is_omap44xx()) { + if (cpu_is_omap34xx() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns -- 1.7.2.2

3 4

[alsa-devel] Fwd: Possible bug in sound/pci/atiixp.c ?
by Amir Shamsuddin 14 Mar '11

14 Mar '11

Hi, My laptop occasionally fails to resume from suspend, and when it does, I see (only) this error message on screen: [nnn.nnn] atiixp: codec reset timeout I had a look at the code where this comes from, and while I'm not sure it's related to my resume problem, it does (to my untrained eyes) look like there is a small bug in the function snd_atiixp_aclink_reset in file sound/pci/atiixp.c: (lines 500 - 536 in latest git) static int snd_atiixp_aclink_reset(struct atiixp *chip) { int timeout; [snip] timeout = 10; while (! (atiixp_read(chip, CMD) & ATI_REG_CMD_ACLINK_ACTIVE)) { /* do a hard reset */ [snip] if (--timeout) { snd_printk(KERN_ERR "atiixp: codec reset timeout\n"); break; } } [snip] } Surely the timeout condition should be negated, so the timeout message and loop break is only hit when 'timeout' reaches zero? ie. if (! --timeout) { Or am I missing something? Amir

2 4

[alsa-devel] [PATCH - ctxfi driver 1/4] Fix: microphone boost was set at +12dB, not +20dB (like in Windows driver and in adc_conf structure declaration), some comments added
by pbruskispam＠op.pl 14 Mar '11

14 Mar '11

From: Przemyslaw Bruski <pbruskispam(a)op.pl> Signed-off-by: Przemyslaw Bruski <pbruskispam(a)op.pl> diff --git a/pci/ctxfi/cthw20k2.c b/pci/ctxfi/cthw20k2.c index b6b11bf..5364164 100644 --- a/pci/ctxfi/cthw20k2.c +++ b/pci/ctxfi/cthw20k2.c @@ -1307,10 +1307,10 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) set_field(&pllctl, PLLCTL_B, 0); if (48000 == rsr) { set_field(&pllctl, PLLCTL_FD, 16 - 2); - set_field(&pllctl, PLLCTL_RD, 1 - 1); + set_field(&pllctl, PLLCTL_RD, 1 - 1); /* 3000*16/1 = 48000 */ } else { /* 44100 */ set_field(&pllctl, PLLCTL_FD, 147 - 2); - set_field(&pllctl, PLLCTL_RD, 10 - 1); + set_field(&pllctl, PLLCTL_RD, 10 - 1); /* 3000*147/10 = 44100 */ } hw_write_20kx(hw, PLL_CTL, pllctl); mdelay(40); @@ -1740,6 +1740,10 @@ static int hw_is_adc_input_selected(struct hw *hw, enum ADCSRC type) return data; } +#define MIC_BOOST_0DB 0xCF +#define MIC_BOOST_STEPS_PER_DB 2 +#define MIC_BOOST_20DB (MIC_BOOST_0DB + 20 * MIC_BOOST_STEPS_PER_DB) + static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) { u32 data; @@ -1751,10 +1755,12 @@ static int hw_adc_input_select(struct hw *hw, enum ADCSRC type) hw_write_20kx(hw, GPIO_DATA, data); hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), MAKE_WM8775_DATA(0x101)); /* Mic-in */ - hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), - MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ - hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), - MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ + hw20k2_i2c_write(hw, + MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB), + MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */ + hw20k2_i2c_write(hw, + MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB), + MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */ break; case ADC_LINEIN: data &= ~(0x1 << 14); @@ -1827,10 +1833,12 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_ADCMC, 0x101), MAKE_WM8775_DATA(0x101)); /* Mic-in */ - hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCL, 0xE7), - MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ - hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_AADCR, 0xE7), - MAKE_WM8775_DATA(0xE7)); /* +12dB boost */ + hw20k2_i2c_write(hw, + MAKE_WM8775_ADDR(WM8775_AADCL, MIC_BOOST_20DB), + MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */ + hw20k2_i2c_write(hw, + MAKE_WM8775_ADDR(WM8775_AADCR, MIC_BOOST_20DB), + MAKE_WM8775_DATA(MIC_BOOST_20DB)); /* +20dB */ } else if (mux == 2) { /* Configures GPIO data to select Line-in */ data &= ~(0x1 << 14); -- 1.7.1

2 4

Re: [alsa-devel] [Device-drivers-devel] [PATCH] Add driver for Analog Devices ADAU1701 SigmaDSP
by Mark Brown 14 Mar '11

14 Mar '11

On Sun, Mar 13, 2011 at 10:23:19PM -0400, Cai, Cliff wrote: > I think I need to have a check of the requirement from customer. > Anyway,if I modified the driver according to the comments here, > Would you like to accept it currently or we need to wait until the missing patch > Has been merged in mainline? Please repost, there were quite a few issues raised with this revision even outside the firmware. For the purpose of getting things into mainline I'd recommend splitting out the functionaltiy which allows firmware to be loaded dynamically at runtime as a second patch.

1 0

Re: [alsa-devel] [PATCH 1/1] hdspmixer: New pixmaps
by Jaroslav Kysela 14 Mar '11

14 Mar '11

On Sun, 13 Mar 2011, Adrian Knoth wrote: > Reuben Martin has provided me with an updated pixmap set to replace > RME's 1998 style no longer in use by the corresponding Win32 tools. > > The Linux pixmaps are based on ardour2. Likewise hdspmixer, ardour2 is > GPLv2. > > Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> Applied. Thanks. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

1 0

[alsa-devel] [PATCH] alsamixer: fix build on uClibc
by Peter Korsgaard 14 Mar '11

14 Mar '11

exp10 is a glibc extension, which isn't supported on uClibc. Luckily, exp10() is trivial to compute based on exp(), so add a wrapper for the uClibc case. Signed-off-by: Peter Korsgaard <jacmet(a)sunsite.dk> --- alsamixer/volume_mapping.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 9cacad8..1c0d7c4 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -37,6 +37,11 @@ #include <stdbool.h> #include "volume_mapping.h" +#ifdef __UCLIBC__ +/* 10^x = 10^(log e^x) = (e^x)^log10 = e^(x * log 10) */ +#define exp10(x) (exp((x) * log(10))) +#endif /* __UCLIBC__ */ + #define MAX_LINEAR_DB_SCALE 24 static inline bool use_linear_dB_scale(long dBmin, long dBmax) -- 1.7.2.3

2 1

[alsa-devel] Help hunting down a regression
by Colin Guthrie 12 Mar '11

12 Mar '11

So you may know from my other thread that I'm trying to find a regression introduced in the alsa drivers sometime ago. I suspect that 2.6.35 kernels were OK but I'm trying to compile the drivers from that era with little success. I'm cloning alsa-driver and alsa-kernel, and while I can checkout v2.6.35 in alsa-kernel OK, it's nigh on impossible to checkout the corresponding alsa-driver. I can't help but feel I'm missing a trick here, so figured I'd ask first before banging my head against a wall for several hours. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

1 0

[PATCH] Ensure the destination of the symlink doesn't yet exists.
by Colin Guthrie 12 Mar '11

12 Mar '11

Similar to my last patch, the symlinking process can also fail due to the destination file already being present. This ensures we wipe the file first. --- include/sound/Makefile | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/sound/Makefile b/include/sound/Makefile index f992c01..0e09df4 100644 --- a/include/sound/Makefile +++ b/include/sound/Makefile @@ -16,6 +16,7 @@ version.h: ../version.h patch -p0 -i $< $@ %.h: + rm -f $@ ln -s $(INCSRC)/$@ .includes.tmp: -- 1.7.4.1 --------------020400050902000603050300--

1 0

[PATCH] Fix broken build due to cp+symlink behaviour
by Colin Guthrie 12 Mar '11

12 Mar '11

Sometimes the destination file is a symlink to the source file. If this happens cp may complain: cp: ./info.h' are the same file This patch just adds a safety rm -f before hand which wont do any harm. --- include/sound/Makefile | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/include/sound/Makefile b/include/sound/Makefile index 4f3c782..f992c01 100644 --- a/include/sound/Makefile +++ b/include/sound/Makefile @@ -27,6 +27,7 @@ version.h: ../version.h b=$$(basename $$s .h); \ if test -f $$b.patch; then \ echo "$$s: $$i $$b.patch" >> $@; \ + echo " rm -f \$$@" >> $@; \ echo " cp \$$(INCSRC)/\$$@ ." >> $@; \ echo " patch -p0 -i $$b.patch \$$@" >> $@; \ echo >> $@; \ -- 1.7.4.1 --------------060405030201020509050606--

1 0

[alsa-devel] [PATCH 0/2] prescript
by Alexey Fisher 12 Mar '11

12 Mar '11

Hall all, here is small patchset for usb devices with wrong volume resolution. First one add warning if too big resolution is found. Second adds quirk for one more webcam. For more info see patch a5c7d797dcce3be5e77cd6ea62cc4920ededc32b > Author: Alexey Fisher <bug-track(a)fisher-privat.net> > Date: Wed Jun 23 14:17:09 2010 +0200 > ALSA: usb-audio - Add volume resolution quirk for some Logitech webcams .... Regards, Alexey

3 4

[alsa-devel] [GIT PULL FOR 2.6.39] Media controller and OMAP3 ISP driver
by Laurent Pinchart 11 Mar '11

11 Mar '11

Hi Mauro, The following changes since commit 85e2efbb1db9a18d218006706d6e4fbeb0216213: Linux 2.6.38-rc5 (2011-02-15 19:23:45 -0800) are available in the git repository at: git://linuxtv.org/pinchartl/media.git media-0005-omap3isp Antti Koskipaa (1): v4l: v4l2_subdev userspace crop API David Cohen (1): omap3isp: Statistics Laurent Pinchart (36): v4l: Share code between video_usercopy and video_ioctl2 v4l: subdev: Don't require core operations v4l: subdev: Add device node support v4l: subdev: Uninline the v4l2_subdev_init function v4l: subdev: Control ioctls support media: Media device node support media: Media device media: Entities, pads and links media: Entity use count media: Media device information query media: Entities, pads and links enumeration media: Links setup media: Pipelines and media streams v4l: Add a media_device pointer to the v4l2_device structure v4l: Make video_device inherit from media_entity v4l: Make v4l2_subdev inherit from media_entity v4l: Move the media/v4l2-mediabus.h header to include/linux v4l: Replace enums with fixed-sized fields in public structure v4l: Rename V4L2_MBUS_FMT_GREY8_1X8 to V4L2_MBUS_FMT_Y8_1X8 v4l: Group media bus pixel codes by types and sort them alphabetically v4l: subdev: Add new file operations v4l: v4l2_subdev pad-level operations v4l: v4l2_subdev userspace format API - documentation binary files v4l: v4l2_subdev userspace format API v4l: v4l2_subdev userspace frame interval API v4l: subdev: Generic ioctl support v4l: Add subdev sensor g_skip_frames operation v4l: Add 8-bit YUYV on 16-bit bus and SGRBG10 media bus pixel codes v4l: Add remaining RAW10 patterns w DPCM pixel code variants v4l: Add missing 12 bits bayer media bus formats v4l: Add 12 bits bayer pixel formats omap3: Add function to register omap3isp platform device structure omap3isp: Video devices and buffers queue omap3isp: CCP2/CSI2 receivers omap3isp: CCDC, preview engine and resizer omap3isp: Kconfig and Makefile Sakari Ailus (3): v4l: subdev: Events support media: Entity graph traversal omap3isp: OMAP3 ISP core Sergio Aguirre (2): omap3: Remove unusued ISP CBUFF resource omap2: Fix camera resources for multiomap Stanimir Varbanov (1): v4l: Create v4l2 subdev file handle structure Tuukka Toivonen (1): ARM: OMAP3: Update Camera ISP definitions for OMAP3630 Documentation/ABI/testing/sysfs-bus-media | 6 + Documentation/DocBook/Makefile | 5 +- Documentation/DocBook/media-entities.tmpl | 50 + Documentation/DocBook/media.tmpl | 3 + Documentation/DocBook/v4l/bayer.pdf | Bin 0 -> 12116 bytes Documentation/DocBook/v4l/bayer.png | Bin 0 -> 9725 bytes Documentation/DocBook/v4l/dev-subdev.xml | 313 +++ Documentation/DocBook/v4l/media-controller.xml | 89 + Documentation/DocBook/v4l/media-func-close.xml | 59 + Documentation/DocBook/v4l/media-func-ioctl.xml | 116 + Documentation/DocBook/v4l/media-func-open.xml | 94 + .../DocBook/v4l/media-ioc-device-info.xml | 133 ++ .../DocBook/v4l/media-ioc-enum-entities.xml | 308 +++ Documentation/DocBook/v4l/media-ioc-enum-links.xml | 207 ++ Documentation/DocBook/v4l/media-ioc-setup-link.xml | 93 + Documentation/DocBook/v4l/pipeline.pdf | Bin 0 -> 20276 bytes Documentation/DocBook/v4l/pipeline.png | Bin 0 -> 12130 bytes Documentation/DocBook/v4l/subdev-formats.xml | 2467 ++++++++++++++++++++ Documentation/DocBook/v4l/v4l2.xml | 7 + Documentation/DocBook/v4l/vidioc-streamon.xml | 9 + .../v4l/vidioc-subdev-enum-frame-interval.xml | 152 ++ .../DocBook/v4l/vidioc-subdev-enum-frame-size.xml | 154 ++ .../DocBook/v4l/vidioc-subdev-enum-mbus-code.xml | 119 + Documentation/DocBook/v4l/vidioc-subdev-g-crop.xml | 155 ++ Documentation/DocBook/v4l/vidioc-subdev-g-fmt.xml | 180 ++ .../DocBook/v4l/vidioc-subdev-g-frame-interval.xml | 141 ++ Documentation/media-framework.txt | 353 +++ Documentation/video4linux/v4l2-framework.txt | 127 +- MAINTAINERS | 6 + arch/arm/mach-omap2/devices.c | 63 +- arch/arm/mach-omap2/devices.h | 19 + arch/arm/plat-omap/include/plat/omap34xx.h | 16 +- drivers/media/Kconfig | 22 + drivers/media/Makefile | 6 + drivers/media/media-device.c | 382 +++ drivers/media/media-devnode.c | 321 +++ drivers/media/media-entity.c | 536 +++++ drivers/media/video/Kconfig | 13 + drivers/media/video/Makefile | 4 +- drivers/media/video/mt9m001.c | 2 +- drivers/media/video/mt9v022.c | 4 +- drivers/media/video/omap3-isp/Makefile | 13 + drivers/media/video/omap3-isp/cfa_coef_table.h | 61 + drivers/media/video/omap3-isp/gamma_table.h | 90 + drivers/media/video/omap3-isp/isp.c | 2220 ++++++++++++++++++ drivers/media/video/omap3-isp/isp.h | 427 ++++ drivers/media/video/omap3-isp/ispccdc.c | 2268 ++++++++++++++++++ drivers/media/video/omap3-isp/ispccdc.h | 219 ++ drivers/media/video/omap3-isp/ispccp2.c | 1173 ++++++++++ drivers/media/video/omap3-isp/ispccp2.h | 98 + drivers/media/video/omap3-isp/ispcsi2.c | 1317 +++++++++++ drivers/media/video/omap3-isp/ispcsi2.h | 166 ++ drivers/media/video/omap3-isp/ispcsiphy.c | 247 ++ drivers/media/video/omap3-isp/ispcsiphy.h | 74 + drivers/media/video/omap3-isp/isph3a.h | 117 + drivers/media/video/omap3-isp/isph3a_aewb.c | 374 +++ drivers/media/video/omap3-isp/isph3a_af.c | 429 ++++ drivers/media/video/omap3-isp/isphist.c | 520 ++++ drivers/media/video/omap3-isp/isphist.h | 40 + drivers/media/video/omap3-isp/isppreview.c | 2113 +++++++++++++++++ drivers/media/video/omap3-isp/isppreview.h | 214 ++ drivers/media/video/omap3-isp/ispqueue.c | 1153 +++++++++ drivers/media/video/omap3-isp/ispqueue.h | 187 ++ drivers/media/video/omap3-isp/ispreg.h | 1589 +++++++++++++ drivers/media/video/omap3-isp/ispresizer.c | 1693 ++++++++++++++ drivers/media/video/omap3-isp/ispresizer.h | 147 ++ drivers/media/video/omap3-isp/ispstat.c | 1092 +++++++++ drivers/media/video/omap3-isp/ispstat.h | 169 ++ drivers/media/video/omap3-isp/ispvideo.c | 1264 ++++++++++ drivers/media/video/omap3-isp/ispvideo.h | 202 ++ drivers/media/video/omap3-isp/luma_enhance_table.h | 42 + drivers/media/video/omap3-isp/noise_filter_table.h | 30 + drivers/media/video/ov6650.c | 10 +- drivers/media/video/sh_mobile_csi2.c | 6 +- drivers/media/video/soc_mediabus.c | 2 +- drivers/media/video/v4l2-dev.c | 76 +- drivers/media/video/v4l2-device.c | 84 +- drivers/media/video/v4l2-ioctl.c | 216 +-- drivers/media/video/v4l2-subdev.c | 332 +++ include/linux/Kbuild | 4 + include/linux/media.h | 132 ++ include/linux/omap3isp.h | 646 +++++ include/linux/v4l2-mediabus.h | 108 + include/linux/v4l2-subdev.h | 141 ++ include/linux/videodev2.h | 4 + include/media/media-device.h | 95 + include/media/media-devnode.h | 97 + include/media/media-entity.h | 151 ++ include/media/soc_mediabus.h | 3 +- include/media/v4l2-dev.h | 25 +- include/media/v4l2-device.h | 10 + include/media/v4l2-ioctl.h | 3 + include/media/v4l2-mediabus.h | 61 +- include/media/v4l2-subdev.h | 111 +- 94 files changed, 28496 insertions(+), 303 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-media create mode 100644 Documentation/DocBook/v4l/bayer.pdf create mode 100644 Documentation/DocBook/v4l/bayer.png create mode 100644 Documentation/DocBook/v4l/dev-subdev.xml create mode 100644 Documentation/DocBook/v4l/media-controller.xml create mode 100644 Documentation/DocBook/v4l/media-func-close.xml create mode 100644 Documentation/DocBook/v4l/media-func-ioctl.xml create mode 100644 Documentation/DocBook/v4l/media-func-open.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-device-info.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-entities.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-enum-links.xml create mode 100644 Documentation/DocBook/v4l/media-ioc-setup-link.xml create mode 100644 Documentation/DocBook/v4l/pipeline.pdf create mode 100644 Documentation/DocBook/v4l/pipeline.png create mode 100644 Documentation/DocBook/v4l/subdev-formats.xml create mode 100644 Documentation/DocBook/v4l/vidioc-subdev-enum-frame- interval.xml create mode 100644 Documentation/DocBook/v4l/vidioc-subdev-enum-frame- size.xml create mode 100644 Documentation/DocBook/v4l/vidioc-subdev-enum-mbus-code.xml create mode 100644 Documentation/DocBook/v4l/vidioc-subdev-g-crop.xml create mode 100644 Documentation/DocBook/v4l/vidioc-subdev-g-fmt.xml create mode 100644 Documentation/DocBook/v4l/vidioc-subdev-g-frame- interval.xml create mode 100644 Documentation/media-framework.txt create mode 100644 arch/arm/mach-omap2/devices.h create mode 100644 drivers/media/media-device.c create mode 100644 drivers/media/media-devnode.c create mode 100644 drivers/media/media-entity.c create mode 100644 drivers/media/video/omap3-isp/Makefile create mode 100644 drivers/media/video/omap3-isp/cfa_coef_table.h create mode 100644 drivers/media/video/omap3-isp/gamma_table.h create mode 100644 drivers/media/video/omap3-isp/isp.c create mode 100644 drivers/media/video/omap3-isp/isp.h create mode 100644 drivers/media/video/omap3-isp/ispccdc.c create mode 100644 drivers/media/video/omap3-isp/ispccdc.h create mode 100644 drivers/media/video/omap3-isp/ispccp2.c create mode 100644 drivers/media/video/omap3-isp/ispccp2.h create mode 100644 drivers/media/video/omap3-isp/ispcsi2.c create mode 100644 drivers/media/video/omap3-isp/ispcsi2.h create mode 100644 drivers/media/video/omap3-isp/ispcsiphy.c create mode 100644 drivers/media/video/omap3-isp/ispcsiphy.h create mode 100644 drivers/media/video/omap3-isp/isph3a.h create mode 100644 drivers/media/video/omap3-isp/isph3a_aewb.c create mode 100644 drivers/media/video/omap3-isp/isph3a_af.c create mode 100644 drivers/media/video/omap3-isp/isphist.c create mode 100644 drivers/media/video/omap3-isp/isphist.h create mode 100644 drivers/media/video/omap3-isp/isppreview.c create mode 100644 drivers/media/video/omap3-isp/isppreview.h create mode 100644 drivers/media/video/omap3-isp/ispqueue.c create mode 100644 drivers/media/video/omap3-isp/ispqueue.h create mode 100644 drivers/media/video/omap3-isp/ispreg.h create mode 100644 drivers/media/video/omap3-isp/ispresizer.c create mode 100644 drivers/media/video/omap3-isp/ispresizer.h create mode 100644 drivers/media/video/omap3-isp/ispstat.c create mode 100644 drivers/media/video/omap3-isp/ispstat.h create mode 100644 drivers/media/video/omap3-isp/ispvideo.c create mode 100644 drivers/media/video/omap3-isp/ispvideo.h create mode 100644 drivers/media/video/omap3-isp/luma_enhance_table.h create mode 100644 drivers/media/video/omap3-isp/noise_filter_table.h create mode 100644 drivers/media/video/v4l2-subdev.c create mode 100644 include/linux/media.h create mode 100644 include/linux/omap3isp.h create mode 100644 include/linux/v4l2-mediabus.h create mode 100644 include/linux/v4l2-subdev.h create mode 100644 include/media/media-device.h create mode 100644 include/media/media-devnode.h create mode 100644 include/media/media-entity.h -- Regards, Laurent Pinchart

8 43

[alsa-devel] : Question cpu_dai->active & codec_dai->active
by Bensaid, Selma 11 Mar '11

11 Mar '11

Hi, Why the cpu_dai->active and codec_dai->active are not used to check if the corresponding shutdown or startup callbacks should be called or not? This could be useful for DAIs that handle capture and playback streams by avoiding: - perform 2 cpu_dai or codec_dai startup when opening both streams (playback and Capture) - close a cpu_dai or codec_dai for a stream (example playback) while the 2nd stream (playback) is still running Thanks, Selma. --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

2 1

[alsa-devel] [PATCH] intel8x0m: add 'm' as "suffix" to static functions
by Paul Bolle 11 Mar '11

11 Mar '11

Adding an 'm' will distinguish them from identical names in intel8x0.c. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- sound/pci/intel8x0m.c | 98 ++++++++++++++++++++++++------------------------ 1 files changed, 49 insertions(+), 49 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 9abff7b..e16b3e9 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -341,9 +341,9 @@ static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int co return -EBUSY; } -static void snd_intel8x0_codec_write(struct snd_ac97 *ac97, - unsigned short reg, - unsigned short val) +static void snd_intel8x0m_codec_write(struct snd_ac97 *ac97, + unsigned short reg, + unsigned short val) { struct intel8x0m *chip = ac97->private_data; @@ -354,8 +354,8 @@ static void snd_intel8x0_codec_write(struct snd_ac97 *ac97, iaputword(chip, reg + ac97->num * 0x80, val); } -static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, - unsigned short reg) +static unsigned short snd_intel8x0m_codec_read(struct snd_ac97 *ac97, + unsigned short reg) { struct intel8x0m *chip = ac97->private_data; unsigned short res; @@ -385,7 +385,7 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, /* * DMA I/O */ -static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev) +static void snd_intel8x0m_setup_periods(struct intel8x0m *chip, struct ichdev *ichdev) { int idx; u32 *bdbar = ichdev->bdbar; @@ -437,7 +437,7 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic * Interrupt handler */ -static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ichdev) +static inline void snd_intel8x0m_update(struct intel8x0m *chip, struct ichdev *ichdev) { unsigned long port = ichdev->reg_offset; int civ, i, step; @@ -489,7 +489,7 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); } -static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id) +static irqreturn_t snd_intel8x0m_interrupt(int irq, void *dev_id) { struct intel8x0m *chip = dev_id; struct ichdev *ichdev; @@ -512,7 +512,7 @@ static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id) for (i = 0; i < chip->bdbars_count; i++) { ichdev = &chip->ichd[i]; if (status & ichdev->int_sta_mask) - snd_intel8x0_update(chip, ichdev); + snd_intel8x0m_update(chip, ichdev); } /* ack them */ @@ -526,7 +526,7 @@ static irqreturn_t snd_intel8x0_interrupt(int irq, void *dev_id) * PCM part */ -static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int snd_intel8x0m_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct intel8x0m *chip = snd_pcm_substream_chip(substream); struct ichdev *ichdev = get_ichdev(substream); @@ -561,18 +561,18 @@ static int snd_intel8x0_pcm_trigger(struct snd_pcm_substream *substream, int cmd return 0; } -static int snd_intel8x0_hw_params(struct snd_pcm_substream *substream, +static int snd_intel8x0m_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } -static int snd_intel8x0_hw_free(struct snd_pcm_substream *substream) +static int snd_intel8x0m_hw_free(struct snd_pcm_substream *substream) { return snd_pcm_lib_free_pages(substream); } -static snd_pcm_uframes_t snd_intel8x0_pcm_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t snd_intel8x0m_pcm_pointer(struct snd_pcm_substream *substream) { struct intel8x0m *chip = snd_pcm_substream_chip(substream); struct ichdev *ichdev = get_ichdev(substream); @@ -600,7 +600,7 @@ static int snd_intel8x0m_pcm_prepare(struct snd_pcm_substream *substream) ichdev->fragsize = snd_pcm_lib_period_bytes(substream); snd_ac97_write(ichdev->ac97, AC97_LINE1_RATE, runtime->rate); snd_ac97_write(ichdev->ac97, AC97_LINE1_LEVEL, 0); - snd_intel8x0_setup_periods(chip, ichdev); + snd_intel8x0m_setup_periods(chip, ichdev); return 0; } @@ -682,22 +682,22 @@ static struct snd_pcm_ops snd_intel8x0m_playback_ops = { .open = snd_intel8x0m_playback_open, .close = snd_intel8x0m_playback_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_intel8x0_hw_params, - .hw_free = snd_intel8x0_hw_free, + .hw_params = snd_intel8x0m_hw_params, + .hw_free = snd_intel8x0m_hw_free, .prepare = snd_intel8x0m_pcm_prepare, - .trigger = snd_intel8x0_pcm_trigger, - .pointer = snd_intel8x0_pcm_pointer, + .trigger = snd_intel8x0m_pcm_trigger, + .pointer = snd_intel8x0m_pcm_pointer, }; static struct snd_pcm_ops snd_intel8x0m_capture_ops = { .open = snd_intel8x0m_capture_open, .close = snd_intel8x0m_capture_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_intel8x0_hw_params, - .hw_free = snd_intel8x0_hw_free, + .hw_params = snd_intel8x0m_hw_params, + .hw_free = snd_intel8x0m_hw_free, .prepare = snd_intel8x0m_pcm_prepare, - .trigger = snd_intel8x0_pcm_trigger, - .pointer = snd_intel8x0_pcm_pointer, + .trigger = snd_intel8x0m_pcm_trigger, + .pointer = snd_intel8x0m_pcm_pointer, }; @@ -710,7 +710,7 @@ struct ich_pcm_table { int ac97_idx; }; -static int __devinit snd_intel8x0_pcm1(struct intel8x0m *chip, int device, +static int __devinit snd_intel8x0m_pcm1(struct intel8x0m *chip, int device, struct ich_pcm_table *rec) { struct snd_pcm *pcm; @@ -759,7 +759,7 @@ static struct ich_pcm_table intel_pcms[] __devinitdata = { }, }; -static int __devinit snd_intel8x0_pcm(struct intel8x0m *chip) +static int __devinit snd_intel8x0m_pcm(struct intel8x0m *chip) { int i, tblsize, device, err; struct ich_pcm_table *tbl, *rec; @@ -791,7 +791,7 @@ static int __devinit snd_intel8x0_pcm(struct intel8x0m *chip) if (! chip->ichd[rec->ac97_idx].ac97) continue; } - err = snd_intel8x0_pcm1(chip, device, rec); + err = snd_intel8x0m_pcm1(chip, device, rec); if (err < 0) return err; device++; @@ -806,20 +806,20 @@ static int __devinit snd_intel8x0_pcm(struct intel8x0m *chip) * Mixer part */ -static void snd_intel8x0_mixer_free_ac97_bus(struct snd_ac97_bus *bus) +static void snd_intel8x0m_mixer_free_ac97_bus(struct snd_ac97_bus *bus) { struct intel8x0m *chip = bus->private_data; chip->ac97_bus = NULL; } -static void snd_intel8x0_mixer_free_ac97(struct snd_ac97 *ac97) +static void snd_intel8x0m_mixer_free_ac97(struct snd_ac97 *ac97) { struct intel8x0m *chip = ac97->private_data; chip->ac97 = NULL; } -static int __devinit snd_intel8x0_mixer(struct intel8x0m *chip, int ac97_clock) +static int __devinit snd_intel8x0m_mixer(struct intel8x0m *chip, int ac97_clock) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; @@ -827,22 +827,22 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0m *chip, int ac97_clock) int err; unsigned int glob_sta = 0; static struct snd_ac97_bus_ops ops = { - .write = snd_intel8x0_codec_write, - .read = snd_intel8x0_codec_read, + .write = snd_intel8x0m_codec_write, + .read = snd_intel8x0m_codec_read, }; chip->in_ac97_init = 1; memset(&ac97, 0, sizeof(ac97)); ac97.private_data = chip; - ac97.private_free = snd_intel8x0_mixer_free_ac97; + ac97.private_free = snd_intel8x0m_mixer_free_ac97; ac97.scaps = AC97_SCAP_SKIP_AUDIO | AC97_SCAP_POWER_SAVE; glob_sta = igetdword(chip, ICHREG(GLOB_STA)); if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &pbus)) < 0) goto __err; - pbus->private_free = snd_intel8x0_mixer_free_ac97_bus; + pbus->private_free = snd_intel8x0m_mixer_free_ac97_bus; if (ac97_clock >= 8000 && ac97_clock <= 48000) pbus->clock = ac97_clock; chip->ac97_bus = pbus; @@ -959,7 +959,7 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) return 0; } -static int snd_intel8x0_chip_init(struct intel8x0m *chip, int probing) +static int snd_intel8x0m_chip_init(struct intel8x0m *chip, int probing) { unsigned int i; int err; @@ -980,7 +980,7 @@ static int snd_intel8x0_chip_init(struct intel8x0m *chip, int probing) return 0; } -static int snd_intel8x0_free(struct intel8x0m *chip) +static int snd_intel8x0m_free(struct intel8x0m *chip) { unsigned int i; @@ -1045,7 +1045,7 @@ static int intel8x0m_resume(struct pci_dev *pci) return -EIO; } pci_set_master(pci); - if (request_irq(pci->irq, snd_intel8x0_interrupt, + if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, card->shortname, chip)) { printk(KERN_ERR "intel8x0m: unable to grab IRQ %d, " "disabling device\n", pci->irq); @@ -1053,7 +1053,7 @@ static int intel8x0m_resume(struct pci_dev *pci) return -EIO; } chip->irq = pci->irq; - snd_intel8x0_chip_init(chip, 0); + snd_intel8x0m_chip_init(chip, 0); snd_ac97_resume(chip->ac97); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -1094,10 +1094,10 @@ static void __devinit snd_intel8x0m_proc_init(struct intel8x0m * chip) #endif /* CONFIG_PROC_FS */ -static int snd_intel8x0_dev_free(struct snd_device *device) +static int snd_intel8x0m_dev_free(struct snd_device *device) { struct intel8x0m *chip = device->device_data; - return snd_intel8x0_free(chip); + return snd_intel8x0m_free(chip); } struct ich_reg_info { @@ -1116,7 +1116,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, unsigned int int_sta_masks; struct ichdev *ichdev; static struct snd_device_ops ops = { - .dev_free = snd_intel8x0_dev_free, + .dev_free = snd_intel8x0m_dev_free, }; static struct ich_reg_info intel_regs[2] = { { ICH_MIINT, 0 }, @@ -1158,7 +1158,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, chip->addr = pci_iomap(pci, 0, 0); if (!chip->addr) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); - snd_intel8x0_free(chip); + snd_intel8x0m_free(chip); return -EIO; } if (pci_resource_flags(pci, 3) & IORESOURCE_MEM) /* ICH4 */ @@ -1167,15 +1167,15 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, chip->bmaddr = pci_iomap(pci, 1, 0); if (!chip->bmaddr) { snd_printk(KERN_ERR "Controller space ioremap problem\n"); - snd_intel8x0_free(chip); + snd_intel8x0m_free(chip); return -EIO; } port_inited: - if (request_irq(pci->irq, snd_intel8x0_interrupt, IRQF_SHARED, + if (request_irq(pci->irq, snd_intel8x0m_interrupt, IRQF_SHARED, card->shortname, chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); - snd_intel8x0_free(chip); + snd_intel8x0m_free(chip); return -EBUSY; } chip->irq = pci->irq; @@ -1210,7 +1210,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), chip->bdbars_count * sizeof(u32) * ICH_MAX_FRAGS * 2, &chip->bdbars) < 0) { - snd_intel8x0_free(chip); + snd_intel8x0m_free(chip); return -ENOMEM; } /* tables must be aligned to 8 bytes here, but the kernel pages @@ -1225,13 +1225,13 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, chip->int_sta_reg = ICH_REG_GLOB_STA; chip->int_sta_mask = int_sta_masks; - if ((err = snd_intel8x0_chip_init(chip, 1)) < 0) { - snd_intel8x0_free(chip); + if ((err = snd_intel8x0m_chip_init(chip, 1)) < 0) { + snd_intel8x0m_free(chip); return err; } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_intel8x0_free(chip); + snd_intel8x0m_free(chip); return err; } @@ -1295,11 +1295,11 @@ static int __devinit snd_intel8x0m_probe(struct pci_dev *pci, } card->private_data = chip; - if ((err = snd_intel8x0_mixer(chip, ac97_clock)) < 0) { + if ((err = snd_intel8x0m_mixer(chip, ac97_clock)) < 0) { snd_card_free(card); return err; } - if ((err = snd_intel8x0_pcm(chip)) < 0) { + if ((err = snd_intel8x0m_pcm(chip)) < 0) { snd_card_free(card); return err; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH v3] intel8x0m: wait a bit before warm reset check
by Paul Bolle 11 Mar '11

11 Mar '11

At every resume a laptop I use prints this message (at KERN_ERR level): ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2] The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what seems to be happening is that the register involved indicated a warm reset for some time (as the ICH_AC97WARM bit was set) but by the time the warning is printed, and that same register is checked again, that bit is already cleared and only the ICH_AC97COLD bit is still set. It turns out a warm reset needs some time to settle, but it is currently checked right away. The test therefore fails the first time it is done and schedule_timeout_uninterruptible() will be called. Once we return from that jiffies is already (far) past end_time on this laptop, so we exit the loop, print a warning, and exit the function while the warm reset actually succeeded. A way to fix this is to call usleep_range() after writing to the register involved. A handful of tests suggest 500 usecs is a safe value. (This might punish the "finish cold reset" case, but on this laptop such a cold reset apparently never happens, so I can't say for sure.) While we're at it drop the extra single tick from end_time, as it looks rather silly. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- changes in v3: - add comment - use usleep_range() sound/pci/intel8x0m.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 13cec1e..b1c926c 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -894,7 +894,8 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) /* finish cold or do warm reset */ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; iputdword(chip, ICHREG(GLOB_CNT), cnt); - end_time = (jiffies + (HZ / 4)) + 1; + usleep_range(500, 1000); /* give warm reset some time */ + end_time = jiffies + HZ / 4; do { if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0) goto __ok; -- 1.7.4.1

2 1

[alsa-devel] [PATCH 0/2] usbaudio:runtime power management
by Oliver Neukum 11 Mar '11

11 Mar '11

Hi, this patch set implements runtime power management for USB audio devices. It uses a coneventional opened == active, close == idle approach with respect to playing and recording sound, but allows devices with opened mixer interfaces to be suspended. Regards Oliver

2 1

[alsa-devel] question on snd_power_change_state()
by Oliver Neukum 11 Mar '11

11 Mar '11

Hi, will calling snd_power_change_state() reset all mixer settings? Regards Oliver

3 7

[alsa-devel] [PATCH 2/2] usbaudio: implement USB autosuspend
by Oliver Neukum 11 Mar '11

11 Mar '11

From 42c15a1770f3dbcdf4ed34722d49ffaeea0c9c19 Mon Sep 17 00:00:00 2001 From: Oliver Neukum <oliver(a)neukum.org> Date: Fri, 11 Mar 2011 11:44:33 +0100 Subject: [PATCH 2/2] usbaudio: implement USB autosuspend Devices are autosuspended if no pcm nor midi channel is open Mixer devices may be opened. This way they are active when in use to play or record sound, but can be suspended while users have a mixer application running. Signed-off-by: Oliver Neukum <oneukum(a)suse.de> --- sound/usb/card.c | 76 ++++++++++++++++++++++++++++++++++++++++--------- sound/usb/midi.c | 8 +++++ sound/usb/mixer.c | 23 ++++++++++++++- sound/usb/pcm.c | 20 ++++++++++--- sound/usb/power.h | 2 + sound/usb/usbaudio.h | 6 +++- 6 files changed, 113 insertions(+), 22 deletions(-) create mode 100644 sound/usb/power.h diff --git a/sound/usb/card.c b/sound/usb/card.c index af76f97..ef33920 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -65,6 +65,7 @@ #include "pcm.h" #include "urb.h" #include "format.h" +#include "power.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -330,6 +331,7 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx, chip->setup = device_setup[idx]; chip->nrpacks = nrpacks; chip->async_unlink = async_unlink; + chip->probing = 1; chip->usb_id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); @@ -451,6 +453,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } chip = usb_chip[i]; + chip->probing = 1; break; } } @@ -466,6 +469,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } snd_card_set_dev(chip->card, &intf->dev); + chip->pm_intf = intf; break; } if (!chip) { @@ -505,6 +509,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, usb_chip[chip->index] = chip; chip->num_interfaces++; + chip->probing = 0; mutex_unlock(&register_mutex); return chip; @@ -581,6 +586,23 @@ static void usb_audio_disconnect(struct usb_interface *intf) } #ifdef CONFIG_PM + +int snd_usb_autoresume(struct snd_usb_audio *chip) +{ + int err = -ENODEV; + + if (!chip->shutdown && !chip->probing) + err = usb_autopm_get_interface(chip->pm_intf); + + return err; +} + +void snd_usb_autosuspend(struct snd_usb_audio *chip) +{ + if (!chip->shutdown && !chip->probing) + usb_autopm_put_interface(chip->pm_intf); +} + static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) { struct snd_usb_audio *chip = usb_get_intfdata(intf); @@ -591,18 +613,26 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (chip == (void *)-1L) return 0; - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); - if (!chip->num_suspended_intf++) { - list_for_each(p, &chip->pcm_list) { - as = list_entry(p, struct snd_usb_stream, list); - snd_pcm_suspend_all(as->pcm); - } - - list_for_each_entry(mixer, &chip->mixer_list, list) { - snd_usb_mixer_inactivate(mixer); - } + if (!(message.event & PM_EVENT_AUTO)) { + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + if (!chip->num_suspended_intf++) { + list_for_each(p, &chip->pcm_list) { + as = list_entry(p, struct snd_usb_stream, list); + snd_pcm_suspend_all(as->pcm); + } + } + } else { + /* + * otherwise we keep the rest of the system in the dark + * to keep this transparent + */ + if (!chip->num_suspended_intf++) + chip->autosuspended = 1; } + list_for_each_entry(mixer, &chip->mixer_list, list) + snd_usb_mixer_inactivate(mixer); + return 0; } @@ -610,6 +640,7 @@ static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct usb_mixer_interface *mixer; + int err = 0; if (chip == (void *)-1L) return 0; @@ -619,16 +650,32 @@ static int usb_audio_resume(struct usb_interface *intf) * ALSA leaves material resumption to user space * we just notify and restart the mixers */ - list_for_each_entry(mixer, &chip->mixer_list, list) - snd_usb_mixer_activate(mixer); + list_for_each_entry(mixer, &chip->mixer_list, list) { + err = snd_usb_mixer_activate(mixer); + if (err < 0) + goto err_out; + } - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); + if (!chip->autosuspended) + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); + chip->autosuspended = 0; - return 0; +err_out: + return err; } #else #define usb_audio_suspend NULL #define usb_audio_resume NULL + +int snd_usb_autoresume(struct snd_usb_audio *chip) +{ + return 0; +} + +void snd_usb_autosuspend(struct snd_usb_audio *chip) +{ + return; +} #endif /* CONFIG_PM */ static struct usb_device_id usb_audio_ids [] = { @@ -652,6 +699,7 @@ static struct usb_driver usb_audio_driver = { .suspend = usb_audio_suspend, .resume = usb_audio_resume, .id_table = usb_audio_ids, + .supports_autosuspend = 1, }; static int __init snd_usb_audio_init(void) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index db2dc5f..b4b39c0 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -54,6 +54,7 @@ #include <sound/asequencer.h> #include "usbaudio.h" #include "midi.h" +#include "power.h" #include "helper.h" /* @@ -1044,6 +1045,7 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) struct snd_usb_midi* umidi = substream->rmidi->private_data; struct usbmidi_out_port* port = NULL; int i, j; + int err; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) if (umidi->endpoints[i].out) @@ -1056,6 +1058,9 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) snd_BUG(); return -ENXIO; } + err = usb_autopm_get_interface(umidi->iface); + if (err < 0) + return -EIO; substream->runtime->private_data = port; port->state = STATE_UNKNOWN; substream_open(substream, 1); @@ -1064,7 +1069,10 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + struct snd_usb_midi* umidi = substream->rmidi->private_data; + substream_open(substream, 0); + usb_autopm_put_interface(umidi->iface); return 0; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index da65130..168c4fa 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -61,6 +61,7 @@ #include "mixer.h" #include "helper.h" #include "mixer_quirks.h" +#include "power.h" #define MAX_ID_ELEMS 256 @@ -295,16 +296,22 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v unsigned char buf[2]; int val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; int timeout = 10; + int err; + err = snd_usb_autoresume(cval->mixer->chip); + if (err < 0) + return -EIO; while (timeout-- > 0) { if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), buf, val_len, 100) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); + snd_usb_autosuspend(cval->mixer->chip); return 0; } } + snd_usb_autosuspend(cval->mixer->chip); snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type); return -EINVAL; @@ -328,12 +335,18 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v memset(buf, 0, sizeof(buf)); + ret = snd_usb_autoresume(chip) ? -EIO : 0; + if (ret) + goto error; + ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), bRequest, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), buf, size, 1000); + snd_usb_autosuspend(chip); if (ret < 0) { +error: snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n", request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type); return ret; @@ -413,7 +426,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, { struct snd_usb_audio *chip = cval->mixer->chip; unsigned char buf[2]; - int val_len, timeout = 10; + int val_len, err, timeout = 10; if (cval->mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; @@ -433,13 +446,19 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + err = snd_usb_autoresume(chip); + if (err < 0) + return -EIO; while (timeout-- > 0) if (snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), - buf, val_len, 100) >= 0) + buf, val_len, 100) >= 0) { + snd_usb_autosuspend(chip); return 0; + } + snd_usb_autosuspend(chip); snd_printdd(KERN_ERR "cannot set ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d, data = %#x/%#x\n", request, validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), cval->val_type, buf[0], buf[1]); return -EINVAL; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e3f6805..b8dcbf4 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -32,6 +32,7 @@ #include "helper.h" #include "pcm.h" #include "clock.h" +#include "power.h" /* * return the current pcm pointer. just based on the hwptr_done value. @@ -739,6 +740,9 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre pt = 125 * (1 << fp->datainterval); ptmin = min(ptmin, pt); } + err = snd_usb_autoresume(subs->stream->chip); + if (err < 0) + return err; param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) @@ -756,21 +760,21 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, -1)) < 0) - return err; + goto rep_err; if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, hw_rule_channels, subs, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_RATE, param_period_time_if_needed, -1)) < 0) - return err; + goto rep_err; if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, hw_rule_format, subs, SNDRV_PCM_HW_PARAM_RATE, SNDRV_PCM_HW_PARAM_CHANNELS, param_period_time_if_needed, -1)) < 0) - return err; + goto rep_err; if (param_period_time_if_needed >= 0) { err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, @@ -780,11 +784,15 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) - return err; + goto rep_err; } if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) - return err; + goto rep_err; return 0; + +rep_err: + snd_usb_autosuspend(subs->stream->chip); + return err; } static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) @@ -798,6 +806,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) runtime->hw = snd_usb_hardware; runtime->private_data = subs; subs->pcm_substream = substream; + /* runtime PM is also done there */ return setup_hw_info(runtime, subs); } @@ -811,6 +820,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) subs->interface = -1; } subs->pcm_substream = NULL; + snd_usb_autosuspend(subs->stream->chip); return 0; } diff --git a/sound/usb/power.h b/sound/usb/power.h new file mode 100644 index 0000000..d2e1a6e --- /dev/null +++ b/sound/usb/power.h @@ -0,0 +1,2 @@ +int snd_usb_autoresume(struct snd_usb_audio *chip); +void snd_usb_autosuspend(struct snd_usb_audio *chip); diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 6e66fff..32f2a97 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -34,10 +34,14 @@ struct snd_usb_audio { int index; struct usb_device *dev; struct snd_card *card; + struct usb_interface *pm_intf; u32 usb_id; - int shutdown; struct mutex shutdown_mutex; + unsigned int shutdown:1; + unsigned int probing:1; + unsigned int autosuspended:1; unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ + int num_interfaces; int num_suspended_intf; -- 1.7.1

1 0

[alsa-devel] [PATCH 1/2] usbaudio:fix suspend/resume
by Oliver Neukum 11 Mar '11

11 Mar '11

From 0926f3aa059c42ccf096184654a9fb859c75f2e5 Mon Sep 17 00:00:00 2001 From: Oliver Neukum <oliver(a)neukum.org> Date: Fri, 11 Mar 2011 10:07:52 +0100 Subject: [PATCH 1/2] usbaudio:fix suspend/resume - ESHUTDOWN must be correctly handled - the optional interrupt endpoint's URB must be stopped and restarted Signed-off-by: Oliver Neukum <oneukum(a)suse.de> --- sound/usb/card.c | 12 ++++++++++-- sound/usb/mixer.c | 26 ++++++++++++++++++++++++-- sound/usb/mixer.h | 2 ++ 3 files changed, 36 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index c0f8270..af76f97 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -586,6 +586,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) struct snd_usb_audio *chip = usb_get_intfdata(intf); struct list_head *p; struct snd_usb_stream *as; + struct usb_mixer_interface *mixer; if (chip == (void *)-1L) return 0; @@ -596,6 +597,10 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) as = list_entry(p, struct snd_usb_stream, list); snd_pcm_suspend_all(as->pcm); } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + snd_usb_mixer_inactivate(mixer); + } } return 0; @@ -604,6 +609,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); + struct usb_mixer_interface *mixer; if (chip == (void *)-1L) return 0; @@ -611,9 +617,11 @@ static int usb_audio_resume(struct usb_interface *intf) return 0; /* * ALSA leaves material resumption to user space - * we just notify + * we just notify and restart the mixers */ - + list_for_each_entry(mixer, &chip->mixer_list, list) + snd_usb_mixer_activate(mixer); + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); return 0; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 85af605..da65130 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2058,8 +2058,9 @@ static void snd_usb_mixer_interrupt(struct urb *urb) { struct usb_mixer_interface *mixer = urb->context; int len = urb->actual_length; + int ustatus = urb->status; - if (urb->status != 0) + if (ustatus != 0) goto requeue; if (mixer->protocol == UAC_VERSION_1) { @@ -2100,12 +2101,33 @@ static void snd_usb_mixer_interrupt(struct urb *urb) } requeue: - if (urb->status != -ENOENT && urb->status != -ECONNRESET) { + if (ustatus != -ENOENT && ustatus != -ECONNRESET && ustatus != -ESHUTDOWN) { urb->dev = mixer->chip->dev; usb_submit_urb(urb, GFP_ATOMIC); } } +/* stop any bus activity of a mixer */ +void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer) +{ + usb_kill_urb(mixer->urb); + usb_kill_urb(mixer->rc_urb); +} + + +int snd_usb_mixer_activate(struct usb_mixer_interface *mixer) +{ + int err; + + if (mixer->urb) { + err = usb_submit_urb(mixer->urb, GFP_NOIO); + if (err < 0) + return err; + } + + return 0; +} + /* create the handler for the optional status interrupt endpoint */ static int snd_usb_mixer_status_create(struct usb_mixer_interface *mixer) { diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 26c636c..b4a2c81 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -52,5 +52,7 @@ void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, int unitid); int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set); +void snd_usb_mixer_inactivate(struct usb_mixer_interface *mixer); +int snd_usb_mixer_activate(struct usb_mixer_interface *mixer); #endif /* __USBMIXER_H */ -- 1.7.1

1 0

[alsa-devel] [PATCH] ASoC: Add MAX9850 codec driver
by Christian Glindkamp 11 Mar '11

11 Mar '11

This patch adds ASoC support for the MAX9850 codec with headphone amplifier. Supported features: - Playback - 16, 20 and 24 bit audio - 8k - 48k sample rates - DAPM Only 16 bit audio was tested while the codec was connected to an AT91SAM9G20 SSC in master mode. Signed-off-by: Christian Glindkamp <christian.glindkamp(a)taskit.de> --- I've all ready sent this patch some time ago, but it hung in the moderation queue. This is a slightly modified version. Unfortunately I do not have the hardware anymore to test suggested changes that alter function. sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9850.c | 358 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9850.h | 41 +++++ 4 files changed, 405 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/max9850.c create mode 100644 sound/soc/codecs/max9850.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e239345..51e9844 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C + select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -179,6 +180,9 @@ config SND_SOC_DMIC config SND_SOC_MAX98088 tristate +config SND_SOC_MAX9850 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ae10507..f2efd1c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o +snd-soc-max9850-objs := max9850.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o @@ -102,6 +103,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o +obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c new file mode 100644 index 0000000..a8c1f95 --- /dev/null +++ b/sound/soc/codecs/max9850.c @@ -0,0 +1,358 @@ +/* + * max9850.c -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * + * Author: Christian Glindkamp <christian.glindkamp(a)taskit.de> + * + * Initial development of this code was funded by + * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/ + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "max9850.h" + +struct max9850_priv { + unsigned int sysclk; +}; + +/* max9850 register cache */ +static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { + 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +}; + +/* these registers are not used at the moment but provided for the sake of + * completeness */ +static int max9850_volatile_register(unsigned int reg) +{ + switch (reg) { + case MAX9850_STATUSA: + case MAX9850_STATUSB: + return 1; + default: + return 0; + } +} + +static const unsigned int max9850_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), + 0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0), + 0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0), + 0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0), +}; + +static const struct snd_kcontrol_new max9850_controls[] = { +SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv), +SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1), +SOC_SINGLE("Mono", MAX9850_GENERAL_PURPOSE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new max9850_mixer_controls[] = { + SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0), +}; + +static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0), +SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0), +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_INPUT("INL"), +SND_SOC_DAPM_INPUT("INR"), +SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0, + &max9850_mixer_controls[0], + ARRAY_SIZE(max9850_mixer_controls)), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* output mixer */ + {"Output Mixer", NULL, "DAC"}, + {"Output Mixer", "Line In Switch", "Line Input"}, + + /* outputs */ + {"Headphone Output", NULL, "Output Mixer"}, + {"HPL", NULL, "Headphone Output"}, + {"HPR", NULL, "Headphone Output"}, + {"OUTL", NULL, "Output Mixer"}, + {"OUTR", NULL, "Output Mixer"}, + + /* inputs */ + {"Line Input", NULL, "INL"}, + {"Line Input", NULL, "INR"}, + + /* supplies */ + {"DAC", NULL, "MCLK"}, +}; + +static int max9850_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + u64 lrclk_div; + u8 sf, da; + + /* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */ + sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1; + lrclk_div = (1 << 22); + lrclk_div *= params_rate(params); + lrclk_div *= sf; + do_div(lrclk_div, max9850->sysclk); + + snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f); + snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff); + + da = snd_soc_read(codec, MAX9850_DIGITAL_AUDIO); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + da |= 0x2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + da |= 0x3; + break; + default: + return -EINVAL; + } + snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da); + + return 0; +} + +static int max9850_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); + + /* calculate mclk -> iclk divider */ + if (freq <= 13000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x0); + else if (freq <= 26000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x4); + else if (freq <= 40000000) + snd_soc_write(codec, MAX9850_CLOCK, 0x8); + else + return -EINVAL; + + max9850->sysclk = freq; + return 0; +} + +static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 da = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + da |= MAX9850_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + da |= MAX9850_DLY; + break; + case SND_SOC_DAIFMT_RIGHT_J: + da |= MAX9850_RTJ; + break; + case SND_SOC_DAIFMT_LEFT_J: + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + da |= MAX9850_BCINV | MAX9850_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + da |= MAX9850_BCINV; + break; + case SND_SOC_DAIFMT_NB_IF: + da |= MAX9850_INV; + break; + default: + return -EINVAL; + } + + /* set da */ + snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da); + + return 0; +} + +static int max9850_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, MAX9850_ENABLE, MAX9850_SHDN, + MAX9850_SHDN); + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, MAX9850_ENABLE, MAX9850_SHDN, 0); + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->dapm.bias_level = level; + return 0; +} + +#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000 + +#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops max9850_dai_ops = { + .hw_params = max9850_hw_params, + .set_sysclk = max9850_set_dai_sysclk, + .set_fmt = max9850_set_dai_fmt, +}; + +static struct snd_soc_dai_driver max9850_dai = { + .name = "max9850-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = MAX9850_RATES, + .formats = MAX9850_FORMATS + }, + .ops = &max9850_dai_ops, +}; + +static int max9850_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + /* enable zero-detect */ + snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1); + /* enable charge pump, disable everything else */ + snd_soc_write(codec, MAX9850_ENABLE, 0x30); + /* enable slew-rate control */ + snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40); + /* set slew-rate 125ms */ + snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0); + + snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets, + ARRAY_SIZE(max9850_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); + + snd_soc_add_controls(codec, max9850_controls, + ARRAY_SIZE(max9850_controls)); + + return 0; +} +static int max9850_remove(struct snd_soc_codec *codec) +{ + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max9850 = { + .probe = max9850_probe, + .remove = max9850_remove, + .set_bias_level = max9850_set_bias_level, + .reg_cache_size = ARRAY_SIZE(max9850_reg), + .reg_word_size = sizeof(u8), + .reg_cache_default = max9850_reg, + .volatile_register = max9850_volatile_register, +}; + +static int __devinit max9850_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max9850_priv *max9850; + int ret; + + max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL); + if (max9850 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, max9850); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_max9850, &max9850_dai, 1); + if (ret < 0) + kfree(max9850); + return ret; +} + +static __devexit int max9850_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max9850_i2c_id[] = { + { "max9850", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9850_i2c_id); + +static struct i2c_driver max9850_i2c_driver = { + .driver = { + .name = "max9850-codec", + .owner = THIS_MODULE, + }, + .probe = max9850_i2c_probe, + .remove = __devexit_p(max9850_i2c_remove), + .id_table = max9850_i2c_id, +}; + +static int __init max9850_init(void) +{ + return i2c_add_driver(&max9850_i2c_driver); +} +module_init(max9850_init); + +static void __exit max9850_exit(void) +{ + i2c_del_driver(&max9850_i2c_driver); +} +module_exit(max9850_exit); + +MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp(a)taskit.de>"); +MODULE_DESCRIPTION("ASoC MAX9850 codec driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h new file mode 100644 index 0000000..5268575 --- /dev/null +++ b/sound/soc/codecs/max9850.h @@ -0,0 +1,41 @@ +/* + * max9850.h -- codec driver for max9850 + * + * Copyright (C) 2011 taskit GmbH + * Author: Christian Glindkamp <christian.glindkamp(a)taskit.de> + * + * 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. + * + */ + +#ifndef _MAX9850_H +#define _MAX9850_H + +#define MAX9850_STATUSA 0x00 +#define MAX9850_STATUSB 0x01 +#define MAX9850_VOLUME 0x02 +#define MAX9850_GENERAL_PURPOSE 0x03 +#define MAX9850_INTERRUPT 0x04 +#define MAX9850_ENABLE 0x05 +#define MAX9850_CLOCK 0x06 +#define MAX9850_CHARGE_PUMP 0x07 +#define MAX9850_LRCLK_MSB 0x08 +#define MAX9850_LRCLK_LSB 0x09 +#define MAX9850_DIGITAL_AUDIO 0x0a + +#define MAX9850_CACHEREGNUM 11 + +/* MAX9850_ENABLE */ +#define MAX9850_SHDN (1<<7) + +/* MAX9850_DIGITAL_AUDIO */ +#define MAX9850_MASTER (1<<7) +#define MAX9850_INV (1<<6) +#define MAX9850_BCINV (1<<5) +#define MAX9850_DLY (1<<3) +#define MAX9850_RTJ (1<<2) + +#endif -- 1.7.2.3

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[alsa-devel] [PATCH 1/2] ASoC: PXA: z2: Mute internal speaker when headphones are connected
by Vasily Khoruzhick 11 Mar '11

11 Mar '11

Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/pxa/z2.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c index 3ceaef6..b0b6c8d 100644 --- a/sound/soc/pxa/z2.c +++ b/sound/soc/pxa/z2.c @@ -95,6 +95,11 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE, }, + { + .pin = "Ext Spk", + .mask = SND_JACK_HEADPHONE, + .invert = 1 + }, }; /* Headset jack detection gpios */ -- 1.7.4.1

3 3

Re: [alsa-devel] [Device-drivers-devel] [PATCH] Add driver for Analog Devices ADAU1701 SigmaDSP
by Mark Brown 11 Mar '11

11 Mar '11

On Fri, Mar 11, 2011 at 02:54:42AM -0500, Cai, Cliff wrote: > 1.The default firmware is used when the chip is just powered-up, > Once driver gets the control,it should load the custom firmware to replace > The default one, > 2.we don't need to worry about this case,the driver which needs loading firmware > Has to be compiled as a module,and of cause,we load it after the filesystem has been mounted. Forcing the driver to be build modular isn't great for usability either, though - it's not very general and some systems don't use modules at all for various reasons (eg, simplicity or difficulties in keeping the fs and the kernel partitions in sync). I'm not sure this gels well with the approach you're taking with the driver just passing the firmware through. On the one hand you don't want the driver to know anything about the firmware or join up the firmware handling with the audio subsystem but on the other hand the driver is requiring that a firmware be loaded as soon as the driver is even though it can provide some very basic functionality without it. It'd seem better to just let the user load firmware at their leisure.

1 0

[alsa-devel] [PATCH] ALSA: HDA: Remove over-initialization of mux in ALC269VB
by David Henningsson 11 Mar '11

11 Mar '11

Takashi, would you like to give this one a thought to make sure I haven't missed any case where this verb is needed. For both models (amic and dmic), the verb is re-issued, and the autoparser would either add a mux, or at least set it to a correct start. The stable-cc (2.6.38+) is set that way because we tried it on top of 2.6.35 and it didn't work on the particular machine. I don't know about 2.6.36 or 2.6.37. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 3

[alsa-devel] [PATCH] intel8x0m: append 'm' to "r_intel8x0"
by Paul Bolle 11 Mar '11

11 Mar '11

Appending an 'm' will distinguish it from a similar struct in intel8x0.c Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- sound/pci/intel8x0m.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index e16b3e9..ae31655 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -1108,7 +1108,7 @@ struct ich_reg_info { static int __devinit snd_intel8x0m_create(struct snd_card *card, struct pci_dev *pci, unsigned long device_type, - struct intel8x0m ** r_intel8x0) + struct intel8x0m **r_intel8x0m) { struct intel8x0m *chip; int err; @@ -1124,7 +1124,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, }; struct ich_reg_info *tbl; - *r_intel8x0 = NULL; + *r_intel8x0m = NULL; if ((err = pci_enable_device(pci)) < 0) return err; @@ -1237,7 +1237,7 @@ static int __devinit snd_intel8x0m_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); - *r_intel8x0 = chip; + *r_intel8x0m = chip; return 0; } -- 1.7.4.1

1 0

[alsa-devel] Fix for recording issue of ASUS N82JV
by Chih-Wei Huang 11 Mar '11

11 Mar '11

Hi, This small patch fixes the unable to record issue on ASUS N82JV. Thanks. Tested OK on android-x86 2.2 & 2.3 with kernel 2.6.38-rc4. Regards,

2 4

[alsa-devel] [PATCH] Four patches to fixup surround internal speakers on Realtek 88x
by David Henningsson 10 Mar '11

10 Mar '11

I spend last Friday together with Bartłomiej Żogała fixing up a long standing issue with Lenovo Y530, which has 4+1 internal speakers on a Realtek 888. And we all want that supported by the auto parser, don't we? So here comes the patches. Would be nice to have in 2.6.38. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 4

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 10 Mar '11

10 Mar '11

The following changes since commit f9c540456ab662a3b5d79c1fa93c51305a97fdf3: ASoC: Remove warnings in ep93xx-i2s.c (2011-03-07 17:32:02 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Alexander Sverdlin (4): ASoC: Manage mode and rate bits correctly for CS4271 CODEC. ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC. ASoC: Improve EP93xx I2S clocks management. ASoC: Enable 192kHz sample rate for EP93xx. Dan Carpenter (1): ASoC: sgtl5000: use after free in ldo_regulator_register() Mark Brown (16): ASoC: Add DAPM widget and path data to CODEC driver structure ASoC: Use data based init for WM9081 DAPM ASoC: Fix section mismatch warnings in WM8994 ASoC: Provide CODEC clocking operations and API calls ASoC: Convert WM9081 SYSCLK configuration to be device wide ASoC: Refactor symmetric_rates check to reduce indentation ASoC: Warn rather than set a silly constraint when we can't do symmetry ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew() ASoC: Simplify WM9081 speaker startup by using widgets for spaker output ASoC: Fix broken bitfield definitions in WM8978 ASoC: Use the correct DAPM context when cleaning up final widget set ASoC: Fix double addition of prefixes due to widget prefixing Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Fix typo in late revision WM8994 DAC2R name ASoC: Ensure WM8958 gets all WM8994 late revision widgets Merge branch 'for-2.6.38' into for-2.6.39 include/sound/soc.h | 20 +++++- sound/soc/codecs/cs4271.c | 83 ++++++++++++++-------- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/wm8994-tables.c | 2 +- sound/soc/codecs/wm8994.c | 10 +++- sound/soc/codecs/wm8994.h | 2 +- sound/soc/codecs/wm9081.c | 51 ++++--------- sound/soc/ep93xx/edb93xx.c | 16 ++-- sound/soc/ep93xx/ep93xx-i2s.c | 23 +++--- sound/soc/ep93xx/ep93xx-pcm.c | 4 +- sound/soc/soc-core.c | 141 ++++++++++++++++++++++++++++++-------- sound/soc/soc-dapm.c | 45 +++++++++++-- 12 files changed, 274 insertions(+), 127 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 10 Mar '11

10 Mar '11

The following changes since commit ffd6eae2a0d18ca4a741615292a9c9ce904307fb: ASoC: AM3517: Update codec name after multi-component update (2011-03-09 11:48:55 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Mark Brown (4): ASoC: Fix broken bitfield definitions in WM8978 ASoC: Use the correct DAPM context when cleaning up final widget set ASoC: Fix typo in late revision WM8994 DAC2R name ASoC: Ensure WM8958 gets all WM8994 late revision widgets The DAPM context fix is essential for multi-component DAPM, without it we will try to apply updates from the end of DAPM sequences to the wrong device which isn't terribly clever. On other systems the change will be a noop. The other three updates are device specific ones, the WM8994/58 ones are slightly more invasive but are critical to WM8958. There may be one more update (for mini2440) coming later. sound/soc/codecs/wm8978.c | 14 ++++++++------ sound/soc/codecs/wm8994.c | 10 +++++++++- sound/soc/soc-dapm.c | 2 +- 3 files changed, 18 insertions(+), 8 deletions(-)

2 1

[alsa-devel] [PATCH] ALSA: Add snd_ctl_activate_id()
by Mark Brown 10 Mar '11

10 Mar '11

From: Takashi Iwai <tiwai(a)suse.de> Added a new API function snd_ctl_activate_id() for activate / inactivate the control element dynamically. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- You wrote this patch a long time ago now - I'm currently working on a driver which I think can make use if it. Is it OK if I cherry-pick it over into my code and push it through the ASoC branch if I do so? Or if you could merge it in the coming merge window, but there's a risk I may come up with another way of doing things that doesn't need it. include/sound/control.h | 2 ++ sound/core/control.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 48 insertions(+), 0 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 7715e6f..e67db28 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -115,6 +115,8 @@ int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); +int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, + int active); struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid); struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id); diff --git a/sound/core/control.c b/sound/core/control.c index 9ce00ed..db51e4e 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -466,6 +466,52 @@ error: } /** + * snd_ctl_activate_id - activate/inactivate the control of the given id + * @card: the card instance + * @id: the control id to activate/inactivate + * @active: non-zero to activate + * + * Finds the control instance with the given id, and activate or + * inactivate the control together with notification, if changed. + * + * Returns 0 if unchanged, 1 if changed, or a negative error code on failure. + */ +int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, + int active) +{ + struct snd_kcontrol *kctl; + struct snd_kcontrol_volatile *vd; + unsigned int index_offset; + int ret; + + down_write(&card->controls_rwsem); + kctl = snd_ctl_find_id(card, id); + if (kctl == NULL) { + ret = -ENOENT; + goto unlock; + } + index_offset = snd_ctl_get_ioff(kctl, &kctl->id); + vd = &kctl->vd[index_offset]; + ret = 0; + if (active) { + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)) + goto unlock; + vd->access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } else { + if (vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) + goto unlock; + vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + } + ret = 1; + unlock: + up_write(&card->controls_rwsem); + if (ret > 0) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id); + return ret; +} +EXPORT_SYMBOL_GPL(snd_ctl_activate_id); + +/** * snd_ctl_rename_id - replace the id of a control on the card * @card: the card instance * @src_id: the old id -- 1.7.2.3

2 2

[alsa-devel] HDA: Problem with sound becoming corrupted
by Colin Guthrie 10 Mar '11

10 Mar '11

Hi, I've been dealing with this issue for several months (from memory I'm pretty sure it's affected me since 2.6.36, possibly before, but I'll have to do some deeper digging to say for sure). The problem is that the sound suddenly goes nuts and starts sounding awful. It's like some buffer pointer is just a little bit out for some reason. I can fix the issue by closing and reopening the PCM device (as I use PA, this is just a simple "pasuspender echo" away). The problem is very easy to reproduce - it happens about 20+ times a day to me - to the extent that I have added a panel launcher with the above fix. We discussed this issue during the meeting but it seems not to be a PA problem (my other PA devices work fine FWIW - various USB things etc, and recording from the monitor source when the sound is odd also records good sounds). You can hear the sounds here (recorded via mic): http://colin.guthr.ie/weird-sound/ And our discussion is here: http://colin.guthr.ie/meetings/pulseaudio-meeting/2011/pulseaudio-meeting.2… My alsa-info is here: http://colin.guthr.ie/alsa-info.txt This mail is really just about finding out if this is a known problem? I know others suffer from the same problem too as I've had bug reports confirming this. Cheers Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mageia Contributor [http://www.mageia.org/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/]

2 2

[alsa-devel] [PATCH 1/2] ASoC: Fix typo in late revision WM8994 DAC2R name
by Mark Brown 09 Mar '11

09 Mar '11

Without this fix the driver won't instantiate properly on relevant devices. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4afbe3b..d926733 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1418,7 +1418,7 @@ SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0, static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = { SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0), -SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), +SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0), SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; -- 1.7.4.1

2 2

[alsa-devel] [GIT PULL] ASoC Fixes for 2.6.38
by Liam Girdwood 09 Mar '11

09 Mar '11

Takashi, If you are planning a final pull for 2.6.38 fixes please include this one. If not, I'll add this for 2.6.39. Thanks Liam --- The following changes since commit 04d286819ba499839d04cbf847f2ea28d5cf4296: ASoC: WM8994: Ensure late enable events are processed for the ADCs (2011-03-01 23:27:07 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.38 Abhilash K V (1): ASoC: AM3517: Update codec name after multi-component update sound/soc/omap/am3517evm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

3 4

Re: [alsa-devel] [BUG] WARNING: at fs/sysfs/dir.c:455 sysfs_add_one+0x83/0x95
by Greg KH 09 Mar '11

09 Mar '11

On Wed, Mar 09, 2011 at 10:39:49AM -0500, Steven Rostedt wrote: > Not sure who to send this to, but last night on v2.6.38-rc8 ktest > triggered this warning: > > > [ 32.062538] YM3812 and OPL-3 driver Copyright (C) by Hannu Savolainen, Rob Hooft 1993-1996 > [ 32.071305] PM: Adding info for No Bus:timer > [ 32.075864] PM: Adding info for platform:snd_dummy.0^M > [ 32.081232] PM: Adding info for No Bus:card0 > [ 32.085853] PM: Adding info for No Bus:pcmC0D0p > [ 32.090650] PM: Adding info for No Bus:pcmC0D0c > [ 32.095667] PM: Adding info for No Bus:dsp > [ 32.099918] ------------[ cut here ]------------ > [ 32.104531] WARNING: at /home/rostedt/work/autotest/nobackup/linux-test.git/fs/sysfs/dir.c:455 sysfs_add_one+0x83/0x95() > [ 32.115379] Hardware name: > [ 32.118859] sysfs: cannot create duplicate filename '/dev/char/14:4' Two modules wanted to create the same minor number, that's not good. 14:4 is the OSS /dev/audio node. > [ 32.125197] Modules linked in: > [ 32.128255] Pid: 1, comm: swapper Not tainted 2.6.38-rc8-test #1 > [ 32.134247] Call Trace:^M > [ 32.136693] [<c044d095>] ? warn_slowpath_common+0x7c/0x91 > [ 32.142168] [<c0552d35>] ? sysfs_add_one+0x83/0x95 > [ 32.147036] [<c0552d35>] ? sysfs_add_one+0x83/0x95 > [ 32.151904] [<c044d135>] ? warn_slowpath_fmt+0x33/0x35 > [ 32.157120] [<c0552d35>] ? sysfs_add_one+0x83/0x95 > [ 32.161989] [<c055393d>] ? sysfs_do_create_link+0xdf/0x181^M > [ 32.167550] [<c073637f>] ? vsnprintf+0x22c/0x266 > [ 32.172246] [<c0553a0f>] ? sysfs_create_link+0x17/0x1c > [ 32.177462] [<c0950ce4>] ? device_add+0x183/0x573 > [ 32.182245] [<c0958474>] ? device_pm_init+0x5d/0x61 > [ 32.187200] [<c09510eb>] ? device_register+0x17/0x1a > [ 32.192243] [<c0951165>] ? device_create_vargs+0x77/0x98 > [ 32.197631] [<c09511b3>] ? device_create+0x2d/0x2f > [ 32.202501] [<c0dc5caa>] ? sound_insert_unit+0x1ba/0x1d5 > [ 32.207891] [<c0dc5e8e>] ? register_sound_special_device+0x103/0x11c > [ 32.214319] [<c0dd7875>] ? snd_register_oss_device+0x10a/0x169 > [ 32.220229] [<c0de6cd6>] ? register_oss_dsp+0x62/0xb5 Yup, sound subsystem... > [ 32.225358] [<c046b95c>] ? sched_clock_cpu+0x16a/0x17a > [ 32.230575] [<c04778f2>] ? mark_held_locks+0x43/0x5b > [ 32.235616] [<c0fa124c>] ? __mutex_lock_common+0x39f/0x3c7 > [ 32.241177] [<c04776d2>] ? mark_lock+0x1e/0x1fb > [ 32.245787] [<c04778f2>] ? mark_held_locks+0x43/0x5b > [ 32.250838] [<c0fa0cff>] ? __mutex_unlock_slowpath+0x137/0x153 > [ 32.256745] [<c0de6d69>] ? snd_pcm_oss_register_minor+0x40/0xe3 > [ 32.262737] [<c0fa0d28>] ? mutex_unlock+0xd/0xf > [ 32.267345] [<c0dd96b7>] ? snd_timer_dev_register+0x10d/0x117 > [ 32.273165] [<c0dd6db9>] ? snd_device_register+0x67/0xd2 > [ 32.278551] [<c07363d5>] ? sprintf+0x1c/0x1e^M > [ 32.282900] [<c0de2db7>] ? snd_pcm_timer_init+0xba/0xef > [ 32.288201] [<c0ddb426>] ? snd_pcm_dev_register+0x1e8/0x224 > [ 32.293848] [<c0951000>] ? device_add+0x49f/0x573 > [ 32.298630] [<c0dd6d32>] ? snd_device_register_all+0x5e/0x7e > [ 32.304363] [<c09511b3>] ? device_create+0x2d/0x2f > [ 32.309230] [<c0dd32d3>] ? snd_card_register+0x92/0x1af > [ 32.314531] [<c0f87b9b>] ? snd_dummy_probe+0x322/0x3b2 > [ 32.319745] [<c0953f57>] ? platform_drv_probe+0x14/0x16 > [ 32.325046] [<c0953154>] ? driver_probe_device+0x116/0x1ed > [ 32.330606] [<c073b414>] ? do_raw_spin_unlock+0x8f/0x94 > [ 32.335906] [<c09532c5>] ? __device_attach+0x2f/0x33^M > [ 32.340946] [<c09524ab>] ? bus_for_each_drv+0x44/0x6e > [ 32.346073] [<c095334e>] ? device_attach+0x61/0x7e^M > [ 32.350941] [<c0953296>] ? __device_attach+0x0/0x33 > [ 32.355895] [<c09522e8>] ? bus_probe_device+0x23/0x38 > [ 32.361023] [<c0950f61>] ? device_add+0x400/0x573 > [ 32.365805] [<c0730ac4>] ? kobject_set_name_vargs+0x46/0x4c > [ 32.371453] [<c0954624>] ? platform_device_add+0x139/0x182 > [ 32.377015] [<c095487d>] ? platform_device_register_resndata+0x4f/0x65 > [ 32.383616] [<c17e2210>] ? alsa_card_dummy_init+0xcc/0x121 > [ 32.389177] [<c0403096>] ? do_one_initcall+0x8c/0x161 > [ 32.394305] [<c17e2144>] ? alsa_card_dummy_init+0x0/0x121 > [ 32.399780] [<c17843cf>] ? kernel_init+0x12c/0x1c2 > [ 32.404646] [<c17842a3>] ? kernel_init+0x0/0x1c2 > [ 32.409340] [<c040a8c2>] ? kernel_thread_helper+0x6/0x10 > [ 32.414782] ---[ end trace 5d9e7ab86ca08fa1 ]--- > [ 32.419772] PM: Adding info for No Bus:controlC0 > [ 32.424867] PM: Adding info for isa:es18xx.0 > [ 32.429167] PM: Removing info for isa:es18xx.0 > > > This was for a 386 build. I could give a .config and full dmesg if > needed. I think the sound developers (now on CC: would probably like to see that.) thanks, greg k-h

3 6

[alsa-devel] Query about Playback device name
by Irfan Shaikh 09 Mar '11

09 Mar '11

Hello, aplay uses following device "aplay -Dplug:mcasp3 411.2Kbps-44.1Khz-16bits-2CH-PCM.wav" Playing WAVE '411.2Kbps-44.1Khz-16bits-2CH-PCM.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo I want to use same device i.e "plug:mcasp3" in my plaback application 1) Earlier i use to use default device //alsaParams->playback_device = strdup("plughw:0,0"); 2) Can i use following section of code to use "plug:mcasp3" as playback device? Will following section of code work ?????? alsaParams->playback_device = strdup("plug:mcasp3"); <<<<<------------------------<<<<<<<-----------------<<<<<<< snd_pcm_hw_params_alloca(&alsaParams->hwparams_playback); /*Open PCM Playback device*/ if (snd_pcm_open(&alsaParams->playback_handle, alsaParams->playback_device, alsaParams->stream, alsaParams->device_mode) < 0) Thanks and regards SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 4

[alsa-devel] [PATCH 2/6] ASoC: Manage mode and rate bits correctly for CS4271 CODEC.
by Alexander Sverdlin 09 Mar '11

09 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC. This is done to make capture work properly. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 83 ++++++++++++++++++++++++++++---------------- 1 files changed, 53 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e5cc4da..c71ad08 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -168,27 +168,6 @@ struct cs4271_private { int gpio_disable; }; -struct cs4271_clk_cfg { - unsigned int ratio; /* MCLK / sample rate */ - u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ - u8 mclk_master; /* ratio bit mask for Master mode */ - u8 mclk_slave; /* ratio bit mask for Slave mode */ -}; - -static struct cs4271_clk_cfg cs4271_clk_tab[] = { - {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, - {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, - {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} -}; - -#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) - /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -297,6 +276,45 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, return cs4271_set_deemph(codec); } +struct cs4271_clk_cfg { + bool master; /* codec mode */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + unsigned short ratio; /* MCLK / sample rate */ + u8 ratio_mask; /* ratio bit mask for Master mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {1, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_3}, + {0, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 512, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 256, CS4271_MODE1_DIV_2}, +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -308,23 +326,28 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, unsigned int ratio, val; cs4271->rate = params_rate(params); + + /* Configure DAC */ + if (cs4271->rate < 50000) + val = CS4271_MODE1_MODE_1X; + else if (cs4271->rate < 100000) + val = CS4271_MODE1_MODE_2X; + else + val = CS4271_MODE1_MODE_4X; + ratio = cs4271->mclk / cs4271->rate; for (i = 0; i < CS4171_NR_RATIOS; i++) - if (cs4271_clk_tab[i].ratio == ratio) + if ((cs4271_clk_tab[i].master == cs4271->master) && + (cs4271_clk_tab[i].speed_mode == val) && + (cs4271_clk_tab[i].ratio == ratio)) break; - if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + if (i == CS4171_NR_RATIOS) { dev_err(codec->dev, "Invalid sample rate\n"); return -EINVAL; } - /* Configure DAC */ - val = cs4271_clk_tab[i].speed_mode; - - if (cs4271->master) - val |= cs4271_clk_tab[i].mclk_master; - else - val |= cs4271_clk_tab[i].mclk_slave; + val |= cs4271_clk_tab[i].ratio_mask; ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);

2 1

[alsa-devel] [PATCH] ASoC: Fix double addition of prefixes due to widget prefixing
by Mark Brown 09 Mar '11

09 Mar '11

We're not only prefixing all controls, we're also prefixing the widget names in the runtime data. This causes us to add the prefix twice - once when using the widget name to generate the control name and once when adding the control. Really we shouldn't be prefixing the widget names at all, the matching code should be handing this as we always know which DAPM context a widget came from and always display the widget name in terms of a DAPM context. However, we're quite close to the merge window and that's relatively invasive. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 29 ++++++++++++++++++++++++++--- 1 files changed, 26 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5335da..81c4052 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -366,7 +366,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *w) { int i, ret = 0; - size_t name_len; + size_t name_len, prefix_len; struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; const char *prefix; @@ -376,6 +376,11 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, else prefix = NULL; + if (prefix) + prefix_len = strlen(prefix) + 1; + else + prefix_len = 0; + /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -403,8 +408,15 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, switch (w->id) { default: + /* The control will get a prefix from + * the control creation process but + * we're also using the same prefix + * for widgets so cut the prefix off + * the front of the widget name. + */ snprintf(path->long_name, name_len, "%s %s", - w->name, w->kcontrols[i].name); + w->name + prefix_len, + w->kcontrols[i].name); break; case snd_soc_dapm_mixer_named_ctl: snprintf(path->long_name, name_len, "%s", @@ -438,6 +450,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; const char *prefix; + size_t prefix_len; int ret = 0; if (!w->num_kcontrols) { @@ -450,7 +463,17 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, else prefix = NULL; - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name, prefix); + if (prefix) + prefix_len = strlen(prefix) + 1; + else + prefix_len = 0; + + /* The control will get a prefix from the control creation + * process but we're also using the same prefix for widgets so + * cut the prefix off the front of the widget name. + */ + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name + prefix_len, + prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) -- 1.7.4.1

3 2

[alsa-devel] [PATCH] ASoC: AM3517: Update codec name after multi-component update
by Abhilash K V 09 Mar '11

09 Mar '11

The i2c client device name (".2-001a" in this case, including the separator period) for the AIC23 codec on the TI AM3517-EVM was appended to the codec_name member of am3517evm_dai to resolve the names mismatch happening in soc_bind_dai_link(), due to which the card was not getting registered. Signed-off-by: Abhilash K V <abhilash.kv(a)ti.com> Acked-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/omap/am3517evm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 979dd50..ddbd2a1 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = { .cpu_dai_name ="omap-mcbsp-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", - .codec_name = "tlv320aic23-codec", + .codec_name = "tlv320aic23-codec.2-001a", .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; -- 1.6.2.4

3 2

[alsa-devel] [PATCH] ASoC: omap: rx51: Enable McBSP2 sidetone
by Jarkko Nikula 09 Mar '11

09 Mar '11

McBSP sidetone is needed in telephony applications. McBSP sidetone is a configurable FIR filter that forms a loopback from McBSP input to output. This patch enables the McBSP2 sidetone ALSA controls so that it can be used on Nokia RX-51/N900. Sidetone feature can be tested with following commands: (set up codec input and output paths) # Enable and configure sidetone amixer -D hw:0 set 'McBSP2 Sidetone' on amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767 echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps # Do not loop audio via CPU arecord -f dat >/dev/null |aplay /dev/zero Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> --- sound/soc/omap/rx51.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 055e447..d098622 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -331,6 +331,10 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42); + err = omap_mcbsp_st_add_controls(codec, 1); + if (err < 0) + return err; + snd_soc_dapm_sync(dapm); /* AV jack detection */ -- 1.7.2.3

3 2

[alsa-devel] [PATCH] ASoC: Use the correct DAPM context when cleaning up final widget set
by Mark Brown 09 Mar '11

09 Mar '11

Now we've got multi-component we need to make sure that the DAPM context (and hence register I/O context) we use to apply the pending updates at the end of a DAPM sequence is the one we were processing rather than the one that was used to initate the state change. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a6fb85d..c5335da 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -984,7 +984,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, } if (!list_empty(&pending)) - dapm_seq_run_coalesced(dapm, &pending); + dapm_seq_run_coalesced(cur_dapm, &pending); if (cur_dapm && cur_dapm->seq_notifier) { for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Simplify WM9081 speaker startup by using widgets for spaker output
by Mark Brown 09 Mar '11

09 Mar '11

Now we have a register write minimisation code in DAPM we don't need to worry about the ordering of the enable and disable of the PGA and the output stage. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 32 ++++++-------------------------- 1 files changed, 6 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index effaf75..55cdf29 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -389,27 +389,6 @@ SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0), SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0), }; -static int speaker_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - reg |= WM9081_SPK_ENA; - break; - - case SND_SOC_DAPM_PRE_PMD: - reg &= ~WM9081_SPK_ENA; - break; - } - - snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg); - - return 0; -} - struct _fll_div { u16 fll_fratio; u16 fll_outdiv; @@ -747,9 +726,8 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0, - speaker_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LINEOUT"), SND_SOC_DAPM_OUTPUT("SPKN"), @@ -780,8 +758,10 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "Speaker PGA", NULL, "TOCLK" }, { "Speaker PGA", NULL, "CLK_SYS" }, - { "SPKN", NULL, "Speaker PGA" }, - { "SPKP", NULL, "Speaker PGA" }, + { "Speaker", NULL, "Speaker PGA" }, + + { "SPKN", NULL, "Speaker" }, + { "SPKP", NULL, "Speaker" }, }; static int wm9081_set_bias_level(struct snd_soc_codec *codec, -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew()
by Mark Brown 09 Mar '11

09 Mar '11

Currently will ignore prefixes when creating DAPM controls. Since currently all control creation goes through snd_soc_cnew() we can fix this by factoring the prefixing into that function. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 ++- sound/soc/soc-core.c | 45 +++++++++++++++++++++++++++++++-------------- sound/soc/soc-dapm.c | 16 ++++++++++++++-- 3 files changed, 47 insertions(+), 17 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8b097f7..876dd5a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -341,7 +341,8 @@ void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); *Controls */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name); + void *data, char *long_name, + const char *prefix); int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls); int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index db3075d..17efacd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2344,22 +2344,45 @@ EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); * @_template: control template * @data: control private data * @long_name: control long name + * @prefix: control name prefix * * Create a new mixer control from a template control. * * Returns 0 for success, else error. */ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, - void *data, char *long_name) + void *data, char *long_name, + const char *prefix) { struct snd_kcontrol_new template; + struct snd_kcontrol *kcontrol; + char *name = NULL; + int name_len; memcpy(&template, _template, sizeof(template)); - if (long_name) - template.name = long_name; template.index = 0; - return snd_ctl_new1(&template, data); + if (!long_name) + long_name = template.name; + + if (prefix) { + name_len = strlen(long_name) + strlen(prefix) + 2; + name = kmalloc(name_len, GFP_ATOMIC); + if (!name) + return NULL; + + snprintf(name, name_len, "%s %s", prefix, long_name); + + template.name = name; + } else { + template.name = long_name; + } + + kcontrol = snd_ctl_new1(&template, data); + + kfree(name); + + return kcontrol; } EXPORT_SYMBOL_GPL(snd_soc_cnew); @@ -2378,22 +2401,16 @@ int snd_soc_add_controls(struct snd_soc_codec *codec, const struct snd_kcontrol_new *controls, int num_controls) { struct snd_card *card = codec->card->snd_card; - char prefixed_name[44], *name; int err, i; for (i = 0; i < num_controls; i++) { const struct snd_kcontrol_new *control = &controls[i]; - if (codec->name_prefix) { - snprintf(prefixed_name, sizeof(prefixed_name), "%s %s", - codec->name_prefix, control->name); - name = prefixed_name; - } else { - name = control->name; - } - err = snd_ctl_add(card, snd_soc_cnew(control, codec, name)); + err = snd_ctl_add(card, snd_soc_cnew(control, codec, + control->name, + codec->name_prefix)); if (err < 0) { dev_err(codec->dev, "%s: Failed to add %s: %d\n", - codec->name, name, err); + codec->name, control->name, err); return err; } } diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 570db88..a6fb85d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -369,6 +369,12 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, size_t name_len; struct snd_soc_dapm_path *path; struct snd_card *card = dapm->card->snd_card; + const char *prefix; + + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -409,7 +415,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, path->long_name[name_len - 1] = '\0'; path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, - path->long_name); + path->long_name, prefix); ret = snd_ctl_add(card, path->kcontrol); if (ret < 0) { dev_err(dapm->dev, @@ -431,6 +437,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; + const char *prefix; int ret = 0; if (!w->num_kcontrols) { @@ -438,7 +445,12 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, return -EINVAL; } - kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); + if (dapm->codec) + prefix = dapm->codec->name_prefix; + else + prefix = NULL; + + kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name, prefix); ret = snd_ctl_add(card, kcontrol); if (ret < 0) -- 1.7.2.3

3 4

[PATCH] ALSA: HDA: Remove over-initialization of mux in ALC269VB
by David Henningsson 09 Mar '11

09 Mar '11

This initialization verb breaks the automatic mux setup in some cases (e g just one internal mic and nothing else), and it's not needed as both laptop_amic and laptop_dmic models set this verb anyway. Cc: stable(a)kernel.org (2.6.38+) BugLink: http://bugs.launchpad.net/bugs/732445 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33346c1..cf66d0f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14574,11 +14574,6 @@ static struct hda_verb alc269vb_init_verbs[] = { {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* FIXME: use Mux-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x22, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* set EAPD */ {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } -- 1.7.1 --------------080502060804070905010802--

1 0

[alsa-devel] [patch] ASoC: sgtl5000: use after free in ldo_regulator_register()
by Dan Carpenter 08 Mar '11

08 Mar '11

The "ldo" variable was dereferenced after free on the error path. Signed-off-by: Dan Carpenter <error27(a)gmail.com> diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b7e97c0..1f7217f 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -875,11 +875,13 @@ static int ldo_regulator_register(struct snd_soc_codec *codec, ldo->dev = regulator_register(&ldo->desc, codec->dev, init_data, ldo); if (IS_ERR(ldo->dev)) { + int ret = PTR_ERR(ldo->dev); + dev_err(codec->dev, "failed to register regulator\n"); kfree(ldo->desc.name); kfree(ldo); - return PTR_ERR(ldo->dev); + return ret; } return 0;

3 2

[alsa-devel] [PATCH 1/2] ASoC: Refactor symmetric_rates check to reduce indentation
by Mark Brown 08 Mar '11

08 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 28 ++++++++++++++-------------- 1 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c2ec6cb..65ff612 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -500,20 +500,20 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates || - rtd->dai_link->symmetric_rates) { - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", - rtd->rate); - - ret = snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, - rtd->rate); - if (ret < 0) { - dev_err(&rtd->dev, - "Unable to apply rate symmetry constraint: %d\n", ret); - return ret; - } + if (!codec_dai->driver->symmetric_rates && + !cpu_dai->driver->symmetric_rates && + !rtd->dai_link->symmetric_rates) + return 0; + + dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + rtd->rate, rtd->rate); + if (ret < 0) { + dev_err(&rtd->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; } return 0; -- 1.7.2.3

2 2

[alsa-devel] [PATCH 1/2] ASoC: Provide CODEC clocking operations and API calls
by Mark Brown 08 Mar '11

08 Mar '11

When multi component systems use DAIless amplifiers which require clocking configuration it is at best hard to use the current clocking API as this requires a DAI even though the device may not even have one. Address this by adding set_sysclk() and set_pll() operations and APIs for CODECs. In order to avoid issues with devices which could be used either with or without DAIs make the DAI variants call through to their CODEC counterparts if there is no DAI specific operation. Converting over entirely would create problems for multi-DAI devices which offer per-DAI clocking setup. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 11 +++++++++++ sound/soc/soc-core.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 57 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 363e3a8..8b097f7 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -260,6 +260,11 @@ enum snd_soc_compress_type { SND_SOC_RBTREE_COMPRESSION }; +int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir); +int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out); + int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); @@ -569,6 +574,12 @@ struct snd_soc_codec_driver { const struct snd_soc_dapm_route *dapm_routes; int num_dapm_routes; + /* codec wide operations */ + int (*set_sysclk)(struct snd_soc_codec *codec, + int clk_id, unsigned int freq, int dir); + int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out); + /* codec IO */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 286a385..9b482dd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3062,12 +3062,34 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, { if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); + else if (dai->codec && dai->codec->driver->set_sysclk) + return dai->codec->driver->set_sysclk(dai->codec, clk_id, + freq, dir); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); /** + * snd_soc_codec_set_sysclk - configure CODEC system or master clock. + * @codec: CODEC + * @clk_id: DAI specific clock ID + * @freq: new clock frequency in Hz + * @dir: new clock direction - input/output. + * + * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. + */ +int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, + unsigned int freq, int dir) +{ + if (codec->driver->set_sysclk) + return codec->driver->set_sysclk(codec, clk_id, freq, dir); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); + +/** * snd_soc_dai_set_clkdiv - configure DAI clock dividers. * @dai: DAI * @div_id: DAI specific clock divider ID @@ -3103,11 +3125,35 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, if (dai->driver && dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); + else if (dai->codec && dai->codec->driver->set_pll) + return dai->codec->driver->set_pll(dai->codec, pll_id, source, + freq_in, freq_out); else return -EINVAL; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); +/* + * snd_soc_codec_set_pll - configure codec PLL. + * @codec: CODEC + * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL + * @freq_in: PLL input clock frequency in Hz + * @freq_out: requested PLL output clock frequency in Hz + * + * Configures and enables PLL to generate output clock based on input clock. + */ +int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + if (codec->driver->set_pll) + return codec->driver->set_pll(codec, pll_id, source, + freq_in, freq_out); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); + /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI -- 1.7.2.3

2 2

[alsa-devel] [PATCH 1/2] ASoC: Add DAPM widget and path data to CODEC driver structure
by Mark Brown 08 Mar '11

08 Mar '11

Allow a slight simplification of CODEC drivers by allowing DAPM routes and widgets to be provided in a table. They will be instantiated at the end of CODEC probe. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 12 ++++++++++-- 2 files changed, 16 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a23f5a5..363e3a8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -563,6 +563,12 @@ struct snd_soc_codec_driver { pm_message_t state); int (*resume)(struct snd_soc_codec *); + /* Default DAPM setup, added after probe() is run */ + const struct snd_soc_dapm_widget *dapm_widgets; + int num_dapm_widgets; + const struct snd_soc_dapm_route *dapm_routes; + int num_dapm_routes; + /* codec IO */ unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8926d38..286a385 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1464,6 +1464,7 @@ static int soc_probe_codec(struct snd_soc_card *card, struct snd_soc_codec *codec) { int ret = 0; + const struct snd_soc_codec_driver *driver = codec->driver; codec->card = card; codec->dapm.card = card; @@ -1472,8 +1473,8 @@ static int soc_probe_codec(struct snd_soc_card *card, if (!try_module_get(codec->dev->driver->owner)) return -ENODEV; - if (codec->driver->probe) { - ret = codec->driver->probe(codec); + if (driver->probe) { + ret = driver->probe(codec); if (ret < 0) { dev_err(codec->dev, "asoc: failed to probe CODEC %s: %d\n", @@ -1482,6 +1483,13 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + if (driver->dapm_widgets) + snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, + driver->num_dapm_widgets); + if (driver->dapm_routes) + snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, + driver->num_dapm_routes); + soc_init_codec_debugfs(codec); /* mark codec as probed and add to card codec list */ -- 1.7.2.3

2 3

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 08 Mar '11

08 Mar '11

The following changes since commit 4a5f7bda8fe9d0ed08ed4c5beb5dc3fa62f09d05: ASoC: Add platform data for WM9081 IRQ pin configuration (2011-03-01 23:30:53 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Alexander Sverdlin (2): ASoC: Extend range of supported sample rates for CS4271 CODEC. ASoC: Remove warnings in ep93xx-i2s.c Axel Lin (2): ASoC: tlv320dac33: add MODULE_DEVICE_TABLE ASoC: Constify i2c_device_id table Dimitris Papastamos (1): ASoC: WM8994: Ensure MICBIAS is provided with a clock Javier Martin (2): ASoC: Add TI tlv320aic32x4 codec support. ASoC: Add machine driver for Visstrim_M10 board. Lars-Peter Clausen (6): ASoC: neo1973_wm8753: Remove scenario management code. ASoC: neo1973_wm8753: Move lm4857 specefic code to its own module ASoC: neo1973_gta02_wm8753: Remove lm4853_{set,get}_state ASoC: neo1973_gta02_wm8753: Use gpio_request_array to request gpios ASoC: Samsung: Merge neo1937_wm8753 and neo1973_gta02_wm8753 sound board driver ASoC: Add driver for the dfbmcs320 bluetooth module Mark Brown (11): ASoC: Fix broken bitfield definitions in WM8978 Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Add TLV information for WM8978 DAC limiter ASoC: Get the card directly from the DAPM context ASoC: Check for a CODEC before dereferencing in DAPM ASoC: Add a per-card DAPM context ASoC: Allow card DAPM widgets and routes to be set up at registration ASoC: Add a late_probe() callback to cards ASoC: Fix outdated API usage in tlv320aic32x4 ASoC: Add missing debugfs conditionals ASoC: Add LM4857 to SND_SOC_ALL_CODECS include/sound/soc.h | 12 + include/sound/tlv320aic32x4.h | 31 ++ sound/soc/codecs/Kconfig | 13 +- sound/soc/codecs/Makefile | 6 + sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 7 +- sound/soc/codecs/dfbmcs320.c | 72 +++ sound/soc/codecs/lm4857.c | 276 +++++++++++ sound/soc/codecs/tlv320aic32x4.c | 794 ++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.h | 143 ++++++ sound/soc/codecs/tlv320dac33.c | 1 + sound/soc/codecs/wm8978.c | 17 +- sound/soc/ep93xx/ep93xx-i2s.c | 8 +- sound/soc/imx/Kconfig | 10 + sound/soc/imx/Makefile | 2 + sound/soc/imx/mx27vis-aic32x4.c | 137 +++++ sound/soc/samsung/Kconfig | 19 +- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/lm4857.h | 32 -- sound/soc/samsung/neo1973_gta02_wm8753.c | 494 ------------------- sound/soc/samsung/neo1973_wm8753.c | 634 +++++++++--------------- sound/soc/soc-core.c | 31 ++ sound/soc/soc-dapm.c | 14 +- 23 files changed, 1795 insertions(+), 962 deletions(-) create mode 100644 include/sound/tlv320aic32x4.h create mode 100644 sound/soc/codecs/dfbmcs320.c create mode 100644 sound/soc/codecs/lm4857.c create mode 100644 sound/soc/codecs/tlv320aic32x4.c create mode 100644 sound/soc/codecs/tlv320aic32x4.h create mode 100644 sound/soc/imx/mx27vis-aic32x4.c delete mode 100644 sound/soc/samsung/lm4857.h delete mode 100644 sound/soc/samsung/neo1973_gta02_wm8753.c

2 1

[alsa-devel] Channel swapping issue on TI OMAP3/TWL4030
by ylin＠mail.com 08 Mar '11

08 Mar '11

Hi, We are using a customized TI OMAP3 with TWL4030, similar to OMAP3EVM board. We noticed occasionally (after bring up few hours) the mic channel audio is swapped. Has anyone seen similar problem? Any suggestion where the problem might be? Thanks, Ying

4 24

[alsa-devel] [PATCH 0/8] Even more hdspmixer fixes
by Adrian Knoth 08 Mar '11

08 Mar '11

Hi! After some feedback with a couple of users, here's another round of fixes for hdspmixer. Anybody interested in a clean rewrite-from-scratch with GTK or Qt? ;) Cheers Adrian Knoth (8): hdspmixer: Replace hardcoded 3 by MAX_CARDS hdspmixer: Fix preset activation segfault on AES(32) hdspmixer: Adjust playback channel count on Multiface. hdspmixer: Introduce output channel count for all cards. hdspmixer: Provide headphones out on Multiface/Digiface hdspmixer: Fix code indentation hdspmixer: [cosmetics] Break code into logic blocks hdspmixer: Show cardname in window title hdspmixer/src/HDSPMixerCard.cxx | 391 +++++++++++++++++-------------- hdspmixer/src/HDSPMixerCard.h | 5 +- hdspmixer/src/HDSPMixerCardSelector.cxx | 1 + hdspmixer/src/HDSPMixerIOMixer.cxx | 2 +- hdspmixer/src/HDSPMixerWindow.cxx | 65 ++++-- hdspmixer/src/HDSPMixerWindow.h | 2 + hdspmixer/src/channelmap.cxx | 11 +- hdspmixer/src/hdspmixer.cxx | 21 +- 8 files changed, 284 insertions(+), 214 deletions(-) -- 1.7.4.1

2 9

[alsa-devel] [PATCH 1/1] ALSA: hdspm - Add firmware ID for older AES32 cards.
by Adrian Knoth 08 Mar '11

08 Mar '11

The current AES32 firmware revision ID is 234, however, a user confirmed that everything works fine with the previous revision, too. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0ed4b4e..a323eaf 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -527,6 +527,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_MADIFACE_REV 213 #define HDSPM_AES_REV 240 #define HDSPM_AES32_REV 234 +#define HDSPM_AES32_OLD_REV 233 /* speed factor modes */ #define HDSPM_SPEED_SINGLE 0 @@ -6397,6 +6398,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, break; case HDSPM_AES_REV: case HDSPM_AES32_REV: + case HDSPM_AES32_OLD_REV: hdspm->io_type = AES32; hdspm->card_name = "RME AES32"; hdspm->midiPorts = 2; -- 1.7.4.1

2 1

[alsa-devel] WM9712/WM9715 - WM97XXPENDOWN issue
by Tony Prisk 08 Mar '11

08 Mar '11

I have any issue with the WM9712 codec that I was hoping someone might be familiar with. I actually have a Wondermedia WM8505 based netbook with the WM9715 codec - although it seems to be compatible with the WM9711/WM9712. I have written the required ASoC support for this board/platform, and wm97xx battery support is working correctly except for one issue. Touchscreen support is required, but not available on this unit (not really an issue outright, but the cause of the problem). Without removing the following code, wm9712_poll_sample always fails when reading the battery ADC. static int wm9712_poll_sample(struct wm97xx *wm, int adcsel, int *sample) ... /* check we have correct sample */ if ((*sample & WM97XX_ADCSEL_MASK) != adcsel) { dev_dbg(wm->dev, "adc wrong sample, read %x got %x", adcsel, *sample & WM97XX_ADCSEL_MASK); return RC_PENUP; } /* <--- PROBLEM CODE HERE ---> if (!(*sample & WM97XX_PEN_DOWN)) { wm->pen_probably_down = 0; return RC_PENUP; } */ return RC_VALID; } Is it possible that with no touchscreen attached, the codec is returning WM97XX_PEN_DOWN on all ADC reads? Any suggestions welcome. Tony Prisk

1 0

[alsa-devel] [PATCH 1/6] ASoC: Extend range of supported sample rates for CS4271 CODEC.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Extend range of supported sample rates for CS4271 CODEC. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> Acked-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/codecs/cs4271.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 1791796..e5cc4da 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -33,6 +33,7 @@ #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000 /* * CS4271 registers @@ -392,14 +393,14 @@ static struct snd_soc_dai_driver cs4271_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .ops = &cs4271_dai_ops,

2 1

[alsa-devel] [PATCH 4/6] ASoC: Remove warnings in ep93xx-i2s.c
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Remove warnings in ep93xx-i2s.c Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> Acked-by: Ryan Mallon <ryan(a)bluewatersys.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/ep93xx/ep93xx-i2s.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 3d4c086..042f4e9 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -313,10 +313,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } static int ep93xx_i2s_resume(struct snd_soc_dai *dai) @@ -324,10 +326,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } #else #define ep93xx_i2s_suspend NULL

2 1

[alsa-devel] [PATCH 6/6] ASoC: Improve EP93xx I2S clocks management.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Improve EP93xx I2S clocks management. Some freqs values are set not exact as they requested for MCLK and original code was not able to find divisors for SCLK and LRCLK. This code just picks up nearest value from 3 possible variants. This patch makes 44100 and 192000 rates working and fixes capture function (by selecting SCLK/LRCLK=64 where possible). All other rates should work as before. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 19 +++++++++---------- 1 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 0458dd4..3d4c086 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); unsigned word_len, div, sdiv, lrdiv; - int found = 0, err; + int err; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, * the codec uses. */ div = clk_get_rate(info->mclk) / params_rate(params); - for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) - if (sdiv * lrdiv == div) { - found = 1; - goto out; - } -out: - if (!found) - return -EINVAL; + sdiv = 4; + if (div > (256 + 512) / 2) { + lrdiv = 128; + } else { + lrdiv = 64; + if (div < (128 + 256) / 2) + sdiv = 2; + } err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); if (err)

1 0

[alsa-devel] [PATCH 5/6] ASoC: Enable 192kHz sample rate for EP93xx.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Enable 192kHz sample rate for EP93xx. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index e1a58d3..87dc1ee 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -351,13 +351,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .ops = &ep93xx_i2s_dai_ops, diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 0667077..a456e49 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_96000, + .rate_max = SNDRV_PCM_RATE_192000, .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |

1 0

[alsa-devel] [PATCH 3/6] ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx machine driver. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/edb93xx.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index b270085..d3aa151 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -41,17 +41,17 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; + unsigned int mclk_rate; unsigned int rate = params_rate(params); + /* - * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, - * because our sample size is 32 bit * 2 channels. - * I2S standard permits us to transmit more bits than - * the codec uses. - * MCLK = SCLK * 4 is the best recommended value, - * but we have to fall back to ratio 2 for higher - * sample rates. + * According to CS4271 datasheet we use MCLK/LRCK=256 for + * rates below 50kHz and 128 for higher sample rates */ - unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + if (rate < 50000) + mclk_rate = rate * 64 * 4; + else + mclk_rate = rate * 64 * 2; err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |

1 0

[alsa-devel] X-Fi Titanium S/PDIF output sample rate
by S. Christian Collins 07 Mar '11

07 Mar '11

I have two sound cards in my system, a Sound Blaster X-Fi Titanium (PCI-Express) and an Echo Gina 3G, with the optical output from the X-Fi connected to the optical input on the Gina 3G. I'm looking for a way to change the S/PDIF output sample rate on the X-Fi to 48 KHz. I've tried using iecset, but none of the commands change any of the values reported by iecset. Here is the output from iecset: Mode: consumer Data: audio Rate: 48000 Hz Copyright: protected Emphasis: none Category: PCM coder Original: original Clock: 1000 ppm Now, here's the weird thing: even though iecset reports the X-Fi S/PDIF rate at 48 KHz, my Echo Gina 3G seems to pick up the stream as 96 KHz instead, ie. the Gina 3G becomes "locked" at 96 KHz upon activating its S/PDIF input and getting its clock from the optical stream. I noticed in the Windows driver that the X-Fi's internal sampling rate can be adjusted in the driver setting (44.1, 48, 88.2, 96), but the S/PDIF rate can't be adjusted. Instead, the X-Fi's S/PDIF rate seems to be locked to the internal sampling rate. So, it appears that I may need to adjust the X-Fi's internal sampling rate somehow, but I have no idea how to do this. Does anybody know if this can even be done? -~Chris

2 1

[alsa-devel] [PATCH] ASoC: Fix section mismatch warnings in WM8994
by Mark Brown 07 Mar '11

07 Mar '11

Annoying as the __devinitdata is actually correct. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994-tables.c | 2 +- sound/soc/codecs/wm8994.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c index 61dfd91..a87adbd 100644 --- a/sound/soc/codecs/wm8994-tables.c +++ b/sound/soc/codecs/wm8994-tables.c @@ -1573,7 +1573,7 @@ const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE] = { { 0x03C3, 0x03C3 }, /* R1569 - Sidetone */ }; -const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { +const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = { 0x8994, /* R0 - Software Reset */ 0x0000, /* R1 - Power Management (1) */ 0x6000, /* R2 - Power Management (2) */ diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 0c355bf..999b885 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -43,6 +43,6 @@ struct wm8994_access_mask { }; extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE]; -extern const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; +extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE]; #endif -- 1.7.2.3

1 0

[alsa-devel] New asoc-tools repository
by Dimitris Papastamos 07 Mar '11

07 Mar '11

Hi folks, I've set up a new asoc-tools repository for various tools/scripts specific to ASoC. So far I've got two scripts, one for visualizing DAPM graphs and rendering them to jpeg and another tool for beautifying the output of the tracepoint hooks of ASoC so that it is more verbose. Information on how to use these scripts is self-contained in the code. The repo is at git://opensource.wolfsonmicro.com/asoc-tools.git Try them out, any feedback is highly appreciated. Thanks, Dimitris

1 0

[alsa-devel] [PATCH] ASoC: sound:soc:soc-core fix one small indentation issue
by Lu Guanqun 07 Mar '11

07 Mar '11

Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- sound/soc/soc-core.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 64befac..f6ba502 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3438,7 +3438,7 @@ int snd_soc_register_platform(struct device *dev, platform = kzalloc(sizeof(struct snd_soc_platform), GFP_KERNEL); if (platform == NULL) - return -ENOMEM; + return -ENOMEM; /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); -- 1.7.2.3

3 2

[alsa-devel] [RFC][PATCH] ASoC: Move symmetric rate check for substreams to hw_params
by Vaibhav Bedia 07 Mar '11

07 Mar '11

Currently the symmetric rate check is present in the snd_pcm_open() call. However this causes snd_pcm_open() to fail for the 2nd substream is the 2nd call to snd_pcm_open() is made before the 1st substream has set a non-zero rate by calling snd_pcm_hw_params(). Fix this by moving the call to snd_pcm_apply_symmetry() to soc_pcm_hw_params(). Signed-off-by: Afzal Mohammed <afzal(a)ti.com> Signed-off-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- This patch is one way of fixing the issues reported here http://comments.gmane.org/gmane.linux.alsa.devel/75489 and http://thread.gmane.org/gmane.linux.alsa.devel/81055/focus=21653 sound/soc/soc-core.c | 25 +++++++++++++++++-------- 1 files changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 64befac..edec9af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -638,13 +638,6 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } - /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); - if (ret != 0) - goto config_err; - } - pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); @@ -742,6 +735,9 @@ static int soc_codec_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if(!codec_dai->active && !cpu_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ @@ -898,12 +894,25 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* Symmetry only applies if we've already got an active stream, + a non zero rtd->rate implies that stream was already active and + rtd->rate has been set previously */ + if (rtd->rate) { + ret = soc_pcm_apply_symmetry(substream); + if (ret < 0) + goto symmetry_err; + } else { + rtd->rate = params_rate(params); + } out: mutex_unlock(&pcm_mutex); return ret; +symmetry_err: + if(platform->driver->ops->hw_free) + platform->driver->ops->hw_free(substream); + platform_err: if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); -- 1.6.2.4

3 3

[alsa-devel] MIDI, snd-serial-u16550 and dropped notes
by r10kindsofpeople 07 Mar '11

07 Mar '11

Sorry for cross-posting, I've been writing about this problem on alsa-user, but it seems to me that this probably belongs over here now (if it hasn't all along). A brief recap: Using the snd-serial-u16550 driver I have two MIDI streams coming in that are near duplicates of each other. Every now and then (after 5-45 minutes of playing) I get a dropped or corrupted message. I've captured this multiple times using both a logic analyzer on the serial data and software logging of the ALSA ports. Sometimes whole events get dropped, sometimes the tail end of one message and the beginning of the next gets dropped to form a completely different event. Aside from the fact that both ports are getting hit simultaneously, the ports are not being hit very hard...normal playing. This has occurred under: Ubuntu 10.04 netbook Ubuntu 10.10 netbook Ubuntu 10.10 with Natty lowlatency kernel It does not appear to occur under: Ubuntu 10.10 server Ubuntu 10.04 netbook with linux-rt kernel I've not eliminated the possibility that it's being caused by my ALSA application - quite the opposite - but what could I be doing that corrupts the behavior of the driver? I've tried several methods of reading the data to no effect. With great respect to the authors of the driver, I don't think it has changed in many years and kernel has certainly changed underneath it. Is there any sense that it could be improved to eliminate the problem and the apparent requirement for the -rt kernel? Or should I be looking at my ALSA application more closely? John

1 0

[alsa-devel] [PATCH] ASoC: Add LM4857 to SND_SOC_ALL_CODECS
by Mark Brown 07 Mar '11

07 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 212859c..10f11dd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -31,6 +31,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC if SOC_JZ4740 + select SND_SOC_LM4857 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Add machine driver for Visstrim_M10 board.
by Javier Martin 07 Mar '11

07 Mar '11

Visstrim_M10 boards have an external tlcv320aic3205 codec attached to SSI1. This driver glues together both interfaces. External amplifier is not supported in this first version. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/Kconfig | 10 +++ sound/soc/imx/Makefile | 2 + sound/soc/imx/mx27vis-aic32x4.c | 137 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 149 insertions(+), 0 deletions(-) create mode 100644 sound/soc/imx/mx27vis-aic32x4.c diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 642270a..15e7c04 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -30,6 +30,16 @@ config SND_MXC_SOC_WM1133_EV1 Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. +config SND_SOC_MX27VIS_AIC32X4 + tristate "SoC audio support for Visstrim M10 boards" + depends on MACH_IMX27_VISSTRIM_M10 + select SND_SOC_TVL320AIC32X4 + select SND_MXC_SOC_SSI + select SND_MXC_SOC_MX2 + help + Say Y if you want to add support for SoC audio on Visstrim SM10 + board with TLV320AIC32X4 codec. + config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index b67fc02..d6d609b 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -10,8 +10,10 @@ obj-$(CONFIG_SND_MXC_SOC_MX2) += snd-soc-imx-mx2.o # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o +snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o +obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c new file mode 100644 index 0000000..054110b --- /dev/null +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -0,0 +1,137 @@ +/* + * mx27vis-aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin <javier.martin(a)vista-silicon.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. + * + * 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 Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/mach-types.h> +#include <mach/audmux.h> + +#include "../codecs/tlv320aic32x4.h" +#include "imx-ssi.h" + +static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + u32 dai_format; + + dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + /* set codec DAI configuration */ + snd_soc_dai_set_fmt(codec_dai, dai_format); + + /* set cpu DAI configuration */ + snd_soc_dai_set_fmt(cpu_dai, dai_format); + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + 25000000, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { + .hw_params = mx27vis_aic32x4_hw_params, +}; + +static struct snd_soc_dai_link mx27vis_aic32x4_dai = { + .name = "tlv320aic32x4", + .stream_name = "TLV320AIC32X4", + .codec_dai_name = "tlv320aic32x4-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "tlv320aic32x4.0-0018", + .cpu_dai_name = "imx-ssi.0", + .ops = &mx27vis_aic32x4_snd_ops, +}; + +static struct snd_soc_card mx27vis_aic32x4 = { + .name = "visstrim_m10-audio", + .dai_link = &mx27vis_aic32x4_dai, + .num_links = 1, +}; + +static struct platform_device *mx27vis_aic32x4_snd_device; + +static int __init mx27vis_aic32x4_init(void) +{ + int ret; + + mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1); + if (!mx27vis_aic32x4_snd_device) + return -ENOMEM; + + platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4); + ret = platform_device_add(mx27vis_aic32x4_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(mx27vis_aic32x4_snd_device); + } + + /* Connect SSI0 as clock slave to SSI1 external pins */ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_TFSDIR | + MXC_AUDMUX_V1_PCR_TCLKDIR | + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1, + MXC_AUDMUX_V1_PCR_SYN | + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + + return ret; +} + +static void __exit mx27vis_aic32x4_exit(void) +{ + platform_device_unregister(mx27vis_aic32x4_snd_device); +} + +module_init(mx27vis_aic32x4_init); +module_exit(mx27vis_aic32x4_exit); + +MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); +MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim"); +MODULE_LICENSE("GPL"); -- 1.7.0.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: tlv320dac33: add MODULE_DEVICE_TABLE
by Axel Lin 07 Mar '11

07 Mar '11

The device table is required to load modules based on modaliases. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320dac33.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 71d7be8..00b6d87 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1615,6 +1615,7 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }, { }, }; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { -- 1.7.2

6 7

Re: [alsa-devel] Process of sumit the Audio driver code to ALSA
by Reddy, MR Swami 07 Mar '11

07 Mar '11

Thanks for quick response! Yes, it ASOC based and currently, the audio drivers supported for Samsung 6410 target. Thanks Swami -----Original Message----- From: Takashi Iwai [mailto:tiwai@suse.de] Sent: Wednesday, January 19, 2011 12:39 PM To: Reddy, MR Swami Subject: Re: Process of sumit the Audio driver code to ALSA At Tue, 18 Jan 2011 20:58:28 -0800, Reddy, MR Swami wrote: > > Hello, > We (National Semi) have developed ALSA based audio drivers for our Audio subsystem. Could you please let me know the process to submit and include as part of ALSA project. > > The ALSA audio driver version info as follows: > - Linux - 2.6.24 base > - Andorid- 2.1 base Is this based on ASoC? If not, I recommend you to port the code to ASoC, thus it's the standard for embedded devices. If you have any questions, just post to alsa-devel ML. thanks, Takashi

3 3

[alsa-devel] [PATCH 1/2] ASoC: Fix outdated API usage in tlv320aic32x4
by Mark Brown 07 Mar '11

07 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic32x4.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ee82e38..e93b9d1 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -340,13 +340,13 @@ static inline int aic32x4_get_divs(int mclk, int rate) static int aic32x4_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, - ARRAY_SIZE(aic32x4_dapm_widgets)); + snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); - snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes, ARRAY_SIZE(aic32x4_dapm_routes)); - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(&codec->dapm); return 0; } @@ -602,7 +602,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } -- 1.7.2.3

2 2

[alsa-devel] [PATCH] ASoC: Extend range of supported sample rates for CS4271 CODEC.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Extend range of supported sample rates for CS4271 CODEC. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 1791796..e5cc4da 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -33,6 +33,7 @@ #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000 /* * CS4271 registers @@ -392,14 +393,14 @@ static struct snd_soc_dai_driver cs4271_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = CS4271_PCM_RATES, .formats = CS4271_PCM_FORMATS, }, .ops = &cs4271_dai_ops,

3 2

[alsa-devel] [PATCH] ASoC: Remove warnings in ep93xx-i2s.c
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Remove warnings in ep93xx-i2s.c Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 3d4c086..042f4e9 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -313,10 +313,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } static int ep93xx_i2s_resume(struct snd_soc_dai *dai) @@ -324,10 +326,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } #else #define ep93xx_i2s_suspend NULL

3 2

[alsa-devel] [PATCH v2] intel8x0m: wait a bit before warm reset check
by Paul Bolle 07 Mar '11

07 Mar '11

At every resume a laptop I use prints this message (at KERN_ERR level): ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2] The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what seems to be happening is that the register involved indicated a warm reset for some time (as the ICH_AC97WARM bit was set) but by the time the warning is printed, and that same register is checked again, that bit is already cleared and only the ICH_AC97COLD bit is still set. It turns out a warm reset needs some time to settle, but it is currently checked right away. The test therefore fails the first time it is done and schedule_timeout_uninterruptible() will be called. Once we return from that jiffies is already (far) past end_time on this laptop, so we exit the loop, print a warning, and exit the function while the warm reset actually succeeded. One way to fix this is to call udelay() after writing to the register involved. A handful of tests suggests 500 usecs is a safe value. (This might punish the "finish cold reset" case, but on this laptop such a cold reset apparently never happens, so I can't say for sure.) While we're at it drop the extra single tick from end_time, as it looks rather silly. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) v1 was called " intel8x0m: schedule timeout before warm reset check". 1) A udelay of 250 usecs always failed while 300 usecs always worked. So 500 usecs might be a bit cautious. sound/pci/intel8x0m.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 13cec1e..62fad24 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -894,7 +894,8 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) /* finish cold or do warm reset */ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; iputdword(chip, ICHREG(GLOB_CNT), cnt); - end_time = (jiffies + (HZ / 4)) + 1; + udelay(500); + end_time = jiffies + HZ / 4; do { if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0) goto __ok; -- 1.7.4.1

3 4

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Use snd_soc_dapm_context as required
by Dimitris Papastamos 07 Mar '11

07 Mar '11

Fix compilation issues by using the snd_soc_dapm_context pointer whenever it is required. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic32x4.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ee82e38..c6c3520 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -340,13 +340,15 @@ static inline int aic32x4_get_divs(int mclk, int rate) static int aic32x4_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, aic32x4_dapm_widgets, ARRAY_SIZE(aic32x4_dapm_widgets)); - snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + snd_soc_dapm_add_routes(dapm, aic32x4_dapm_routes, ARRAY_SIZE(aic32x4_dapm_routes)); - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(dapm); return 0; } @@ -602,7 +604,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: Improve EP93xx I2S clocks management.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Improve EP93xx I2S clocks management. Some freqs values are set not exact as they requested for MCLK and original code was not able to find divisors for SCLK and LRCLK. This code just picks up nearest value from 3 possible variants. This patch makes 44100 and 192000 rates working and fixes capture function. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 19 +++++++++---------- 1 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 0458dd4..3d4c086 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); unsigned word_len, div, sdiv, lrdiv; - int found = 0, err; + int err; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, * the codec uses. */ div = clk_get_rate(info->mclk) / params_rate(params); - for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) - if (sdiv * lrdiv == div) { - found = 1; - goto out; - } -out: - if (!found) - return -EINVAL; + sdiv = 4; + if (div > (256 + 512) / 2) { + lrdiv = 128; + } else { + lrdiv = 64; + if (div < (128 + 256) / 2) + sdiv = 2; + } err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); if (err)

2 2

[alsa-devel] hda : computer using EAPD as external amplifier controller
by Anisse Astier 06 Mar '11

06 Mar '11

Hi, I have this computer, with an integrated speaker. It's using an ALC662, see attached alsa-info output. By default, speaker doesn't work. Now, after playing a bit with hda-analyzer, I figured that output PIN 0x14 has EAPD enabled, which is the default way of doing things in auto configure mode. But disabling EAPD on this PIN makes speaker work. See second alsa-info attached. Is it necessary to write a full blown quirk just to modify this parameter? From what I've seen in patch_realtek, it seems we need to write the whole config for the device when adding a quirk: we can't rely on auto configuration and then put a small quirk on top of that. Am I right ? Regards, Anisse

4 6

[alsa-devel] 2.6.37, snd-hda-intel: BUG: unable to handle kernel paging request at ffffc90011c08000
by Paul Menzel 06 Mar '11

06 Mar '11

Dear ALSA folks, is the following oops already known to you [1]? ** Kernel log: [ 41.265636] parport0: PC-style at 0x378, irq 7 [PCSPP,TRISTATE] [ 41.680357] HDA Intel 0000:20:01.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 [ 41.680444] HDA Intel 0000:20:01.0: setting latency timer to 64 [ 41.680462] BUG: unable to handle kernel paging request at ffffc90011c08000 [ 41.680617] IP: [<ffffffffa0578402>] azx_probe+0x3ad/0x86b [snd_hda_intel] [ 41.680728] PGD 13781a067 PUD 13781b067 PMD 1300ba067 PTE 800000fd00000173 [ 41.680956] Oops: 0009 [#1] SMP [ 41.681098] last sysfs file: /sys/module/snd_pcm/initstate [ 41.681159] CPU 0 [ 41.681203] Modules linked in: snd_hda_intel(+) snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_midi snd_rawmidi snd_seq_midi_event tpm_tis tpm snd_seq tpm_bios psmouse parport_pc snd_timer snd_seq_device parport processor evdev snd i2c_viapro thermal_sys amd64_edac_mod k8temp i2c_core soundcore shpchp pcspkr serio_raw asus_atk0110 pci_hotplug edac_core button snd_page_alloc edac_mce_amd ext3 jbd mbcache sha256_generic cryptd aes_x86_64 aes_generic cbc dm_crypt dm_mod raid1 md_mod usbhid hid sg sd_mod crc_t10dif sr_mod cdrom ata_generic uhci_hcd sata_via pata_via libata ehci_hcd usbcore scsi_mod via_rhine mii nls_base [last unloaded: scsi_wait_scan] [ 41.684180] [ 41.684180] Pid: 1153, comm: work_for_cpu Not tainted 2.6.37-1-amd64 #1 M2V-MX SE/System Product Name [ 41.684180] RIP: 0010:[<ffffffffa0578402>] [<ffffffffa0578402>] azx_probe+0x3ad/0x86b [snd_hda_intel] [ 41.684180] RSP: 0018:ffff88013153fe50 EFLAGS: 00010286 [ 41.684180] RAX: ffffc90011c08000 RBX: ffff88013029ec00 RCX: 0000000000000006 [ 41.684180] RDX: 0000000000000000 RSI: 0000000000000246 RDI: 0000000000000246 [ 41.684180] RBP: ffff88013341d000 R08: 0000000000000000 R09: 0000000000000040 [ 41.684180] R10: 0000000000000286 R11: 0000000000003731 R12: ffff88013029c400 [ 41.684180] R13: 0000000000000000 R14: 0000000000000000 R15: ffff88013341d090 [ 41.684180] FS: 0000000000000000(0000) GS:ffff8800bfc00000(0000) knlGS:00000000f7610ab0 [ 41.684180] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 41.684180] CR2: ffffc90011c08000 CR3: 0000000132f57000 CR4: 00000000000006f0 [ 41.684180] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 41.684180] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 41.684180] Process work_for_cpu (pid: 1153, threadinfo ffff88013153e000, task ffff8801303c86c0) [ 41.684180] Stack: [ 41.684180] 0000000000000005 ffffffff8123ad65 00000000000136c0 ffff88013029c400 [ 41.684180] ffff8801303c8998 ffff88013341d000 ffff88013341d090 ffff8801322d9dc8 [ 41.684180] ffff88013341d208 0000000000000000 0000000000000000 ffffffff811ad232 [ 41.684180] Call Trace: [ 41.684180] [<ffffffff8123ad65>] ? __pm_runtime_set_status+0x162/0x186 [ 41.684180] [<ffffffff811ad232>] ? local_pci_probe+0x49/0x92 [ 41.684180] [<ffffffff8105afc5>] ? do_work_for_cpu+0x0/0x1b [ 41.684180] [<ffffffff8105afc5>] ? do_work_for_cpu+0x0/0x1b [ 41.684180] [<ffffffff8105afd0>] ? do_work_for_cpu+0xb/0x1b [ 41.684180] [<ffffffff8105fd3f>] ? kthread+0x7a/0x82 [ 41.684180] [<ffffffff8100a824>] ? kernel_thread_helper+0x4/0x10 [ 41.684180] [<ffffffff8105fcc5>] ? kthread+0x0/0x82 [ 41.684180] [<ffffffff8100a820>] ? kernel_thread_helper+0x0/0x10 [ 41.684180] Code: f4 01 00 00 ef 31 f6 48 89 df e8 29 dd ff ff 85 c0 0f 88 2b 03 00 00 48 89 ef e8 b4 39 c3 e0 8b 7b 40 e8 fc 9d b1 e0 48 8b 43 38 <66> 8b 10 66 89 14 24 8b 43 14 83 e8 03 83 f8 01 77 32 31 d2 be [ 41.684180] RIP [<ffffffffa0578402>] azx_probe+0x3ad/0x86b [snd_hda_intel] [ 41.684180] RSP <ffff88013153fe50> [ 41.684180] CR2: ffffc90011c08000 [ 41.684180] ---[ end trace 8d1f3ebc136437fd ]--- [ 42.127750] input: ImPS/2 Logitech Wheel Mouse as /devices/platform/i8042/serio1/input/input5 [ 63.455908] EXT3-fs (dm-1): using internal journal Searching the Web a comment to a report [2] says that such problems are hard to debug. Can you give me any hints? I will try to provide the output of `alsa-info.sh` in some hours. Thanks, Paul [1] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=613979 [2] https://bugzilla.kernel.org/show_bug.cgi?id=9305

1 1

[alsa-devel] [PATCH] ASoC: Enable 192kHz sample rate for EP92xx.
by Alexander Sverdlin 05 Mar '11

05 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Enable 192kHz sample rate for EP92xx. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index e1a58d3..87dc1ee 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -351,13 +351,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .ops = &ep93xx_i2s_dai_ops, diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 0667077..a456e49 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_96000, + .rate_max = SNDRV_PCM_RATE_192000, .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |

2 2

[alsa-devel] [PATCH] add new Fermi 5xx codec IDs to snd-hda
by Richard Samson 05 Mar '11

05 Mar '11

Hello, It's my first patch/post here, i'm not sure if this list is the right place to suggest this patch. To follow up a previous commit on snd-hda : http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commitdiff;… I added a new entry in patch_hdmi.c to get sound working on my own 560 Ti card : --- patch_hdmi.c 2011-02-15 22:29:51.000000000 +0100 +++ patch_hdmi_GTX-560Ti.c 2011-02-27 23:52:56.984613414 +0100 @@ -1634,6 +1634,7 @@ { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, @@ -1676,6 +1677,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0012"); MODULE_ALIAS("snd-hda-codec-id:10de0013"); MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0016"); MODULE_ALIAS("snd-hda-codec-id:10de0018"); MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); I think other 5xx cards need new value also. Regards, --- Richard Samson samson.richard(a)gmail.com

4 11

[alsa-devel] [PATCH] ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.
by Alexander Sverdlin 05 Mar '11

05 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx machine driver. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/edb93xx.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index b270085..d3aa151 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -41,17 +41,17 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; + unsigned int mclk_rate; unsigned int rate = params_rate(params); + /* - * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, - * because our sample size is 32 bit * 2 channels. - * I2S standard permits us to transmit more bits than - * the codec uses. - * MCLK = SCLK * 4 is the best recommended value, - * but we have to fall back to ratio 2 for higher - * sample rates. + * According to CS4271 datasheet we use MCLK/LRCK=256 for + * rates below 50kHz and 128 for higher sample rates */ - unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + if (rate < 50000) + mclk_rate = rate * 64 * 4; + else + mclk_rate = rate * 64 * 2; err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |

1 0

[alsa-devel] [PATCH] ASoC: Manage mode and rate bits correctly for CS4271 CODEC.
by Alexander Sverdlin 05 Mar '11

05 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC. This is done to make capture work properly. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 83 ++++++++++++++++++++++++++++---------------- 1 files changed, 53 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e5cc4da..c71ad08 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -168,27 +168,6 @@ struct cs4271_private { int gpio_disable; }; -struct cs4271_clk_cfg { - unsigned int ratio; /* MCLK / sample rate */ - u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ - u8 mclk_master; /* ratio bit mask for Master mode */ - u8 mclk_slave; /* ratio bit mask for Slave mode */ -}; - -static struct cs4271_clk_cfg cs4271_clk_tab[] = { - {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, - {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, - {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} -}; - -#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) - /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -297,6 +276,45 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, return cs4271_set_deemph(codec); } +struct cs4271_clk_cfg { + bool master; /* codec mode */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + unsigned short ratio; /* MCLK / sample rate */ + u8 ratio_mask; /* ratio bit mask for Master mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {1, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_3}, + {0, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 512, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 256, CS4271_MODE1_DIV_2}, +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -308,23 +326,28 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, unsigned int ratio, val; cs4271->rate = params_rate(params); + + /* Configure DAC */ + if (cs4271->rate < 50000) + val = CS4271_MODE1_MODE_1X; + else if (cs4271->rate < 100000) + val = CS4271_MODE1_MODE_2X; + else + val = CS4271_MODE1_MODE_4X; + ratio = cs4271->mclk / cs4271->rate; for (i = 0; i < CS4171_NR_RATIOS; i++) - if (cs4271_clk_tab[i].ratio == ratio) + if ((cs4271_clk_tab[i].master == cs4271->master) && + (cs4271_clk_tab[i].speed_mode == val) && + (cs4271_clk_tab[i].ratio == ratio)) break; - if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + if (i == CS4171_NR_RATIOS) { dev_err(codec->dev, "Invalid sample rate\n"); return -EINVAL; } - /* Configure DAC */ - val = cs4271_clk_tab[i].speed_mode; - - if (cs4271->master) - val |= cs4271_clk_tab[i].mclk_master; - else - val |= cs4271_clk_tab[i].mclk_slave; + val |= cs4271_clk_tab[i].ratio_mask; ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);

1 0

[alsa-devel] [PATCH v3] ASoC: Add TI tlv320aic32x4 codec support.
by Javier Martin 04 Mar '11

04 Mar '11

This patch adds support for tlv320aic3205 and tlv320aic3254 codecs. It doesn't include miniDSP support for aic3254. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- include/sound/tlv320aic32x4.h | 31 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic32x4.c | 794 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.h | 143 +++++++ 5 files changed, 974 insertions(+), 0 deletions(-) create mode 100644 include/sound/tlv320aic32x4.h create mode 100644 sound/soc/codecs/tlv320aic32x4.c create mode 100644 sound/soc/codecs/tlv320aic32x4.h diff --git a/include/sound/tlv320aic32x4.h b/include/sound/tlv320aic32x4.h new file mode 100644 index 0000000..c009f70 --- /dev/null +++ b/include/sound/tlv320aic32x4.h @@ -0,0 +1,31 @@ +/* + * tlv320aic32x4.h -- TLV320AIC32X4 Soc Audio driver platform data + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin <javier.martin(a)vista-silicon.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. + */ + +#ifndef _AIC32X4_PDATA_H +#define _AIC32X4_PDATA_H + +#define AIC32X4_PWR_MICBIAS_2075_LDOIN 0x00000001 +#define AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE 0x00000002 +#define AIC32X4_PWR_AIC32X4_LDO_ENABLE 0x00000004 +#define AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36 0x00000008 +#define AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED 0x00000010 + +#define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K 0x00000001 +#define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K 0x00000002 + +struct aic32x4_pdata { + u32 power_cfg; + u32 micpga_routing; + bool swapdacs; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3b5690d..497c9a6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TVL320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -182,6 +183,9 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI +config SND_SOC_TVL320AIC32X4 + tristate + config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f67a2d6..5709ca6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o @@ -98,6 +99,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c new file mode 100644 index 0000000..ee82e38 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -0,0 +1,794 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin <javier.martin(a)vista-silicon.com> + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * 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. + * + * 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 Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/slab.h> + +#include <sound/tlv320aic32x4.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tlv320aic32x4.h" + +struct aic32x4_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; + u8 blck_N; +}; + +struct aic32x4_priv { + u32 sysclk; + s32 master; + u8 page_no; + void *control_data; + u32 power_cfg; + u32 micpga_routing; + bool swapdacs; +}; + +/* 0dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0); +/* 0dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0); + +static const struct snd_kcontrol_new aic32x4_snd_controls[] = { + SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL, + AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 7, 0x01, 1), + + SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0), + SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0), + + SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL, + AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5), + SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + + SOC_SINGLE("AGC Left Switch", AIC32X4_LAGC1, 7, 1, 0), + SOC_SINGLE("AGC Right Switch", AIC32X4_RAGC1, 7, 1, 0), + SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1, + 4, 0x07, 0), + SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1, + 0, 0x03, 0), + SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2, + 6, 0x03, 0), + SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2, + 1, 0x1F, 0), + SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3, + 0, 0x7F, 0), + SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6, + 0, 0x1F, 0), + SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7, + 0, 0x0F, 0), +}; + +static const struct aic32x4_rate_divs aic32x4_divs[] = { + /* 8k rate */ + {AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24}, + {AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24}, + {AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24}, + /* 11.025k rate */ + {AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16}, + {AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16}, + /* 16k rate */ + {AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12}, + {AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12}, + {AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12}, + /* 22.05k rate */ + {AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8}, + {AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8}, + {AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8}, + /* 32k rate */ + {AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6}, + {AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6}, + /* 44.1k rate */ + {AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4}, + /* 48k rate */ + {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} +}; + +static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new hpr_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_HPRROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_R Switch", AIC32X4_HPRROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new lol_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_LOLROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new lor_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0), + SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0, + &hpl_output_mixer_controls[0], + ARRAY_SIZE(hpl_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0, + &lol_output_mixer_controls[0], + ARRAY_SIZE(lol_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0), + SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0, + &hpr_output_mixer_controls[0], + ARRAY_SIZE(hpr_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0, + &lor_output_mixer_controls[0], + ARRAY_SIZE(lor_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOL"), + SND_SOC_DAPM_OUTPUT("LOR"), + SND_SOC_DAPM_INPUT("IN1_L"), + SND_SOC_DAPM_INPUT("IN1_R"), + SND_SOC_DAPM_INPUT("IN2_L"), + SND_SOC_DAPM_INPUT("IN2_R"), + SND_SOC_DAPM_INPUT("IN3_L"), + SND_SOC_DAPM_INPUT("IN3_R"), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + /* Left Output */ + {"HPL Output Mixer", "L_DAC Switch", "Left DAC"}, + {"HPL Output Mixer", "IN1_L Switch", "IN1_L"}, + + {"HPL Power", NULL, "HPL Output Mixer"}, + {"HPL", NULL, "HPL Power"}, + + {"LOL Output Mixer", "L_DAC Switch", "Left DAC"}, + + {"LOL Power", NULL, "LOL Output Mixer"}, + {"LOL", NULL, "LOL Power"}, + + /* Right Output */ + {"HPR Output Mixer", "R_DAC Switch", "Right DAC"}, + {"HPR Output Mixer", "IN1_R Switch", "IN1_R"}, + + {"HPR Power", NULL, "HPR Output Mixer"}, + {"HPR", NULL, "HPR Power"}, + + {"LOR Output Mixer", "R_DAC Switch", "Right DAC"}, + + {"LOR Power", NULL, "LOR Output Mixer"}, + {"LOR", NULL, "LOR Power"}, + + /* Left input */ + {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, + {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, + {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, + + {"Left ADC", NULL, "Left Input Mixer"}, + + /* Right Input */ + {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, + {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, + {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, + + {"Right ADC", NULL, "Right Input Mixer"}, +}; + +static inline int aic32x4_change_page(struct snd_soc_codec *codec, + unsigned int new_page) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data[2]; + int ret; + + data[0] = 0x00; + data[1] = new_page & 0xff; + + ret = codec->hw_write(codec->control_data, data, 2); + if (ret == 2) { + aic32x4->page_no = new_page; + return 0; + } else { + return ret; + } +} + +static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + u8 data[2]; + int ret; + + /* A write to AIC32X4_PSEL is really a non-explicit page change */ + if (reg == AIC32X4_PSEL) + return aic32x4_change_page(codec, val); + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + + data[0] = fixed_reg & 0xff; + data[1] = val & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + int ret; + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); +} + +static inline int aic32x4_get_divs(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { + if ((aic32x4_divs[i].rate == rate) + && (aic32x4_divs[i].mclk == mclk)) { + return i; + } + } + printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + return -EINVAL; +} + +static int aic32x4_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + ARRAY_SIZE(aic32x4_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case AIC32X4_FREQ_12000000: + case AIC32X4_FREQ_24000000: + case AIC32X4_FREQ_25000000: + aic32x4->sysclk = freq; + return 0; + } + printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + return -EINVAL; +} + +static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 iface_reg_1; + u8 iface_reg_2; + u8 iface_reg_3; + + iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); + iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); + iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); + iface_reg_2 = 0; + iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); + iface_reg_3 = iface_reg_3 & ~(1 << 3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic32x4->master = 1; + iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic32x4->master = 0; + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + iface_reg_2 = 0x01; /* add offset 1 */ + break; + case SND_SOC_DAIFMT_DSP_B: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg_1 |= + (AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg_1 |= + (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1); + snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2); + snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3); + return 0; +} + +static int aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data; + int i; + + i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); + if (i < 0) { + printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + return i; + } + + /* Use PLL as CODEC_CLKIN and DAC_MOD_CLK as BDIV_CLKIN */ + snd_soc_write(codec, AIC32X4_CLKMUX, AIC32X4_PLLCLKIN); + snd_soc_write(codec, AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK); + + /* We will fix R value to 1 and will make P & J=K.D as varialble */ + data = snd_soc_read(codec, AIC32X4_PLLPR); + data &= ~(7 << 4); + snd_soc_write(codec, AIC32X4_PLLPR, + (data | (aic32x4_divs[i].p_val << 4) | 0x01)); + + snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j); + + snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8)); + snd_soc_write(codec, AIC32X4_PLLDLSB, + (aic32x4_divs[i].pll_d & 0xff)); + + /* NDAC divider value */ + data = snd_soc_read(codec, AIC32X4_NDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac); + + /* MDAC divider value */ + data = snd_soc_read(codec, AIC32X4_MDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac); + + /* DOSR MSB & LSB values */ + snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); + snd_soc_write(codec, AIC32X4_DOSRLSB, + (aic32x4_divs[i].dosr & 0xff)); + + /* NADC divider value */ + data = snd_soc_read(codec, AIC32X4_NADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc); + + /* MADC divider value */ + data = snd_soc_read(codec, AIC32X4_MADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc); + + /* AOSR value */ + snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr); + + /* BCLK N divider */ + data = snd_soc_read(codec, AIC32X4_BCLKN); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N); + + data = snd_soc_read(codec, AIC32X4_IFACE1); + data = data & ~(3 << 4); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT); + break; + } + snd_soc_write(codec, AIC32X4_IFACE1, data); + + return 0; +} + +static int aic32x4_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dac_reg; + + dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON; + if (mute) + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON); + else + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg); + return 0; +} + +static int aic32x4_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 value; + + switch (level) { + case SND_SOC_BIAS_ON: + if (aic32x4->master) { + /* Switch on PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value | AIC32X4_PLLEN)); + + /* Switch on NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value | AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value | AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value | AIC32X4_MDACEN); + + /* Switch on MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value | AIC32X4_MDACEN); + + /* Switch on BCLK_N Divider */ + value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_write(codec, AIC32X4_BCLKN, + value | AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (aic32x4->master) { + /* Switch off PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value & ~AIC32X4_PLLEN)); + + /* Switch off NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value & ~AIC32X4_NDACEN); + + /* Switch off MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value & ~AIC32X4_MDACEN); + + /* Switch off NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value & ~AIC32X4_NDACEN); + + /* Switch off MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value & ~AIC32X4_MDACEN); + value = snd_soc_read(codec, AIC32X4_BCLKN); + + /* Switch off BCLK_N Divider */ + snd_soc_write(codec, AIC32X4_BCLKN, + value & ~AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops aic32x4_ops = { + .hw_params = aic32x4_hw_params, + .digital_mute = aic32x4_mute, + .set_fmt = aic32x4_set_dai_fmt, + .set_sysclk = aic32x4_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver aic32x4_dai = { + .name = "tlv320aic32x4-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .ops = &aic32x4_ops, + .symmetric_rates = 1, +}; + +static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic32x4_resume(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic32x4_probe(struct snd_soc_codec *codec) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u32 tmp_reg; + + codec->hw_write = (hw_write_t) i2c_master_send; + codec->control_data = aic32x4->control_data; + + snd_soc_write(codec, AIC32X4_RESET, 0x01); + + /* Power platform configuration */ + if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { + snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | + AIC32X4_MICBIAS_2075V); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { + snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); + } + if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) { + snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN); + } + tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE); + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) { + tmp_reg |= AIC32X4_LDOIN_18_36; + } + if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) { + tmp_reg |= AIC32X4_LDOIN2HP; + } + snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg); + + /* Do DACs need to be swapped? */ + if (aic32x4->swapdacs) { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN); + } else { + snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN); + } + + /* Mic PGA routing */ + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) { + snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K); + } + if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) { + snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K); + } + + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_add_controls(codec, aic32x4_snd_controls, + ARRAY_SIZE(aic32x4_snd_controls)); + aic32x4_add_widgets(codec); + + return 0; +} + +static int aic32x4_remove(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { + .read = aic32x4_read, + .write = aic32x4_write, + .probe = aic32x4_probe, + .remove = aic32x4_remove, + .suspend = aic32x4_suspend, + .resume = aic32x4_resume, + .set_bias_level = aic32x4_set_bias_level, +}; + +static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic32x4_pdata *pdata = i2c->dev.platform_data; + struct aic32x4_priv *aic32x4; + int ret; + + aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL); + if (aic32x4 == NULL) + return -ENOMEM; + + aic32x4->control_data = i2c; + i2c_set_clientdata(i2c, aic32x4); + + if (pdata) { + aic32x4->power_cfg = pdata->power_cfg; + aic32x4->swapdacs = pdata->swapdacs; + aic32x4->micpga_routing = pdata->micpga_routing; + } else { + aic32x4->power_cfg = 0; + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_aic32x4, &aic32x4_dai, 1); + if (ret < 0) + kfree(aic32x4); + return ret; +} + +static __devexit int aic32x4_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4", + .owner = THIS_MODULE, + }, + .probe = aic32x4_i2c_probe, + .remove = __devexit_p(aic32x4_i2c_remove), + .id_table = aic32x4_i2c_id, +}; + +static int __init aic32x4_modinit(void) +{ + int ret = 0; + + ret = i2c_add_driver(&aic32x4_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", + ret); + } + return ret; +} +module_init(aic32x4_modinit); + +static void __exit aic32x4_exit(void) +{ + i2c_del_driver(&aic32x4_i2c_driver); +} +module_exit(aic32x4_exit); + +MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); +MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h new file mode 100644 index 0000000..aae2b24 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -0,0 +1,143 @@ +/* + * tlv320aic32x4.h + * + * 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. + */ + + +#ifndef _TLV320AIC32X4_H +#define _TLV320AIC32X4_H + +/* tlv320aic32x4 register space (in decimal to match datasheet) */ + +#define AIC32X4_PAGE1 128 + +#define AIC32X4_PSEL 0 +#define AIC32X4_RESET 1 +#define AIC32X4_CLKMUX 4 +#define AIC32X4_PLLPR 5 +#define AIC32X4_PLLJ 6 +#define AIC32X4_PLLDMSB 7 +#define AIC32X4_PLLDLSB 8 +#define AIC32X4_NDAC 11 +#define AIC32X4_MDAC 12 +#define AIC32X4_DOSRMSB 13 +#define AIC32X4_DOSRLSB 14 +#define AIC32X4_NADC 18 +#define AIC32X4_MADC 19 +#define AIC32X4_AOSR 20 +#define AIC32X4_CLKMUX2 25 +#define AIC32X4_CLKOUTM 26 +#define AIC32X4_IFACE1 27 +#define AIC32X4_IFACE2 28 +#define AIC32X4_IFACE3 29 +#define AIC32X4_BCLKN 30 +#define AIC32X4_IFACE4 31 +#define AIC32X4_IFACE5 32 +#define AIC32X4_IFACE6 33 +#define AIC32X4_DOUTCTL 53 +#define AIC32X4_DINCTL 54 +#define AIC32X4_DACSPB 60 +#define AIC32X4_ADCSPB 61 +#define AIC32X4_DACSETUP 63 +#define AIC32X4_DACMUTE 64 +#define AIC32X4_LDACVOL 65 +#define AIC32X4_RDACVOL 66 +#define AIC32X4_ADCSETUP 81 +#define AIC32X4_ADCFGA 82 +#define AIC32X4_LADCVOL 83 +#define AIC32X4_RADCVOL 84 +#define AIC32X4_LAGC1 86 +#define AIC32X4_LAGC2 87 +#define AIC32X4_LAGC3 88 +#define AIC32X4_LAGC4 89 +#define AIC32X4_LAGC5 90 +#define AIC32X4_LAGC6 91 +#define AIC32X4_LAGC7 92 +#define AIC32X4_RAGC1 94 +#define AIC32X4_RAGC2 95 +#define AIC32X4_RAGC3 96 +#define AIC32X4_RAGC4 97 +#define AIC32X4_RAGC5 98 +#define AIC32X4_RAGC6 99 +#define AIC32X4_RAGC7 100 +#define AIC32X4_PWRCFG (AIC32X4_PAGE1 + 1) +#define AIC32X4_LDOCTL (AIC32X4_PAGE1 + 2) +#define AIC32X4_OUTPWRCTL (AIC32X4_PAGE1 + 9) +#define AIC32X4_CMMODE (AIC32X4_PAGE1 + 10) +#define AIC32X4_HPLROUTE (AIC32X4_PAGE1 + 12) +#define AIC32X4_HPRROUTE (AIC32X4_PAGE1 + 13) +#define AIC32X4_LOLROUTE (AIC32X4_PAGE1 + 14) +#define AIC32X4_LORROUTE (AIC32X4_PAGE1 + 15) +#define AIC32X4_HPLGAIN (AIC32X4_PAGE1 + 16) +#define AIC32X4_HPRGAIN (AIC32X4_PAGE1 + 17) +#define AIC32X4_LOLGAIN (AIC32X4_PAGE1 + 18) +#define AIC32X4_LORGAIN (AIC32X4_PAGE1 + 19) +#define AIC32X4_HEADSTART (AIC32X4_PAGE1 + 20) +#define AIC32X4_MICBIAS (AIC32X4_PAGE1 + 51) +#define AIC32X4_LMICPGAPIN (AIC32X4_PAGE1 + 52) +#define AIC32X4_LMICPGANIN (AIC32X4_PAGE1 + 54) +#define AIC32X4_RMICPGAPIN (AIC32X4_PAGE1 + 55) +#define AIC32X4_RMICPGANIN (AIC32X4_PAGE1 + 57) +#define AIC32X4_FLOATINGINPUT (AIC32X4_PAGE1 + 58) +#define AIC32X4_LMICPGAVOL (AIC32X4_PAGE1 + 59) +#define AIC32X4_RMICPGAVOL (AIC32X4_PAGE1 + 60) + +#define AIC32X4_FREQ_12000000 12000000 +#define AIC32X4_FREQ_24000000 24000000 +#define AIC32X4_FREQ_25000000 25000000 + +#define AIC32X4_WORD_LEN_16BITS 0x00 +#define AIC32X4_WORD_LEN_20BITS 0x01 +#define AIC32X4_WORD_LEN_24BITS 0x02 +#define AIC32X4_WORD_LEN_32BITS 0x03 + +#define AIC32X4_I2S_MODE 0x00 +#define AIC32X4_DSP_MODE 0x01 +#define AIC32X4_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC32X4_LEFT_JUSTIFIED_MODE 0x03 + +#define AIC32X4_AVDDWEAKDISABLE 0x08 +#define AIC32X4_LDOCTLEN 0x01 + +#define AIC32X4_LDOIN_18_36 0x01 +#define AIC32X4_LDOIN2HP 0x02 + +#define AIC32X4_DACSPBLOCK_MASK 0x1f +#define AIC32X4_ADCSPBLOCK_MASK 0x1f + +#define AIC32X4_PLLJ_SHIFT 6 +#define AIC32X4_DOSRMSB_SHIFT 4 + +#define AIC32X4_PLLCLKIN 0x03 + +#define AIC32X4_MICBIAS_LDOIN 0x08 +#define AIC32X4_MICBIAS_2075V 0x60 + +#define AIC32X4_LMICPGANIN_IN2R_10K 0x10 +#define AIC32X4_RMICPGANIN_IN1L_10K 0x10 + +#define AIC32X4_LMICPGAVOL_NOGAIN 0x80 +#define AIC32X4_RMICPGAVOL_NOGAIN 0x80 + +#define AIC32X4_BCLKMASTER 0x08 +#define AIC32X4_WCLKMASTER 0x04 +#define AIC32X4_PLLEN (0x01 << 7) +#define AIC32X4_NDACEN (0x01 << 7) +#define AIC32X4_MDACEN (0x01 << 7) +#define AIC32X4_NADCEN (0x01 << 7) +#define AIC32X4_MADCEN (0x01 << 7) +#define AIC32X4_BCLKEN (0x01 << 7) +#define AIC32X4_DACEN (0x03 << 6) +#define AIC32X4_RDAC2LCHN (0x02 << 2) +#define AIC32X4_LDAC2RCHN (0x02 << 4) +#define AIC32X4_LDAC2LCHN (0x01 << 4) +#define AIC32X4_RDAC2RCHN (0x01 << 2) + +#define AIC32X4_SSTEP2WCLK 0x01 +#define AIC32X4_MUTEON 0x0C +#define AIC32X4_DACMOD2BCLK 0x01 + +#endif /* _TLV320AIC32X4_H */ -- 1.7.0.4

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[PATCH 4/4] ALSA: HDA: Realtek ALC88x: Do not over-initialize speakers and hp that are primary outputs
by David Henningsson 04 Mar '11

04 Mar '11

Do not initialize again the what has already been initialized as multi outs, as this breaks surround speakers. Tested-by: Bartłomiej Żogała <nusch88(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 37 +++++++++++++++++++++---------------- 1 files changed, 21 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f94b12f..3dc88ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10843,23 +10843,28 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec) hda_nid_t pin, dac; int i; - for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { - pin = spec->autocfg.hp_pins[i]; - if (!pin) - break; - dac = spec->multiout.hp_nid; - if (!dac) - dac = spec->multiout.dac_nids[0]; /* to front */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + if (spec->autocfg.line_out_type != AUTO_PIN_HP_OUT) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.hp_pins); i++) { + pin = spec->autocfg.hp_pins[i]; + if (!pin) + break; + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; /* to front */ + alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } } - for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { - pin = spec->autocfg.speaker_pins[i]; - if (!pin) - break; - dac = spec->multiout.extra_out_nid[0]; - if (!dac) - dac = spec->multiout.dac_nids[0]; /* to front */ - alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + + if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT) { + for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { + pin = spec->autocfg.speaker_pins[i]; + if (!pin) + break; + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; /* to front */ + alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } } -- 1.7.1 --------------070904000204070302040307--

1 0

[alsa-devel] 1.0.24 alsa-driver compilation problem
by Grzegorz Daniluk 04 Mar '11

04 Mar '11

Hi, I'm trying to cross-compile the latest 1.0.24 alsa-driver for my embedded system (using SH4 processor and linux kernel 2.6.35). Before compiling 1.0.24 I did turn off the alsa in kernel configuration (leaving just Sound card support enabled). So I run as follows: ./configure --with-kernel=<where_my_kernel_src's_are> --with-build=<where_my_kernel_src's_are> --with-cross=sh4-linux-gnu- and then 'make'. However, during 'make' I've noticed the following warning: WARNING: "register_sound_special_device" [/root/alsa-driver-1.0.24/acore/snd.ko] undefined! WARNING: "unregister_sound_special" [/root/alsa-driver-1.0.24/acore/snd.ko] undefined! and then when trying to insmod the snd.ko module the following error appears: snd: Unknown symbol unregister_sound_special (err 0) snd: Unknown symbol register_sound_special_device (err 0) thank you for your help, best regards, Greg

2 2

[PATCH 3/4] ALSA: HDA: Fixup unnecessary volume control index on Realtek ALC88x
by David Henningsson 04 Mar '11

04 Mar '11

Without this change, a volume control named "Surround" or "Side" would get an unnecessary index, causing it to be ignored by the vmaster and PulseAudio. Tested-by: Bartłomiej Żogała <nusch88(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 ++++++--- 1 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fe45c61..f94b12f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5206,16 +5206,19 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } else { const char *name = pfx; - if (!name) + int index = i; + if (!name) { name = chname[i]; + index = 0; + } err = __add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = __add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, - name, i, + name, index, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) -- 1.7.1 --------------070904000204070302040307 Content-Type: text/x-patch; name="0004-ALSA-HDA-Realtek-ALC88x-Do-not-over-initialize-speak.patch" Content-Transfer-Encoding: 8bit Content-Disposition: attachment; filename*0="0004-ALSA-HDA-Realtek-ALC88x-Do-not-over-initialize-speak.pa"; filename*1="tch"

1 0

[PATCH 2/4] ALSA: HDA: Fix volume control naming for surround speakers on Realtek auto-parser
by David Henningsson 04 Mar '11

04 Mar '11

When more than one pair of internal speakers is present, allow names according to their channels. Tested-by: Bartłomiej Żogała <nusch88(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ed4873c..fe45c61 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5150,7 +5150,9 @@ static const char *alc_get_line_out_pfx(const struct auto_pin_cfg *cfg, switch (cfg->line_out_type) { case AUTO_PIN_SPEAKER_OUT: - return "Speaker"; + if (cfg->line_outs == 1) + return "Speaker"; + break; case AUTO_PIN_HP_OUT: return "Headphone"; default: -- 1.7.1 --------------070904000204070302040307 Content-Type: text/x-patch; name="0003-ALSA-HDA-Fixup-unnecessary-volume-control-index-on-R.patch" Content-Transfer-Encoding: 8bit Content-Disposition: attachment; filename*0="0003-ALSA-HDA-Fixup-unnecessary-volume-control-index-on-R.pa"; filename*1="tch"

1 0

[PATCH 1/4] ALSA: HDA: Enable surround and subwoofer on Lenovo Ideapad Y530
by David Henningsson 04 Mar '11

04 Mar '11

The pin config values would change the association instead of the sequence, this commit fixes that up. Tested-by: Bartłomiej Żogała <nusch88(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f721a18..ed4873c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10747,6 +10747,7 @@ static struct alc_config_preset alc882_presets[] = { */ enum { PINFIX_ABIT_AW9D_MAX, + PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, }; @@ -10761,6 +10762,14 @@ static const struct alc_fixup alc882_fixups[] = { { } } }, + [PINFIX_LENOVO_Y530] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x15, 0x99130112 }, /* rear int speakers */ + { 0x16, 0x99130111 }, /* subwoofer */ + { } + } + }, [PINFIX_PB_M5210] = { .type = ALC_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -10776,6 +10785,7 @@ static const struct alc_fixup alc882_fixups[] = { static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", PINFIX_PB_M5210), + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), {} -- 1.7.1 --------------070904000204070302040307 Content-Type: text/x-patch; name="0002-ALSA-HDA-Fix-volume-control-naming-for-surround-spea.patch" Content-Transfer-Encoding: 8bit Content-Disposition: attachment; filename*0="0002-ALSA-HDA-Fix-volume-control-naming-for-surround-spea.pa"; filename*1="tch"

1 0

[alsa-devel] aplay: pcm_write:1603: write error: Input/output error
by asim khan 04 Mar '11

04 Mar '11

Hi, Iam using Qemu 0.14.0 and linux 2.6.33 but whenever Iam playing any wav file with aplay Iam getting error *"aplay: pcm_write:1603: write error: Input/output error*"..donno whats going wrong..I googled but not much help...its bit urgent plz update me as soon as possible. --Thanx AK

2 1

[alsa-devel] CS4206 and S/PDIF (again)
by Arun Raghavan 04 Mar '11

04 Mar '11

Hello, I saw an earlier thread where someone with a MacBook Pro 6.2 couldn't get S/PDIF working. I'm facing the same problem on my MacBook Pro 5.5. I know this used to work at some point because the LED from the digital out used to glow (but my last recollection of this is from ca. Jan 2010, so not too helpful). I tried jiggling some controls in hda-analyzer, but I don't yet know enough about this to be sure what I was doing made any sense. I've uploaded alsa-info [1] and codecgraph [2] outout, and would really appreciate help figuring out how I can go about debugging this problem. [1] http://people.collabora.co.uk/~arun/alsa-info.txt [2] http://people.collabora.co.uk/~arun/cs4206-mbp55.svg Cheers, Arun p.s.: fwiw, I'm testing with a: aplay -v -D iec958:CARD=NVidia,DEV=0 SURROUNDTEST_011212.wav

4 16

[alsa-devel] K42Jc Notebook(ALC269) port switch between headset, headphone, mic
by Madura Anushanga 03 Mar '11

03 Mar '11

Hi, I want to switch between headphone, headset, mic input/output modes when I plug in devices. Alsamixer shows two capture channels, one is for the internal mic. I don't know how to use the other. I've tried switching some things on and off in hda-analyzer but I failed to get the functionality that I wanted. This notebook only has one audio port(TRS 3.5mm), so the selection is vital if I'm going to use an external mic. Currently the port is acting as an output only port. And something fishy goes on when I select "Analog Stereo Input" in profiles in Sound Preferences. It lets me use the external mic through the audio port but only if I push the jack half way, the input is detected and I can record it(but there's no sound output after I select this profile, its input only). I some how think that this is the "mic" mode and if I can get it to work on "headset" mode then I'd be able to use it for input and output. This is the information that I got from the script, http://www.alsa-project.org/db/?f=e70dae201e5d830874536be93090aef0454c8c1e Thank you, - Madura A.

2 1

[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

4 13

[alsa-devel] [PATCH 1/5] ASoC: Get the card directly from the DAPM context
by Mark Brown 03 Mar '11

03 Mar '11

Rather than indirecting through the CODEC we can look the card up directly from the card pointer in the DAPM context. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b8e6ab7..8240ab8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -368,7 +368,7 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, int i, ret = 0; size_t name_len; struct snd_soc_dapm_path *path; - struct snd_card *card = dapm->codec->card->snd_card; + struct snd_card *card = dapm->card->snd_card; /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { @@ -430,7 +430,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; - struct snd_card *card = dapm->codec->card->snd_card; + struct snd_card *card = dapm->card->snd_card; int ret = 0; if (!w->num_kcontrols) { @@ -480,7 +480,7 @@ static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) */ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) { - int level = snd_power_get_state(widget->dapm->codec->card->snd_card); + int level = snd_power_get_state(widget->dapm->card->snd_card); switch (level) { case SNDRV_CTL_POWER_D3hot: @@ -1083,7 +1083,7 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) */ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) { - struct snd_soc_card *card = dapm->codec->card; + struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; struct snd_soc_dapm_context *d; LIST_HEAD(up_list); -- 1.7.2.3

2 5

[alsa-devel] [PATCH 1/2] ASoC: Fix broken bitfield definitions in WM8978
by Mark Brown 03 Mar '11

03 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8978.c | 14 ++++++++------ 1 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 30fb48e..7ce4f49 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -145,18 +145,18 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { SOC_SINGLE("DAC Playback Limiter Threshold", WM8978_DAC_LIMITER_2, 4, 7, 0), SOC_SINGLE("DAC Playback Limiter Boost", - WM8978_DAC_LIMITER_2, 0, 15, 0), + WM8978_DAC_LIMITER_2, 0, 12, 0), SOC_ENUM("ALC Enable Switch", alc1), SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0), SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0), - SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0), + SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0), SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0), SOC_ENUM("ALC Capture Mode", alc3), - SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0), - SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0), + SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0), + SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0), SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0), SOC_SINGLE("ALC Capture Noise Gate Threshold", @@ -211,8 +211,10 @@ static const struct snd_kcontrol_new wm8978_snd_controls[] = { WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1), /* DAC / ADC oversampling */ - SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0), - SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0), + SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, + 5, 1, 0), + SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, + 5, 1, 0), }; /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */ -- 1.7.2.3

2 2

[alsa-devel] [FIX]: No SPDIF out from MacBookPro 6, 2, since 2.6.36
by Mohammad Bahathir Hashim 03 Mar '11

03 Mar '11

Hi, Problem : No SPDIF since 2.6.36 HW : MacBookPro 15" HiRes (mid-2010) Audio : 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) HDA codec : Cirrus CS4206 This simple patch will bypass the errata workaround for CS4207, and I able to use the SPDIF (Toslink) out again. Suggestion : Make a conditional expression to load the errata verb/code if the driver detects CS4207. Thank you. --- orig/patch_cirrus.c 2011-03-03 11:56:31.726000009 +0800 +++ patch_cirrus.c 2011-03-03 12:05:38.556000063 +0800 @@ -1066,7 +1066,7 @@ struct cs_spec *spec = codec->spec; /* init_verb sequence for C0/C1/C2 errata*/ - snd_hda_sequence_write(codec, cs_errata_init_verbs); + /*snd_hda_sequence_write(codec, cs_errata_init_verbs);*/ snd_hda_sequence_write(codec, cs_coef_init_verbs);

1 0

[alsa-devel] [PATCH v21 0/3] ASoC/MFD/V4L2: WL1273 FM Radio Driver
by Matti J. Aaltonen 02 Mar '11

02 Mar '11

Hi. And thanks for the comment. Samuel wrote: >> Remove two unnecessary calls to i2c_set_clientdata. >Provided that you add a changelog relevant to the patch itself, and not to the >v1->v2 diff: Replaced the incremental changelog stuff with the original descriptions. >Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> Cheers, Matti Matti J. Aaltonen (3): MFD: WL1273 FM Radio: MFD driver for the FM radio. V4L2: WL1273 FM Radio: TI WL1273 FM radio driver ASoC: WL1273 FM radio: Access I2C IO functions through pointers. drivers/media/radio/radio-wl1273.c | 360 +++++++++++------------------------- drivers/mfd/Kconfig | 2 +- drivers/mfd/wl1273-core.c | 149 +++++++++++++++- include/linux/mfd/wl1273-core.h | 2 + sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 11 +- 6 files changed, 264 insertions(+), 262 deletions(-)

4 6

[alsa-devel] [PATCH 0/2] twl4030_codec cleanup
by Ilkka Koskinen 02 Mar '11

02 Mar '11

This series of patches removes duplicate audio_mclk fields from twl4030_codec driver. The patches have been compiled on sound-2.6/for-2.6.39 branch. I haven't tested them on any target board though. Ilkka Koskinen (2): omap: Remove unnecessary twl4030_codec_audio settings from board files mfd: twl4030_codec: Remove unused and duplicate audio_mclk fields arch/arm/mach-omap2/board-3430sdp.c | 4 +--- arch/arm/mach-omap2/board-devkit8000.c | 4 +--- arch/arm/mach-omap2/board-igep0020.c | 4 +--- arch/arm/mach-omap2/board-omap3beagle.c | 4 +--- arch/arm/mach-omap2/board-omap3evm.c | 4 +--- arch/arm/mach-omap2/board-omap3pandora.c | 4 +--- arch/arm/mach-omap2/board-omap3stalker.c | 4 +--- arch/arm/mach-omap2/board-omap3touchbook.c | 4 +--- arch/arm/mach-omap2/board-overo.c | 4 +--- arch/arm/mach-omap2/board-zoom-peripherals.c | 4 +--- include/linux/i2c/twl.h | 2 -- 11 files changed, 10 insertions(+), 32 deletions(-)

5 7

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 02 Mar '11

02 Mar '11

The following changes since commit 61a142b7e4b5c4cce1b4ea52a829984959120089: ASoC: Staticise non-exported symbols in SGTL5000 (2011-02-28 14:33:01 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Dimitris Papastamos (3): ASoC: WM8994: Don't disable the AIF[1|2]CLK_ENA unconditionaly ASoC: WM8994: Ensure late enable events are processed for the ADCs ASoC: WM8994: Ensure MICBIAS is provided with a clock Javier Martin (1): ASoC: Fix burstsize and DSP_B format problems in imx-ssi. Mark Brown (4): ASoC: Fix WM9081 platform data initialisation Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Remove module probe announcements from CODEC drivers ASoC: Add platform data for WM9081 IRQ pin configuration Stephen Warren (1): ASoC: Tegra: Fix error handling in DMA channel alloc include/sound/wm9081.h | 9 ++++-- sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/cs4270.c | 2 - sound/soc/codecs/wm8994.c | 64 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/wm9081.c | 30 +++++++++++++++----- sound/soc/imx/imx-ssi.c | 5 +++- sound/soc/tegra/tegra_pcm.c | 4 +- 7 files changed, 89 insertions(+), 26 deletions(-)

2 1

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 02 Mar '11

02 Mar '11

The following changes since commit 43c63188821dc21b2af23a40a18faea6e386e90a: eukrea-tlv320: fix platform_name (2011-02-25 12:57:49 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Dimitris Papastamos (2): ASoC: WM8994: Don't disable the AIF[1|2]CLK_ENA unconditionaly ASoC: WM8994: Ensure late enable events are processed for the ADCs Mark Brown (1): ASoC: Fix WM9081 platform data initialisation sound/soc/codecs/wm8994.c | 51 +++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/wm9081.c | 5 ++++ 2 files changed, 47 insertions(+), 9 deletions(-)

2 1

Re: [alsa-devel] Realtek ALC269VB Asus 1001px intern mic ist not working
by Paul Menzel 02 Mar '11

02 Mar '11

Dear Marty, please remember to reply to the list, e. g. in Evolution the shortcut is Ctrl + l, and to send just plain text messages to lists as you did in your original post. Am Dienstag, den 01.03.2011, 22:34 +0100 schrieb Marty: > Am 28.02.2011 20:10, schrieb Paul Menzel: > > On Mo, 2011-02-28 at 18:53 +0100, Marty wrote: > > > >> Hi, hope I get some help with this problem. At the moment I work with > >> Pulse Audio but the results are very bad. > >> > > how do you get it to work using PulseAudio? I am pasting Marty’s first private answer. Am Montag, den 28.02.2011, 21:35 +0100 schrieb Marty: > Dear Paul, as first my basic os informations. > > Host/Kernel/OS "boaconstrictor" running Linux 2.6.38-5-generic i686 [ > Kanotix Hellfire 20101213-09:07 ] > CPU Info (1) Intel Atom N450 @ clocked at [ 1667.000 MHz ] > Intel Atom N450 @ clocked at [ 1000.000 MHz ] > Videocard Intel N10 Family Integrated Graphics Controller X.Org > 1.7.7 [ 1024x600 ] > Processes 128 | Uptime 17min | Memory 258.2/2002.9MB | HDD Size 250GB > (16%used) | Client Unknown Perl client | Infobash v2.67.2 > > Advanced Linux Sound Architecture Driver Version 1.0.23. > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xf7cf8000 ir > > 1: : sequencer > 2: [ 0- 0]: digital audio playback > 3: [ 0- 0]: digital audio capture > 4: [ 0- 0]: hardware dependent > 5: [ 0] : control > 33: : timer > > 00-00: ALC269VB Analog : ALC269VB Analog : playback 1 : capture 1 > > > I tried nearly all settings in alsamixer m and i could`nt activate the > intern mic. After installing Pulse Audio with basic settings I had > either no success. > > 1. I modify /etc/modprobe.d/alsa-base.conf options snd-hda-intel model=auto > ( i did`nt tried yet options snd-hda-intel model=fujitsu options > snd-hda-intel model=lifebook) > 2. save & reboot > 3. load driver and check settings in alsamixer m > 4. Pulse Audio settings - set all the capture devices to only the left > (or right) channel > 5. Suspend to Ram > 6. reboot > > After this I had postive results in skype. The intern mic works but the > quality is worse. I modified all prossible soundsettings for better > results - no luck. > > Best & thx > Marty > >> check: > >> http://www.alsa-project.org/db/?f=0286416d03a64a2e34a4f73831c4f0f5cba002d6 > >> > > Please always additionally paste this information into the message. > > > > […] > > > > !!Kernel Information > > !!------------------ > > > > Kernel release: 2.6.38-5-generic > > Operating System: GNU/Linux > > Architecture: i686 > > Processor: unknown > > SMP Enabled: Yes > > > > > > !!ALSA Version > > !!------------ > > > > Driver version: 1.0.23 > > Library version: 1.0.23 > > Utilities version: 1.0.23 > > > > […] > > > > Could you please try a current ALSA version, i. e. 1.0.24, (for example > > using a live image) or test if some `model` parameters, e. g. `fujitsu`, > > work [1][2][3]? > > > > Searching for »1001PX alsa« also brought up several hits. Which one did > > you try? > > > > > > Thanks, > > > > Paul > > > > > > [1] http://alsa-project.org/main/index.php/Help_To_Debug > > [2] http://alsa-project.org/main/index.php/Help_To_Debug_Intel_HDA > > [3] https://wiki.archlinux.org/index.php/Asus_Eee_PC_1001px#Sound > Hi Paul, I fixed it finaly with perfect quality. > > Alsa Driver v1.0.24 > ./configure --with-oss=no > make -j2 > make install I guess `--with-oss=no` will not make a difference. But unless somebody tried we will not know. > purge Pulse Audio > reboot > modify all settings > suspend to ram > > and after this I had clear sound ! Great to hear that. Could you please confirm that it does not work with PulseAudio installed. I would imagine that upgrading ALSA driver to 1.0.24 did the trick. Thank you very much for keeping the list/me updated with your progress and success. Paul

1 0

[alsa-devel] (no subject)
by Javier Martin 02 Mar '11

02 Mar '11

This series of patches provides support for audio in Visstrim_M10 boards. This second version has some fixes in the aic32x4 codec driver as asked by Mark Brown.

3 9

[alsa-devel] [PATCH v2 2.6.38] ASoC: WM8994: Ensure late enable events are processed for the ADCs
by Dimitris Papastamos 02 Mar '11

02 Mar '11

Ensure that the ADCs are provided with a clock as the previous patch "ASoC: WM8994: Improve playback robustness" did not handle this case properly. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 26 +++++++++++++++++++++++--- 1 files changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 9e91525..4afbe3b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1096,6 +1096,13 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int adc_mux_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + late_enable_ev(w, kcontrol, event); + return 0; +} + static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1416,6 +1423,18 @@ SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; +static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = { +SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = { +SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), +SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1510,9 +1529,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), -SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), - SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), @@ -3293,11 +3309,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) if (wm8994->revision < 4) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, + ARRAY_SIZE(wm8994_adc_revd_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets, ARRAY_SIZE(wm8994_dac_revd_widgets)); } else { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, ARRAY_SIZE(wm8994_dac_widgets)); } -- 1.7.4.1

3 2

[alsa-devel] [PATCH 1/2 2.6.38] ASoC: WM8994: Don't disable the AIF[1|2]CLK_ENA unconditionaly
by Dimitris Papastamos 02 Mar '11

02 Mar '11

Since we began using the late clock disable functionality, ensure that we don't disable the clock if any of the ADC or DAC paths are still enabled. This happens when we have simultaneous playback and recording. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++------ 1 files changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ebaee5c..9e91525 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -110,6 +110,9 @@ struct wm8994_priv { unsigned int aif1clk_enable:1; unsigned int aif2clk_enable:1; + + unsigned int aif1clk_disable:1; + unsigned int aif2clk_disable:1; }; static int wm8994_readable(unsigned int reg) @@ -1015,14 +1018,18 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (wm8994->aif1clk_enable) + if (wm8994->aif1clk_enable) { snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, WM8994_AIF1CLK_ENA); - if (wm8994->aif2clk_enable) + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, WM8994_AIF2CLK_ENA); + wm8994->aif2clk_enable = 0; + } break; } @@ -1037,15 +1044,15 @@ static int late_disable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMD: - if (wm8994->aif1clk_enable) { + if (wm8994->aif1clk_disable) { snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, 0); - wm8994->aif1clk_enable = 0; + wm8994->aif1clk_disable = 0; } - if (wm8994->aif2clk_enable) { + if (wm8994->aif2clk_disable) { snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, 0); - wm8994->aif2clk_enable = 0; + wm8994->aif2clk_disable = 0; } break; } @@ -1063,6 +1070,9 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: wm8994->aif1clk_enable = 1; break; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif1clk_disable = 1; + break; } return 0; @@ -1078,6 +1088,9 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: wm8994->aif2clk_enable = 1; break; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif2clk_disable = 1; + break; } return 0; -- 1.7.4.1

3 3

[alsa-devel] [PATCH] ASoC: Remove module probe announcements from CODEC drivers
by Mark Brown 01 Mar '11

01 Mar '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/cs4270.c | 2 -- 2 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c27f8f5..cbf0b6d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -294,7 +294,6 @@ static struct spi_driver ak4104_spi_driver = { static int __init ak4104_init(void) { - pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n"); return spi_register_driver(&ak4104_spi_driver); } module_init(ak4104_init); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index c0fccad..65f578f 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -743,8 +743,6 @@ static struct i2c_driver cs4270_i2c_driver = { static int __init cs4270_init(void) { - pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n"); - return i2c_add_driver(&cs4270_i2c_driver); } module_init(cs4270_init); -- 1.7.2.3

3 3

[alsa-devel] [PATCH] ASoC: Add platform data for WM9081 IRQ pin configuration
by Mark Brown 01 Mar '11

01 Mar '11

The WM9081 IRQ output can be either active high or active low and can support either CMOS or open drain modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm9081.h | 9 ++++++--- sound/soc/codecs/wm9081.c | 29 +++++++++++++++++++---------- 2 files changed, 25 insertions(+), 13 deletions(-) diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h index e173ddb..f34b0b1 100644 --- a/include/sound/wm9081.h +++ b/include/sound/wm9081.h @@ -17,9 +17,12 @@ struct wm9081_retune_mobile_setting { u16 config[20]; }; -struct wm9081_retune_mobile_config { - struct wm9081_retune_mobile_setting *configs; - int num_configs; +struct wm9081_pdata { + bool irq_high; /* IRQ is active high */ + bool irq_cmos; /* IRQ is in CMOS mode */ + + struct wm9081_retune_mobile_setting *retune_configs; + int num_retune_configs; }; #endif diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 2103623..7883f3e 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -167,7 +167,7 @@ struct wm9081_priv { int fll_fref; int fll_fout; int tdm_width; - struct wm9081_retune_mobile_config *retune; + struct wm9081_pdata pdata; }; static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) @@ -1082,21 +1082,22 @@ static int wm9081_hw_params(struct snd_pcm_substream *substream, aif4 |= wm9081->bclk / wm9081->fs; /* Apply a ReTune Mobile configuration if it's in use */ - if (wm9081->retune) { - struct wm9081_retune_mobile_config *retune = wm9081->retune; + if (wm9081->pdata.num_retune_configs) { + struct wm9081_pdata *pdata = &wm9081->pdata; struct wm9081_retune_mobile_setting *s; int eq1; best = 0; - best_val = abs(retune->configs[0].rate - wm9081->fs); - for (i = 0; i < retune->num_configs; i++) { - cur_val = abs(retune->configs[i].rate - wm9081->fs); + best_val = abs(pdata->retune_configs[0].rate - wm9081->fs); + for (i = 0; i < pdata->num_retune_configs; i++) { + cur_val = abs(pdata->retune_configs[i].rate - + wm9081->fs); if (cur_val < best_val) { best_val = cur_val; best = i; } } - s = &retune->configs[best]; + s = &pdata->retune_configs[best]; dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n", s->name, s->rate); @@ -1255,6 +1256,14 @@ static int wm9081_probe(struct snd_soc_codec *codec) return ret; } + reg = 0; + if (wm9081->pdata.irq_high) + reg |= WM9081_IRQ_POL; + if (!wm9081->pdata.irq_cmos) + reg |= WM9081_IRQ_OP_CTRL; + snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL, + WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg); + wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ @@ -1266,7 +1275,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); - if (!wm9081->retune) { + if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, "No ReTune Mobile data, using normal EQ\n"); snd_soc_add_controls(codec, wm9081_eq_controls, @@ -1343,8 +1352,8 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, wm9081->control_data = i2c; if (dev_get_platdata(&i2c->dev)) - memcpy(&wm9081->retune, dev_get_platdata(&i2c->dev), - sizeof(wm9081->retune)); + memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), + sizeof(wm9081->pdata)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Fix WM9081 platform data initialisation
by Mark Brown 01 Mar '11

01 Mar '11

It went AWOL in the multi-component conversion. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b62e989..2103623 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -15,6 +15,7 @@ #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/device.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> @@ -1341,6 +1342,10 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, wm9081->control_type = SND_SOC_I2C; wm9081->control_data = i2c; + if (dev_get_platdata(&i2c->dev)) + memcpy(&wm9081->retune, dev_get_platdata(&i2c->dev), + sizeof(wm9081->retune)); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) -- 1.7.2.3

2 1

[alsa-devel] [PATCH] audio:am3517evm support for AIC23
by Abhilash K V 01 Mar '11

01 Mar '11

This patch aims to fix the registration of the AIC23-based audio module on the AM3517-EVM, with the following two changes: 1. The i2c_board_info entry supporting aic23 codec was added into the i2c2 bus. 2. The i2c client device name (".2-001a" in this case, including the separator period) was appended to the codec_name member of am3517evm_dai to resolve the names mismatch happening in soc_bind_dai_link(). Signed-off-by: Abhilash K V <abhilash.kv(a)ti.com> --- arch/arm/mach-omap2/board-am3517evm.c | 3 +++ sound/soc/omap/am3517evm.c | 2 +- 2 files changed, 4 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap2/board-am3517evm.c b/arch/arm/mach-omap2/board-am3517evm.c index ad71cbf..8da8d20 100644 --- a/arch/arm/mach-omap2/board-am3517evm.c +++ b/arch/arm/mach-omap2/board-am3517evm.c @@ -358,6 +358,9 @@ static struct pca953x_platform_data am3517evm_gpio_expander_info_0 = { }; static struct i2c_board_info __initdata am3517evm_i2c2_boardinfo[] = { { + I2C_BOARD_INFO("tlv320aic23", 0x1A), + }, + { I2C_BOARD_INFO("tca6416", 0x21), .platform_data = &am3517evm_gpio_expander_info_0, }, diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 979dd50..ddbd2a1 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -139,7 +139,7 @@ static struct snd_soc_dai_link am3517evm_dai = { .cpu_dai_name ="omap-mcbsp-dai.0", .codec_dai_name = "tlv320aic23-hifi", .platform_name = "omap-pcm-audio", - .codec_name = "tlv320aic23-codec", + .codec_name = "tlv320aic23-codec.2-001a", .init = am3517evm_aic23_init, .ops = &am3517evm_ops, }; -- 1.6.2.4

2 1

[alsa-devel] period_time in slave definition
by Pavel Hofman 01 Mar '11

01 Mar '11

Hi, According to http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html , my .asoundrc contains: pcm.test { type plug slave { pcm "hw:0" format "unchanged" channels "unchanged" rate "unchanged" period_size 1024 buffer_size 4096 } } While pcm, format, channels, and rate are accepted, the period and buffer fields are refused: ALSA lib pcm.c:6811:(snd_pcm_slave_conf) Unknown field period_time I checked the source code and the keywords seem to be ok pcm.c: static const char *const names[SND_PCM_HW_PARAM_LAST_INTERVAL + 1] = { [SND_PCM_HW_PARAM_FORMAT] = "format", [SND_PCM_HW_PARAM_CHANNELS] = "channels", [SND_PCM_HW_PARAM_RATE] = "rate", [SND_PCM_HW_PARAM_PERIOD_TIME] = "period_time", [SND_PCM_HW_PARAM_PERIOD_SIZE] = "period_size", [SND_PCM_HW_PARAM_BUFFER_TIME] = "buffer_time", [SND_PCM_HW_PARAM_BUFFER_SIZE] = "buffer_size", [SND_PCM_HW_PARAM_PERIODS] = "periods" }; Has any change been introduced recently which is not reflected on wiki yet? Is there a way to specify "default" period and buffer parameters? Thanks a lot, Pavel.

2 7

[alsa-devel] [PATCH 21/21]sound:soc:samsung:dma.c remove one to many l's in the word.
by Justin P. Mattock 01 Mar '11

01 Mar '11

The patch below removes an extra "l" in the word. Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> --- sound/soc/samsung/dma.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 2124019..807efa5 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -313,7 +313,7 @@ dma_pointer(struct snd_pcm_substream *substream) /* we seem to be getting the odd error from the pcm library due * to out-of-bounds pointers. this is maybe due to the dma engine * not having loaded the new values for the channel before being - * callled... (todo - fix ) + * called... (todo - fix ) */ if (res >= snd_pcm_lib_buffer_bytes(substream)) { -- 1.7.4.1

5 6

[alsa-devel] [PATCH 1/4] ASoC: Add TI tlv320aic32x4 codec support.
by Javier Martin 01 Mar '11

01 Mar '11

This patch adds support for tlv320aic3205 and tlv320aic3254 codecs. It doesn't include miniDSP support for aic3254. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic32x4.c | 884 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.h | 138 ++++++ 4 files changed, 1028 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic32x4.c create mode 100644 sound/soc/codecs/tlv320aic32x4.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3b5690d..497c9a6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TVL320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -182,6 +183,9 @@ config SND_SOC_TLV320AIC26 tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE depends on SPI +config SND_SOC_TVL320AIC32X4 + tristate + config SND_SOC_TLV320AIC3X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f67a2d6..5709ca6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o @@ -98,6 +99,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c new file mode 100644 index 0000000..5ae7b20 --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -0,0 +1,884 @@ +/* + * linux/sound/soc/codecs/tlv320aic32x4.c + * + * Copyright 2011 Vista Silicon S.L. + * + * Author: Javier Martin <javier.martin(a)vista-silicon.com> + * + * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27. + * + * 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. + * + * 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 Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/cdev.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "tlv320aic32x4.h" + +#define SOC_DOUBLE_R_AIC32X4(xname, reg_left, reg_right, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_soc_info_volsw_2r_aic32x4, \ + .get = snd_soc_get_volsw_2r_aic32x4, \ + .put = snd_soc_put_volsw_2r_aic32x4, \ + .private_value = (reg_left) | ((shift) << 8) | \ + ((mask) << 12) | ((invert) << 20) | ((reg_right) << 24) } + +struct aic32x4_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; + u8 blck_N; +}; + +struct aic32x4_priv { + u32 sysclk; + s32 master; + u8 page_no; + void *control_data; +}; + +struct aic32x4_configs { + u8 reg_offset; + u8 reg_val; +}; + +#define aic32x4_reset(c) snd_soc_write(c, AIC32X4_RESET, 0x01); + +static int snd_soc_info_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); + +static int snd_soc_get_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + +static int snd_soc_put_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + +static const struct snd_kcontrol_new aic32x4_snd_controls[] = { + SOC_DOUBLE_R_AIC32X4("PCM Playback Volume", AIC32X4_LDACVOL, + AIC32X4_RDACVOL, 0, 0xAf, 0), + SOC_DOUBLE_R_AIC32X4("HP Driver Gain", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 0, 0x23, 0), + SOC_DOUBLE_R_AIC32X4("LO Driver Gain", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 0, 0x23, 0), + SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 6, 0x01, 1), + SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 6, 0x01, 1), + + SOC_DOUBLE_R("ADC Volume Level", AIC32X4_LADCVOL, + AIC32X4_RADCVOL, 0, 0x28, 0), + SOC_DOUBLE_R("PGA Gain Level", AIC32X4_LMICPGAVOL, + AIC32X4_RMICPGAVOL, 0, 0x5f, 0), + + SOC_SINGLE("Auto-mute", AIC32X4_DACMUTE, 4, 7, 0), + + SOC_SINGLE("AGC Left Enable Switch", AIC32X4_LAGC1, 7, 1, 0), + SOC_SINGLE("AGC Right Enable Switch", AIC32X4_RAGC1, 7, 1, 0), + SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1, + 4, 0x07, 0), + SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1, + 0, 0x03, 0), + SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2, + 6, 0x03, 0), + SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2, + 1, 0x1F, 0), + SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3, + 0, 0x7F, 0), + SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5, + 3, 0x1F, 0), + SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6, + 0, 0x1F, 0), + SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7, + 0, 0x0F, 0), + + SOC_SINGLE("MIC BIAS", AIC32X4_MICBIAS, 6, 1, 0), +}; + +static const struct aic32x4_rate_divs aic32x4_divs[] = { + /* 8k rate */ + {AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24}, + {AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24}, + {AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24}, + /* 11.025k rate */ + {AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16}, + {AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16}, + /* 16k rate */ + {AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12}, + {AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12}, + {AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12}, + /* 22.05k rate */ + {AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8}, + {AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8}, + {AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8}, + /* 32k rate */ + {AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6}, + {AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6}, + /* 44.1k rate */ + {AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4}, + /* 48k rate */ + {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4}, + {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4}, + {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4} +}; + +static const struct aic32x4_configs aic32x4_reg_init[] = { + {AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE}, + {AIC32X4_LDOCTL, AIC32X4_LDOCTLEN}, + {AIC32X4_CMMODE, AIC32X4_LDOIN_18_36 | AIC32X4_LDOIN2HP}, + {AIC32X4_CLKMUX, AIC32X4_PLLCLKIN}, + {AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK}, + {AIC32X4_DACSETUP, + AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN | AIC32X4_SSTEP2WCLK}, + {AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K}, + {AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K}, + {AIC32X4_LMICPGAVOL, 0x00}, + {AIC32X4_RMICPGAVOL, 0x00}, + /* Unmute ADC left and right channels */ + {AIC32X4_ADCFGA, 0x00}, + /* MICBIAS = 2.075V(CM=0.75V) generated from LDOIN */ + {AIC32X4_MICBIAS, 0x68}, + {AIC32X4_DACSPB, AIC32X4_DACSPBLOCK_MASK & 0x1}, + {AIC32X4_ADCSPB, AIC32X4_ADCSPBLOCK_MASK & 0x1}, +}; + +static const struct snd_kcontrol_new hpl_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC switch", AIC32X4_HPLROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_L switch", AIC32X4_HPLROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new hpr_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC switch", AIC32X4_HPRROUTE, 3, 1, 0), + SOC_DAPM_SINGLE("IN1_R switch", AIC32X4_HPRROUTE, 2, 1, 0), +}; + +static const struct snd_kcontrol_new lol_output_mixer_controls[] = { + SOC_DAPM_SINGLE("L_DAC switch", AIC32X4_LOLROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new lor_output_mixer_controls[] = { + SOC_DAPM_SINGLE("R_DAC switch", AIC32X4_LORROUTE, 3, 1, 0), +}; + +static const struct snd_kcontrol_new left_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_L P switch", AIC32X4_LMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_L P switch", AIC32X4_LMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_L P switch", AIC32X4_LMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_kcontrol_new right_input_mixer_controls[] = { + SOC_DAPM_SINGLE("IN1_R P switch", AIC32X4_RMICPGAPIN, 6, 1, 0), + SOC_DAPM_SINGLE("IN2_R P switch", AIC32X4_RMICPGAPIN, 4, 1, 0), + SOC_DAPM_SINGLE("IN3_R P switch", AIC32X4_RMICPGAPIN, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0), + SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0, + &hpl_output_mixer_controls[0], + ARRAY_SIZE(hpl_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0, + &lol_output_mixer_controls[0], + ARRAY_SIZE(lol_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0), + SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0, + &hpr_output_mixer_controls[0], + ARRAY_SIZE(hpr_output_mixer_controls)), + SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0, + &lor_output_mixer_controls[0], + ARRAY_SIZE(lor_output_mixer_controls)), + SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, + &left_input_mixer_controls[0], + ARRAY_SIZE(left_input_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, + &right_input_mixer_controls[0], + ARRAY_SIZE(right_input_mixer_controls)), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOL"), + SND_SOC_DAPM_OUTPUT("LOR"), + SND_SOC_DAPM_INPUT("IN1_L"), + SND_SOC_DAPM_INPUT("IN1_R"), + SND_SOC_DAPM_INPUT("IN2_L"), + SND_SOC_DAPM_INPUT("IN2_R"), + SND_SOC_DAPM_INPUT("IN3_L"), + SND_SOC_DAPM_INPUT("IN3_R"), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + /* Left Output */ + {"HPL Output Mixer", "L_DAC switch", "Left DAC"}, + {"HPL Output Mixer", "IN1_L switch", "IN1_L"}, + + {"HPL Power", NULL, "HPL Output Mixer"}, + {"HPL", NULL, "HPL Power"}, + + {"LOL Output Mixer", "L_DAC switch", "Left DAC"}, + + {"LOL Power", NULL, "LOL Output Mixer"}, + {"LOL", NULL, "LOL Power"}, + + /* Right Output */ + {"HPR Output Mixer", "R_DAC switch", "Right DAC"}, + {"HPR Output Mixer", "IN1_R switch", "IN1_R"}, + + {"HPR Power", NULL, "HPR Output Mixer"}, + {"HPR", NULL, "HPR Power"}, + + {"LOR Output Mixer", "R_DAC switch", "Right DAC"}, + + {"LOR Power", NULL, "LOR Output Mixer"}, + {"LOR", NULL, "LOR Power"}, + + /* Left input */ + {"Left Input Mixer", "IN1_L P switch", "IN1_L"}, + {"Left Input Mixer", "IN2_L P switch", "IN2_L"}, + {"Left Input Mixer", "IN3_L P switch", "IN3_L"}, + + {"Left ADC", NULL, "Left Input Mixer"}, + + /* Right Input */ + {"Right Input Mixer", "IN1_R P switch", "IN1_R"}, + {"Right Input Mixer", "IN2_R P switch", "IN2_R"}, + {"Right Input Mixer", "IN3_R P switch", "IN3_R"}, + + {"Right ADC", NULL, "Right Input Mixer"}, +}; + +static inline int aic32x4_change_page(struct snd_soc_codec *codec, + unsigned int new_page) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data[2]; + + data[0] = 0x00; + data[1] = new_page & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) { + aic32x4->page_no = new_page; + return 0; + } else { + return -EIO; + } +} + +static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + u8 data[2]; + int ret; + + /* A write to AIC32X4_PSEL is really a non-explicit page change */ + if (reg == AIC32X4_PSEL) + return aic32x4_change_page(codec, val); + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + + data[0] = fixed_reg & 0xff; + data[1] = val & 0xff; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + unsigned int page = reg / 128; + unsigned int fixed_reg = reg % 128; + int ret; + + if (aic32x4->page_no != page) { + ret = aic32x4_change_page(codec, page); + if (ret != 0) + return ret; + } + return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); +} + +static int snd_soc_info_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 12) & 0xff; + + uinfo->type = + mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + return 0; +} + +int snd_soc_get_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int reg2 = (kcontrol->private_value >> 24) & 0xff; + int mask; + int shift; + unsigned short val, val2; + + if (!strcmp(kcontrol->id.name, "PCM Playback Volume")) { + mask = 0xff; + shift = 0; + } else if ((!strcmp(kcontrol->id.name, "HP Driver Gain")) || + (!strcmp(kcontrol->id.name, "LO Driver Gain"))) { + mask = 0x3F; + shift = 0; + } else if (!strcmp(kcontrol->id.name, "PGA Capture Volume")) { + mask = 0x7F; + shift = 0; + } else { + printk(KERN_ERR "aic32x4: invalid kcontrol name\n"); + return -1; + } + + val = (snd_soc_read(codec, reg) >> shift) & mask; + val2 = (snd_soc_read(codec, reg2) >> shift) & mask; + + if (!strcmp(kcontrol->id.name, "PCM Playback Volume")) { + ucontrol->value.integer.value[0] = + (val <= 48) ? (val + 127) : (val - 129); + ucontrol->value.integer.value[1] = + (val2 <= 48) ? (val2 + 127) : (val2 - 129); + } else if ((!strcmp(kcontrol->id.name, "HP Driver Gain")) + || (!strcmp(kcontrol->id.name, "LO Driver Gain"))) { + ucontrol->value.integer.value[0] = + (val <= 29) ? (val + 6) : (val - 58); + ucontrol->value.integer.value[1] = + (val2 <= 29) ? (val2 + 6) : (val2 - 58); + } else if (!strcmp(kcontrol->id.name, "PGA Capture Volume")) { + ucontrol->value.integer.value[0] = + (val <= 38) ? (val + 25) : (val - 103); + ucontrol->value.integer.value[1] = + (val2 <= 38) ? (val2 + 25) : (val2 - 103); + } + return 0; +} + +int snd_soc_put_volsw_2r_aic32x4(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int reg2 = (kcontrol->private_value >> 24) & 0xff; + int err; + unsigned short val, val2, val_mask; + + val = ucontrol->value.integer.value[0]; + val2 = ucontrol->value.integer.value[1]; + + if (!strcmp(kcontrol->id.name, "PCM Playback Volume")) { + val = (val >= 127) ? (val - 127) : (val + 129); + val2 = (val2 >= 127) ? (val2 - 127) : (val2 + 129); + val_mask = 0xff; /* 8 bits */ + } else if ((!strcmp(kcontrol->id.name, "HP Driver Gain")) || + (!strcmp(kcontrol->id.name, "LO Driver Gain"))) { + val = (val >= 6) ? (val - 6) : (val + 58); + val2 = (val2 >= 6) ? (val2 - 6) : (val2 + 58); + val_mask = 0x3F; /* 6 bits */ + } else if (!strcmp(kcontrol->id.name, "PGA Capture Volume")) { + val = (val >= 25) ? (val - 25) : (val + 103); + val2 = (val2 >= 25) ? (val2 - 25) : (val2 + 103); + val_mask = 0x7F; /* 7 bits */ + } else { + printk(KERN_ERR "aic32x4: invalid control name\n"); + return -1; + } + + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) { + printk(KERN_ERR "aic32x4: error while updating bits\n"); + return err; + } + + err = snd_soc_update_bits(codec, reg2, val_mask, val2); + return err; +} + +static inline int aic32x4_get_divs(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) { + if ((aic32x4_divs[i].rate == rate) + && (aic32x4_divs[i].mclk == mclk)) { + return i; + } + } + printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n"); + return -EINVAL; +} + +static int aic32x4_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + ARRAY_SIZE(aic32x4_dapm_routes)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case AIC32X4_FREQ_12000000: + case AIC32X4_FREQ_24000000: + case AIC32X4_FREQ_25000000: + aic32x4->sysclk = freq; + return 0; + } + printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n"); + return -EINVAL; +} + +static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 iface_reg_1; + u8 iface_reg_2; + u8 iface_reg_3; + + iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1); + iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2); + iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2); + iface_reg_2 = 0; + iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3); + iface_reg_3 = iface_reg_3 & ~(1 << 3); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aic32x4->master = 1; + iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + aic32x4->master = 0; + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + iface_reg_2 = 0x01; /* add offset 1 */ + break; + case SND_SOC_DAIFMT_DSP_B: + iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT); + iface_reg_3 |= (1 << 3); /* invert bit clock */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg_1 |= + (AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg_1 |= + (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT); + break; + default: + printk(KERN_ERR "aic32x4: invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1); + snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2); + snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3); + return 0; +} + +static int aic32x4_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 data; + int i; + + i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params)); + if (i < 0) { + printk(KERN_ERR "aic32x4: sampling rate not supported\n"); + return i; + } + + /* We will fix R value to 1 and will make P & J=K.D as varialble */ + data = snd_soc_read(codec, AIC32X4_PLLPR); + data &= ~(7 << 4); + snd_soc_write(codec, AIC32X4_PLLPR, + (data | (aic32x4_divs[i].p_val << 4) | 0x01)); + + snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j); + + snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8)); + snd_soc_write(codec, AIC32X4_PLLDLSB, + (aic32x4_divs[i].pll_d & 0xff)); + + /* NDAC divider value */ + data = snd_soc_read(codec, AIC32X4_NDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac); + + /* MDAC divider value */ + data = snd_soc_read(codec, AIC32X4_MDAC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac); + + /* DOSR MSB & LSB values */ + snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8); + snd_soc_write(codec, AIC32X4_DOSRLSB, + (aic32x4_divs[i].dosr & 0xff)); + + /* NADC divider value */ + data = snd_soc_read(codec, AIC32X4_NADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc); + + /* MADC divider value */ + data = snd_soc_read(codec, AIC32X4_MADC); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc); + + /* AOSR value */ + snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr); + + /* BCLK N divider */ + data = snd_soc_read(codec, AIC32X4_BCLKN); + data &= ~(0x7f); + snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N); + + data = snd_soc_read(codec, AIC32X4_IFACE1); + data = data & ~(3 << 4); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S24_LE: + data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT); + break; + case SNDRV_PCM_FORMAT_S32_LE: + data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT); + break; + } + snd_soc_write(codec, AIC32X4_IFACE1, data); + + return 0; +} + +static int aic32x4_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 dac_reg; + + dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON; + if (mute) + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON); + else + snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg); + return 0; +} + +static int aic32x4_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + u8 value; + + switch (level) { + case SND_SOC_BIAS_ON: + if (aic32x4->master) { + /* Switch on PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value | AIC32X4_PLLEN)); + + /* Switch on NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value | AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value | AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value | AIC32X4_MDACEN); + + /* Switch on MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value | AIC32X4_MDACEN); + + /* Switch on BCLK_N Divider */ + value = snd_soc_read(codec, AIC32X4_BCLKN); + snd_soc_write(codec, AIC32X4_BCLKN, + value | AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (aic32x4->master) { + /* Switch off PLL */ + value = snd_soc_read(codec, AIC32X4_PLLPR); + snd_soc_write(codec, AIC32X4_PLLPR, + (value & ~AIC32X4_PLLEN)); + + /* Switch off NDAC Divider */ + value = snd_soc_read(codec, AIC32X4_NDAC); + snd_soc_write(codec, AIC32X4_NDAC, + value & ~AIC32X4_NDACEN); + + /* Switch off MDAC Divider */ + value = snd_soc_read(codec, AIC32X4_MDAC); + snd_soc_write(codec, AIC32X4_MDAC, + value & ~AIC32X4_MDACEN); + + /* Switch off NADC Divider */ + value = snd_soc_read(codec, AIC32X4_NADC); + snd_soc_write(codec, AIC32X4_NADC, + value & ~AIC32X4_NDACEN); + + /* Switch off MADC Divider */ + value = snd_soc_read(codec, AIC32X4_MADC); + snd_soc_write(codec, AIC32X4_MADC, + value & ~AIC32X4_MDACEN); + value = snd_soc_read(codec, AIC32X4_BCLKN); + + /* Switch off BCLK_N Divider */ + snd_soc_write(codec, AIC32X4_BCLKN, + value & ~AIC32X4_BCLKEN); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + codec->bias_level = level; + return 0; +} + +#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000 +#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops aic32x4_ops = { + .hw_params = aic32x4_hw_params, + .digital_mute = aic32x4_mute, + .set_fmt = aic32x4_set_dai_fmt, + .set_sysclk = aic32x4_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver aic32x4_dai = { + .name = "tlv320aic32x4-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC32X4_RATES, + .formats = AIC32X4_FORMATS,}, + .ops = &aic32x4_ops, + .symmetric_rates = 1, +}; + +static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic32x4_resume(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic32x4_probe(struct snd_soc_codec *codec) +{ + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + int i; + + codec->hw_write = (hw_write_t) i2c_master_send; + codec->control_data = aic32x4->control_data; + + aic32x4_reset(codec); + + /* Initial value for some registers */ + for (i = 0; i < ARRAY_SIZE(aic32x4_reg_init); i++) + snd_soc_write(codec, aic32x4_reg_init[i].reg_offset, + aic32x4_reg_init[i].reg_val); + + aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_add_controls(codec, aic32x4_snd_controls, + ARRAY_SIZE(aic32x4_snd_controls)); + aic32x4_add_widgets(codec); + + return 0; +} + + +static int aic32x4_remove(struct snd_soc_codec *codec) +{ + aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { + .read = aic32x4_read, + .write = aic32x4_write, + .probe = aic32x4_probe, + .remove = aic32x4_remove, + .suspend = aic32x4_suspend, + .resume = aic32x4_resume, + .set_bias_level = aic32x4_set_bias_level, +}; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic32x4_priv *aic32x4; + int ret; + + aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL); + if (aic32x4 == NULL) + return -ENOMEM; + + aic32x4->control_data = i2c; + i2c_set_clientdata(i2c, aic32x4); + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_aic32x4, &aic32x4_dai, 1); + if (ret < 0) + kfree(aic32x4); + return ret; +} + +static __devexit int aic32x4_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id aic32x4_i2c_id[] = { + { "tlv320aic32x4", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); + +static struct i2c_driver aic32x4_i2c_driver = { + .driver = { + .name = "tlv320aic32x4-codec", + .owner = THIS_MODULE, + }, + .probe = aic32x4_i2c_probe, + .remove = __devexit_p(aic32x4_i2c_remove), + .id_table = aic32x4_i2c_id, +}; +#endif + +static int __init aic32x4_modinit(void) +{ + int ret = 0; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&aic32x4_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(aic32x4_modinit); + +static void __exit aic32x4_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&aic32x4_i2c_driver); +#endif +} +module_exit(aic32x4_exit); + +MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver"); +MODULE_AUTHOR("Javier Martin <javier.martin(a)vista-silicon.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h new file mode 100644 index 0000000..48c0a3d --- /dev/null +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -0,0 +1,138 @@ +/* + * tlv320aic32x4.h + * + * 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. + */ + + +#ifndef _TLV320AIC32X4_H +#define _TLV320AIC32X4_H + +/* tlv320aic32x4 register space (in decimal to match datasheet) */ + +#define AIC32X4_PAGE1 128 + +#define AIC32X4_PSEL 0 +#define AIC32X4_RESET 1 +#define AIC32X4_CLKMUX 4 +#define AIC32X4_PLLPR 5 +#define AIC32X4_PLLJ 6 +#define AIC32X4_PLLDMSB 7 +#define AIC32X4_PLLDLSB 8 +#define AIC32X4_NDAC 11 +#define AIC32X4_MDAC 12 +#define AIC32X4_DOSRMSB 13 +#define AIC32X4_DOSRLSB 14 +#define AIC32X4_NADC 18 +#define AIC32X4_MADC 19 +#define AIC32X4_AOSR 20 +#define AIC32X4_CLKMUX2 25 +#define AIC32X4_CLKOUTM 26 +#define AIC32X4_IFACE1 27 +#define AIC32X4_IFACE2 28 +#define AIC32X4_IFACE3 29 +#define AIC32X4_BCLKN 30 +#define AIC32X4_IFACE4 31 +#define AIC32X4_IFACE5 32 +#define AIC32X4_IFACE6 33 +#define AIC32X4_DOUTCTL 53 +#define AIC32X4_DINCTL 54 +#define AIC32X4_DACSPB 60 +#define AIC32X4_ADCSPB 61 +#define AIC32X4_DACSETUP 63 +#define AIC32X4_DACMUTE 64 +#define AIC32X4_LDACVOL 65 +#define AIC32X4_RDACVOL 66 +#define AIC32X4_ADCSETUP 81 +#define AIC32X4_ADCFGA 82 +#define AIC32X4_LADCVOL 83 +#define AIC32X4_RADCVOL 84 +#define AIC32X4_LAGC1 86 +#define AIC32X4_LAGC2 87 +#define AIC32X4_LAGC3 88 +#define AIC32X4_LAGC4 89 +#define AIC32X4_LAGC5 90 +#define AIC32X4_LAGC6 91 +#define AIC32X4_LAGC7 92 +#define AIC32X4_RAGC1 94 +#define AIC32X4_RAGC2 95 +#define AIC32X4_RAGC3 96 +#define AIC32X4_RAGC4 97 +#define AIC32X4_RAGC5 98 +#define AIC32X4_RAGC6 99 +#define AIC32X4_RAGC7 100 +#define AIC32X4_PWRCFG (AIC32X4_PAGE1 + 1) +#define AIC32X4_LDOCTL (AIC32X4_PAGE1 + 2) +#define AIC32X4_OUTPWRCTL (AIC32X4_PAGE1 + 9) +#define AIC32X4_CMMODE (AIC32X4_PAGE1 + 10) +#define AIC32X4_HPLROUTE (AIC32X4_PAGE1 + 12) +#define AIC32X4_HPRROUTE (AIC32X4_PAGE1 + 13) +#define AIC32X4_LOLROUTE (AIC32X4_PAGE1 + 14) +#define AIC32X4_LORROUTE (AIC32X4_PAGE1 + 15) +#define AIC32X4_HPLGAIN (AIC32X4_PAGE1 + 16) +#define AIC32X4_HPRGAIN (AIC32X4_PAGE1 + 17) +#define AIC32X4_LOLGAIN (AIC32X4_PAGE1 + 18) +#define AIC32X4_LORGAIN (AIC32X4_PAGE1 + 19) +#define AIC32X4_HEADSTART (AIC32X4_PAGE1 + 20) +#define AIC32X4_MICBIAS (AIC32X4_PAGE1 + 51) +#define AIC32X4_LMICPGAPIN (AIC32X4_PAGE1 + 52) +#define AIC32X4_LMICPGANIN (AIC32X4_PAGE1 + 54) +#define AIC32X4_RMICPGAPIN (AIC32X4_PAGE1 + 55) +#define AIC32X4_RMICPGANIN (AIC32X4_PAGE1 + 57) +#define AIC32X4_FLOATINGINPUT (AIC32X4_PAGE1 + 58) +#define AIC32X4_LMICPGAVOL (AIC32X4_PAGE1 + 59) +#define AIC32X4_RMICPGAVOL (AIC32X4_PAGE1 + 60) + +#define AIC32X4_FREQ_12000000 12000000 +#define AIC32X4_FREQ_24000000 24000000 +#define AIC32X4_FREQ_25000000 25000000 + +#define AIC32X4_WORD_LEN_16BITS 0x00 +#define AIC32X4_WORD_LEN_20BITS 0x01 +#define AIC32X4_WORD_LEN_24BITS 0x02 +#define AIC32X4_WORD_LEN_32BITS 0x03 + +#define AIC32X4_I2S_MODE 0x00 +#define AIC32X4_DSP_MODE 0x01 +#define AIC32X4_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC32X4_LEFT_JUSTIFIED_MODE 0x03 + +#define AIC32X4_AVDDWEAKDISABLE 0x08 +#define AIC32X4_LDOCTLEN 0x01 + +#define AIC32X4_LDOIN_18_36 0x01 +#define AIC32X4_LDOIN2HP 0x02 + +#define AIC32X4_DACSPBLOCK_MASK 0x1f +#define AIC32X4_ADCSPBLOCK_MASK 0x1f + +#define AIC32X4_PLLJ_SHIFT 6 +#define AIC32X4_DOSRMSB_SHIFT 4 + +#define AIC32X4_PLLCLKIN 0x03 + +#define AIC32X4_LMICPGANIN_IN2R_10K 0x10 +#define AIC32X4_RMICPGANIN_IN1L_10K 0x10 + +#define AIC32X4_LMICPGAVOL_NOGAIN 0x80 +#define AIC32X4_RMICPGAVOL_NOGAIN 0x80 + +#define AIC32X4_BCLKMASTER 0x08 +#define AIC32X4_WCLKMASTER 0x04 +#define AIC32X4_PLLEN (0x01 << 7) +#define AIC32X4_NDACEN (0x01 << 7) +#define AIC32X4_MDACEN (0x01 << 7) +#define AIC32X4_NADCEN (0x01 << 7) +#define AIC32X4_MADCEN (0x01 << 7) +#define AIC32X4_BCLKEN (0x01 << 7) +#define AIC32X4_DACEN (0x03 << 6) +#define AIC32X4_LDAC2LCHN (0x01 << 4) +#define AIC32X4_RDAC2RCHN (0x01 << 2) + +#define AIC32X4_SSTEP2WCLK 0x01 +#define AIC32X4_MUTEON 0x0C +#define AIC32X4_DACMOD2BCLK 0x01 + +#endif /* _TLV320AIC32X4_H */ -- 1.7.0.4

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[alsa-devel] [PATCH v22 0/3] ASoC/MFD/V4L2: WL1273 FM Radio Driver
by Matti J. Aaltonen 01 Mar '11

01 Mar '11

Hello. Thanks for the comment Mark. On Tue, 2011-03-01 at 11:54 +0000, ext Mark Brown wrote: On Tue, Mar 01, 2011 at 10:00:50AM +0200, Matti J. Aaltonen wrote: > > These changes are needed to keep up with the changes in the > > MFD core and V4L2 parts of the wl1273 FM radio driver. > > > > Use function pointers instead of exported functions for I2C IO. > > Also move all preprocessor constants from the wl1273.h to > > include/linux/mfd/wl1273-core.h. > > > > Also update the year in the copyright statement. > > It's not actually doing that: > > > - * Copyright: (C) 2010 Nokia Corporation > > + * Copyright: (C) 2011 Nokia Corporation > > It's replacing it - portions are still 2010. Kept also the year 2010 on the copyright line. > Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> On Tue, 2011-03-01 at 12:43 +0100, ext Samuel Ortiz wrote: > Acked-by: Samuel Ortiz <sameo(a)linux.intel.com> Cheers, Matti Matti J. Aaltonen (3): MFD: WL1273 FM Radio: MFD driver for the FM radio. V4L2: WL1273 FM Radio: TI WL1273 FM radio driver ASoC: WL1273 FM radio: Access I2C IO functions through pointers. drivers/media/radio/radio-wl1273.c | 360 +++++++++++------------------------- drivers/mfd/Kconfig | 2 +- drivers/mfd/wl1273-core.c | 149 +++++++++++++++- include/linux/mfd/wl1273-core.h | 2 + sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 11 +- 6 files changed, 264 insertions(+), 262 deletions(-)

3 6

[alsa-devel] snd_pcm_prepare FAILS
by Irfan Shaikh 01 Mar '11

01 Mar '11

Hello, I have been using following section of code, my snd_pcm_prepare() always fails and i am not able to play audio using snd_pcm_writei(). /******************************Code*********************************/ err=snd_pcm_prepare(handle); if (err < 0) RSE_TRACE_ERROR(MP_ALSA_TRACE,("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err))); /**********************************************************************/ $$$$$$$$$$$$ Output : Can't recovery from underrun, prepare failed: Device or resource busy $$$$$$$$$$$$$ Can some one please share there comments on the above. What is going run above. Regards, SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 1

[alsa-devel] Underrun
by Irfan Shaikh 01 Mar '11

01 Mar '11

Hello All, A) I am getting many under-run in my audio capture and playback application. 1) My CAPTURE buffer is of 1024 * 8 (Period size * no of periods), PLAYBACK buffer is of (512 * 16) 2) I try to read 8 K frames each time and write returned frames to playback device. 3) Stereo audio, sample size 2 bytes, frame size 4 bytes, 48000 Hz I have done following to reduce under run : 1) Increased the number of periods on playback device (to increase playback latency). 2) optimized my code Please sugest me some more way so that i can avoid underrun of my playback device B) Can i use snd_pcm_prepare() twice in two consecutive lines For eg : { snd_pcm_prepare(pcm); snd_pcm_prepare(pcm); } I just want to know if this is OK for playback device state. Regards SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html

2 1

[alsa-devel] [PATCH 0/5] Even more hdspmixer fixes
by Adrian Knoth 01 Mar '11

01 Mar '11

Hi! Next round of hdspmixer updates. Mostly cosmetics, but also fixing a segfault with H9652 cards. Thanks to Craig Bourne for immediate testing, also tested and confirmed to be working on RayDAT, Multiface and AES32. Adrian Knoth (5): hdspmixer: Fix compiler warnings. hdspmixer: Don't use hardcoded number of cards. hdspmixer: Make static destination maps extern hdspmixer: Lower nesting depth in label assignment hdspmixer: Fix labels on H9652 hdspmixer/pixmaps/Slider1.xpm | 2 +- hdspmixer/pixmaps/Slider2.xpm | 2 +- hdspmixer/pixmaps/Zlmloop2.xpm | 2 +- hdspmixer/pixmaps/Zlmloopm2.xpm | 2 +- hdspmixer/pixmaps/b_blank.xpm | 2 +- hdspmixer/pixmaps/b_card1.xpm | 2 +- hdspmixer/pixmaps/b_card2.xpm | 2 +- hdspmixer/pixmaps/b_card3.xpm | 2 +- hdspmixer/pixmaps/b_mute.xpm | 2 +- hdspmixer/pixmaps/b_pre1.xpm | 2 +- hdspmixer/pixmaps/b_pre2.xpm | 2 +- hdspmixer/pixmaps/b_pre3.xpm | 2 +- hdspmixer/pixmaps/b_pre4.xpm | 2 +- hdspmixer/pixmaps/b_pre5.xpm | 2 +- hdspmixer/pixmaps/b_pre6.xpm | 2 +- hdspmixer/pixmaps/b_pre7.xpm | 2 +- hdspmixer/pixmaps/b_pre8.xpm | 2 +- hdspmixer/pixmaps/b_save.xpm | 2 +- hdspmixer/pixmaps/b_solo.xpm | 2 +- hdspmixer/pixmaps/b_zlm.xpm | 2 +- hdspmixer/pixmaps/buttons.xpm | 2 +- hdspmixer/pixmaps/empty.xpm | 2 +- hdspmixer/pixmaps/empty_2.xpm | 2 +- hdspmixer/pixmaps/empty_linux.xpm | 2 +- hdspmixer/pixmaps/iomixer.xpm | 2 +- hdspmixer/pixmaps/iomixer_r.xpm | 2 +- hdspmixer/pixmaps/level.xpm | 2 +- hdspmixer/pixmaps/matrix_black.xpm | 2 +- hdspmixer/pixmaps/matrix_grey.xpm | 2 +- hdspmixer/pixmaps/matrix_grey_l.xpm | 2 +- hdspmixer/pixmaps/matrix_grey_o.xpm | 2 +- hdspmixer/pixmaps/matrix_grey_ol.xpm | 2 +- hdspmixer/pixmaps/matrix_mute.xpm | 2 +- hdspmixer/pixmaps/matrix_white.xpm | 2 +- hdspmixer/pixmaps/matrix_yellow.xpm | 2 +- hdspmixer/pixmaps/monitor.xpm | 2 +- hdspmixer/pixmaps/mute.xpm | 2 +- hdspmixer/pixmaps/output.xpm | 2 +- hdspmixer/pixmaps/output_r.xpm | 2 +- hdspmixer/pixmaps/over.xpm | 2 +- hdspmixer/pixmaps/peak.xpm | 2 +- hdspmixer/pixmaps/solo.xpm | 2 +- hdspmixer/src/HDSPMixerAbout.cxx | 2 +- hdspmixer/src/HDSPMixerAbout.h | 2 +- hdspmixer/src/HDSPMixerAboutText.h | 2 +- hdspmixer/src/HDSPMixerGain.cxx | 2 +- hdspmixer/src/HDSPMixerGain.h | 2 +- hdspmixer/src/HDSPMixerIOMixer.h | 4 +- hdspmixer/src/HDSPMixerOutput.cxx | 189 ++++++++++-------- hdspmixer/src/HDSPMixerOutput.h | 4 +- hdspmixer/src/HDSPMixerSelector.cxx | 38 ++-- hdspmixer/src/HDSPMixerSelector.h | 2 +- hdspmixer/src/HDSPMixerSetup.cxx | 2 +- hdspmixer/src/HDSPMixerSetup.h | 2 +- hdspmixer/src/Makefile.am | 1 + hdspmixer/src/channelmap.cxx | 368 ++++++++++++++++++++++++++++++++++ hdspmixer/src/channelmap.h | 348 ++++++--------------------------- hdspmixer/src/pixmaps.h | 76 ++++---- 58 files changed, 642 insertions(+), 486 deletions(-) create mode 100644 hdspmixer/src/channelmap.cxx -- 1.7.4.1

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[alsa-devel] [PATCH v2] ASoC: WM8994: Ensure late enable events are processed for the ADCs
by Dimitris Papastamos 01 Mar '11

01 Mar '11

Ensure that the ADCs are provided with a clock as the previous patch "ASoC: WM8994: Improve playback robustness" did not handle this case properly. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 26 +++++++++++++++++++++++--- 1 files changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 4dfbaee..b17ac19 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1095,6 +1095,13 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int adc_mux_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + late_enable_ev(w, kcontrol, event); + return 0; +} + static int dac_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1415,6 +1422,18 @@ SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0), SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0), }; +static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = { +SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux, + adc_mux_ev, SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = { +SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), +SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), +}; + static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = { SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), @@ -1509,9 +1528,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux), -SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux), - SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), @@ -3282,11 +3298,15 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) if (wm8994->revision < 4) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, + ARRAY_SIZE(wm8994_adc_revd_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets, ARRAY_SIZE(wm8994_dac_revd_widgets)); } else { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets, ARRAY_SIZE(wm8994_lateclk_widgets)); + snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets, + ARRAY_SIZE(wm8994_adc_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets, ARRAY_SIZE(wm8994_dac_widgets)); } -- 1.7.4.1

1 0

[alsa-devel] [TLV320AIC23] [AM3517EVM][AUDIO] codec name mis-match in soc_bind_dai_link
by Koyamangalath, Abhilash 01 Mar '11

01 Mar '11

While trying to register the sound card on the AM3517EVM (which uses a tlv320aic23 codec), with the 2.6.37 kernel I get the following error message: CODEC tlv320aic23-codec not registered This is despite having gotten previously: Registered codec 'tlv320aic23-codec.2-001a' I realized that there was a names' mismatch in soc_bind_dai_link(): In fn soc_bind_dai_link:1249:codec->name is tlv320aic23-codec.2-001a, dai_link->codec_name is tlv320aic23-codec The 2-001a is the <i2c_bus>-<i2c_address> pair as defined in arch/arm/mach-omap2/board-am3517evm.c: static struct i2c_board_info __initdata am3517evm_i2c2_boardinfo[] = { { I2C_BOARD_INFO("tlv320aic23", 0x1A), }, This is the device name of the i2c client set automatically by i2c_new_device() when __process_new_driver() on the device is called, in response to a i2c_register_driver() request. Now, all devices with names in <i2c_bus>-<i2c_address> are assumed to be i2c clients by snd_soc_register_codec() and so <driver_name>.<dev_name> is the name supplied to the codec. Now, if I change the dai-link->codec_name to force-match by appending 2-001a to the codec_name in (sound/soc/omap/am3517evm.c): static struct snd_soc_dai_link am3517evm_dai = { : .codec_name = "tlv320aic23-codec.2-001a ", : }; the registration proceeds smoothly and I can see device nodes getting created. Since now there is no way to provide a customized i2c device name to a client device (or so I believe), what could be the best solution to this problem ? -Abhilash PS: OMAP3 evm does not have this problem as the audio codec twl4030 is on platform bus, I believe. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

Re: [alsa-devel] hdspmixer segfault (1.0.24+)
by Adrian Knoth 01 Mar '11

01 Mar '11

On Mon, Feb 28, 2011 at 04:35:36PM -0500, Craig Bourne wrote: Hi! > S/PDIF], this is all very reassuring. I will carry on with the testing on > Wednesday and report back my results. For all who care, foremost probably Fernando: $ git clone http://cluster.inf-ra.uni-jena.de/~adi/alsa-tools.git $ git checkout adi-fixes (or simply pull this branch into your local alsa-tools git repo). I've reworked the code in question. At least the label thing should be fine now, but let's wait for Craig to confirm it. Unfortunately, I won't be around on Wednesday. Maybe Fernando will prepare another test package, if not, compiling from source isn't that hard: After these two git commands, go into hdspmixer an run ./gitcompile. Then, if everythings builds correctly, run ./src/hdspmixer so see if it's working. Note that the git repo above might disappear once we have fixed all remaining issues, so if you read this mail in a couple of months from now, don't expect the URL to be still functional. ;) Cheers -- mail: adi(a)thur.de http://adi.thur.de PGP/GPG: key via keyserver

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Re: [alsa-devel] hdspmixer segfault (1.0.24+)
by Adrian Knoth 28 Feb '11

28 Feb '11

On 02/28/11 18:23, Craig Bourne wrote: > PCI Busmaster Digital I/O System > 2 + 24 Channels SPDIF / ADAT Interface I've digged into the driver source. Your H9652 is more or less a Digiface. The channel *count* was never a question, this is defined by the driver anyway. The remaining question simply was how to label the output destinations. _From the manual, the answer is pretty obvious: ADAT-1.1 to ADAT-1.8, followed by ADAT-2.1 to ADAT-2.8 and ADAT 3.1-to 3.8, last is S/PDIF. In double-speed mode (everything above 48kHz), the ADAT channels are cut in half, so ADAT 1.1 to ADAT 1.4, ADAT 2.1 to 2.4 and so on. S/PDIF remains unchanged as it supports higher sample rates at static channel counts (like AES/EBU). > At various times in the past when one was required to specify (e.g., > Input/Output) channels to an ALSA tool (or a program that relied on > the ALSA sound architecture) a mismatch in this number (presumably of > pcm channels) had less than desirable consequences. We're not talking driver here, the critical parts remain unchanged in the kernel. > All of this leads me to the conjecture that fudging an arbitrary > number that stands in for "channels" is perhaps not the best course > (not to forget that, at the end of our testing, I would hope that this > sound card that costs as much as the computer that houses it will not > have suffered any damage). Of course, this is open source. It comes without any warranty and you're free to avoid it, but to the best of my knowledge, you must not be afraid. Firstly, it's userspace, so nothing can really kill your card. Second, the whole logic is an FPGA, so it can always be flashed to a working state if something goes wrong (it cannot go wrong, as you cannot accidentally flash your card). Last but not least: the channel count has always been correct, 26 all the time, only the name array was missing. So we have this "read 26 channel names from the naming array", and it's now a question which names to write in there. I've given the answer above, but want to stress again that we're talking pure C pure userland here, so it's close to a hello world program: printf("Channel name is %s\n", name_array[channel_number]); with name_array being something like { "ADAT-1.1", "ADAT-1.2", .. } and so on. Right now, with the package prepared by Fernando, you'll simply see "1", "2", .. "26" instead of "ADAT-1.1", but this will change in a minute. HTH

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Re: [alsa-devel] hdspmixer segfault (1.0.24+)
by Fernando Lopez-Lezcano 28 Feb '11

28 Feb '11

On 02/28/2011 09:23 AM, Craig Bourne wrote: > I expect that I may have time Wednesday of this week to help with the > testing, etc. (I released a new package of alsa-tools in the planetccrma-testing repository so you can test - you have to install planetccrma-repo-testing and then yum --enablerepo planetccrma-testing install alsa-tools to get it). -- Fernando > > Though low level technical details are mostly absent from the > documentation that RME provided, I will try to locate and forward a > copy of their technical manual just in the event that that might be of > some help. [Ah! Here it is! > <http://media.zzounds.com/media/hdsp9652_e-2c1bba79532db8cd1c9b6f0beb31e855.…>] > > Meanwhile, please have a look at /RME9652.conf/ which reads, in part > (starting from about line 38): > > ... > > slave.pcm { > type route > slave { > pcm { > type hw > card $CARD > } > channels 26 > } > ttable.0.24 1 > ttable.1.25 1 > } > ... > > This number also reflects RME's characterization of the card (from the > title page of their manual) > > PCI Busmaster Digital I/O System > 2 + 24 Channels SPDIF / ADAT Interface > 24 Bit / 96 kHz Digital Audio > ADAT Sync In > MIDI I/O > TotalMix™ > 24 Bit / 96 kHz ü > SyncAlign® ZLM® SyncCheck® > ...

1 0

[alsa-devel] Realtek ALC269VB Asus 1001px intern mic ist not working
by Marty 28 Feb '11

28 Feb '11

Hi, hope I get some help with this problem. At the moment I work with Pulse Audio but the results are very bad. check: http://www.alsa-project.org/db/?f=0286416d03a64a2e34a4f73831c4f0f5cba002d6 best marty

2 1

[alsa-devel] [PATCH 1/2] ASoC: WM8994: Don't disable the AIF[1|2]CLK_ENA unconditionaly
by Dimitris Papastamos 28 Feb '11

28 Feb '11

Since we began using the late clock disable functionality, ensure that we don't disable the clock if any of the ADC or DAC paths are still enabled. This happens when we have simultaneous playback and recording. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++------ 1 files changed, 19 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0dc1411..4dfbaee 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -109,6 +109,9 @@ struct wm8994_priv { unsigned int aif1clk_enable:1; unsigned int aif2clk_enable:1; + + unsigned int aif1clk_disable:1; + unsigned int aif2clk_disable:1; }; static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) @@ -1014,14 +1017,18 @@ static int late_enable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (wm8994->aif1clk_enable) + if (wm8994->aif1clk_enable) { snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, WM8994_AIF1CLK_ENA); - if (wm8994->aif2clk_enable) + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, WM8994_AIF2CLK_ENA); + wm8994->aif2clk_enable = 0; + } break; } @@ -1036,15 +1043,15 @@ static int late_disable_ev(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMD: - if (wm8994->aif1clk_enable) { + if (wm8994->aif1clk_disable) { snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, WM8994_AIF1CLK_ENA_MASK, 0); - wm8994->aif1clk_enable = 0; + wm8994->aif1clk_disable = 0; } - if (wm8994->aif2clk_enable) { + if (wm8994->aif2clk_disable) { snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, WM8994_AIF2CLK_ENA_MASK, 0); - wm8994->aif2clk_enable = 0; + wm8994->aif2clk_disable = 0; } break; } @@ -1062,6 +1069,9 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: wm8994->aif1clk_enable = 1; break; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif1clk_disable = 1; + break; } return 0; @@ -1077,6 +1087,9 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMU: wm8994->aif2clk_enable = 1; break; + case SND_SOC_DAPM_POST_PMD: + wm8994->aif2clk_disable = 1; + break; } return 0; -- 1.7.4.1

1 1

[alsa-devel] hdspmixer segfault (1.0.24+)
by Fernando Lopez-Lezcano 28 Feb '11

28 Feb '11

Hi, I'm getting a segfault from hdspmixer using the current git alsa-tools (1.0.23 works, 1.0.24+ does not). See below for a backtrace. I'm running 2.6.33.x rt30, could this be the problem? (a dependency on newer kernels?) -- Fernando $ gdb ./hdspmixer/src/hdspmixer GNU gdb (GDB) Fedora (7.2-41.fc14) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i686-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /scratch/nando/alsa-tools/alsa-tools.git/hdspmixer/src/hdspmixer...done. (gdb) run Starting program: /scratch/nando/alsa-tools/alsa-tools.git/hdspmixer/src/hdspmixer [Thread debugging using libthread_db enabled] HDSPMixer 1.6 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org> This program comes with ABSOLUTELY NO WARRANTY HDSPMixer is free software, see the file COPYING for details Looking for RME cards: Card 0: RME Hammerfall DSP + Multiface at 0xe9200000, irq 19 Multiface found! Card 1: HDA Intel at 0xe9300000 irq 32 1 RME cards card found. Initializing default presets Program received signal SIGSEGV, Segmentation fault. HDSPMixerPresets::restore_preset (this=0x81fe500, prst=1) at HDSPMixerPresets.cxx:176 176 basew->inputs->strips[i]->pan->pos[z] = basew->inputs->strips[i]->data[card][speed][p]->pan_pos[z]; Missing separate debuginfos, use: debuginfo-install alsa-lib-1.0.24-1.fc14.i686 expat-2.0.1-10.fc13.i686 fltk-1.1.10-2.fc14.i686 fontconfig-2.8.0-2.fc14.i686 freetype-2.4.2-4.fc14.i686 glibc-2.13-1.i686 libX11-1.3.4-3.fc14.i686 libXau-1.0.6-1.fc14.i686 libXext-1.1.2-2.fc14.i686 libXft-2.1.14-1.fc13.i686 libXinerama-1.1-2.fc13.i686 libXrender-0.9.6-1.fc14.i686 libgcc-4.5.1-4.fc14.i686 libstdc++-4.5.1-4.fc14.i686 libxcb-1.7-1.fc14.i686 (gdb) bt #0 HDSPMixerPresets::restore_preset (this=0x81fe500, prst=1) at HDSPMixerPresets.cxx:176 #1 0x080552de in HDSPMixerPresets::preset_change (this=0x81fe500, p=1) at HDSPMixerPresets.cxx:258 #2 0x0804b436 in readregisters_cb (arg=0x80a0710) at HDSPMixerWindow.cxx:67 #3 0x4dd311d3 in Fl::wait(double) () from /usr/lib/libfltk.so.1.1 #4 0x4dd313e4 in Fl::run() () from /usr/lib/libfltk.so.1.1 #5 0x0804af34 in main (argc=1, argv=0xbffff284) at hdspmixer.cxx:114 (gdb)

2 4

[alsa-devel] [PATCH] ALSA: HDA: Realtek: Fixup jack detection to input subsystem
by David Henningsson 28 Feb '11

28 Feb '11

-- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 28 Feb '11

28 Feb '11

The following changes since commit 9b34e6cc3bc2bf826c078c93c81b46f6c08da25f: ASoC: Add Freescale SGTL5000 codec support (2011-02-25 12:09:33 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Eric Bénard (2): eukrea-tlv320: fix platform_name ASoC: eukrea-tlv320: add MBIMXSD51 support Mark Brown (3): ASoC: Move WM2000 to dev_pm_ops Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Staticise non-exported symbols in SGTL5000 Zeng Zhaoming (1): ASoC: remove unnecessary header including in SGTL5000 codec driver sound/soc/codecs/sgtl5000.c | 7 +++---- sound/soc/codecs/sgtl5000.h | 2 -- sound/soc/codecs/wm2000.c | 14 +++++++------- sound/soc/imx/Kconfig | 3 ++- sound/soc/imx/eukrea-tlv320.c | 5 +++-- 5 files changed, 15 insertions(+), 16 deletions(-)

2 1

[alsa-devel] [PATCH 0/6] hdspmixer fixes
by Adrian Knoth 28 Feb '11

28 Feb '11

Hi! Here's a collection of fixes to hdspmixer. Besides style fixes, it re-enables support for Multiface/Digiface and other non-MADI cards. The support was accidently lost when merging hdspmixer64 with hdspmixer. Today, RME has provided me with two cards (Multiface and AES32), so I had the opportunity to fix the broken bits. Also, the old code never handled three cards correctly, but that's fixed, too. (and tested against the three RME cards currently in my workstation) Cheers Adrian Knoth (6): hdspmixer: Improve code coherency hdspmixer: Update FSF address. hdspmixer: Increase height of about window. hdspmixer: Bump version number to 1.10 hdspmixer: Fix ALSA snd_ctl_open error when running with three cards. hdspmixer: Fix metering for non-MADI cards hdspmixer/configure.in | 2 +- hdspmixer/src/HDSPMixerAboutText.cxx | 13 ++- hdspmixer/src/HDSPMixerCard.cxx | 163 ++--------------------- hdspmixer/src/HDSPMixerOutput.cxx | 4 +- hdspmixer/src/HDSPMixerSelector.cxx | 7 +- hdspmixer/src/HDSPMixerWindow.cxx | 75 +++++++---- hdspmixer/src/HDSPMixerWindow.h | 4 +- hdspmixer/src/channelmap.h | 234 +++++++++++++++++++++++++++++++--- hdspmixer/src/defines.h | 4 +- hdspmixer/src/hdspmixer.cxx | 3 + hdspmixer/src/mappings.h | 93 -------------- 11 files changed, 308 insertions(+), 294 deletions(-) delete mode 100644 hdspmixer/src/mappings.h -- 1.7.2.3

2 14

[alsa-devel] [PATCH 0/4] More hdspm fixes
by Adrian Knoth 28 Feb '11

28 Feb '11

Hi! Here are four more patches to hdspm. Most importantly, they add the missing PCI firmware revision for AES32. Without it, loading hdspm would crash on machines with these cards. To prevent such a crash upon unknown firmware revisions, I've created a second patch adding a default case to the firmware switch-case statement. Also worth mentioning: midi port initialization is now no longer hardcoded, thus fixing a bootup problem on MADIface. All patches have been runtime tested on MADIface, AES32, AES (PCIe) and RayDAT. Cheers Adrian Knoth (4): ALSA: hdspm - AES32: Use define instead of hardcoded channel count ALSA: hdspm - Add updated firmware revision for AES32 ALSA: hdspm - Abort on unknown firmware revision ALSA: hdspm - Fix midi port initialization sound/pci/rme9652/hdspm.c | 24 ++++++++++++++++++------ 1 files changed, 18 insertions(+), 6 deletions(-) -- 1.7.4.1

2 5

[alsa-devel] [PATCH] ASoC: Tegra: Fix error handling in DMA channel alloc
by Stephen Warren 28 Feb '11

28 Feb '11

tegra_dma_allocate_channel() returns NULL on errors, not an error pointer. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra_pcm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 40540b1..3c271f9 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -161,8 +161,8 @@ static int tegra_pcm_open(struct snd_pcm_substream *substream) prtd->dma_req[1].dev = prtd; prtd->dma_chan = tegra_dma_allocate_channel(TEGRA_DMA_MODE_ONESHOT); - if (IS_ERR(prtd->dma_chan)) { - ret = PTR_ERR(prtd->dma_chan); + if (prtd->dma_chan == NULL) { + ret = -ENOMEM; goto err; } -- 1.7.1

3 2

[alsa-devel] [PATCH] ASoC: Staticise non-exported symbols in SGTL5000
by Mark Brown 28 Feb '11

28 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b528f97..b7e97c0 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -124,7 +124,7 @@ static struct regulator_consumer_supply ldo_consumer[] = { REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), }; -struct regulator_init_data ldo_init_data = { +static struct regulator_init_data ldo_init_data = { .constraints = { .min_uV = 850000, .max_uV = 1600000, @@ -946,7 +946,7 @@ static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -struct snd_soc_dai_ops sgtl5000_ops = { +static struct snd_soc_dai_ops sgtl5000_ops = { .hw_params = sgtl5000_pcm_hw_params, .digital_mute = sgtl5000_digital_mute, .set_fmt = sgtl5000_set_dai_fmt, @@ -1421,7 +1421,7 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver sgtl5000_driver = { +static struct snd_soc_codec_driver sgtl5000_driver = { .probe = sgtl5000_probe, .remove = sgtl5000_remove, .suspend = sgtl5000_suspend, -- 1.7.2.3

2 1

[PATCH] ALSA: HDA: Realtek: Fixup jack detection to input subsystem
by David Henningsson 28 Feb '11

28 Feb '11

This patch fixes an error in the jack detection reporting, causing the jack detection sometimes not to be reported correctly to the input subsystem. It should apply to several Realtek codecs. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3328a25..c052fc5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1133,11 +1133,8 @@ static void alc_automute_speaker(struct hda_codec *codec, int pinctl) nid = spec->autocfg.hp_pins[i]; if (!nid) break; - if (snd_hda_jack_detect(codec, nid)) { - spec->jack_present = 1; - break; - } - alc_report_jack(codec, spec->autocfg.hp_pins[i]); + alc_report_jack(codec, nid); + spec->jack_present |= snd_hda_jack_detect(codec, nid); } mute = spec->jack_present ? HDA_AMP_MUTE : 0; -- 1.7.1 --------------060702070005020905070804--

1 0

Re: [alsa-devel] [PATCH v5] ASoC: Add Freescale SGTL5000 codec support
by Mark Brown 28 Feb '11

28 Feb '11

On Fri, Feb 25, 2011 at 11:56:52PM +0800, Zeng Zhaoming wrote: > On Fri, Feb 25, 2011 at 8:09 PM, Mark Brown > > This is going to fail on systems without voltage readback capabilities, > > including systems which have the regulator API disabled and stubbed out. > if the regulator driver not allow readback, I have no way to get the > voltage of vdda and vddio, > the voltage levels are critical to codec initialization. > Can you shed some light on howto around this? If you can't read the data back you need to guess what the configuration is somehow; I'd suggest just picking values and warning about what you're doing. > >> + memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), > >> + sgtl5000_dap_regs, > >> + SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); > > As has been pointed out on previous revisions of the patch there is no > > need to cast away from void in C and it can be actively harmful. > I have not remove void* since my toolchain ( gcc 4.4.4) complains > about the type mismatch, I will test > it with other version toolchain to confirm. As Timur indicated the reason this is complaining is that you're memcpy() is writing to const data. This is *clearly* buggy - the fact that casting away removes the warning is an example of why you should never have casts like this. You should take a copy of the array and modify that, having the defaults as const is correct.

1 0

[alsa-devel] [PATCH] ASoC: remove unnecessary header including in SGTL5000 codec driver
by zhaoming.zeng＠freescale.com 28 Feb '11

28 Feb '11

From: Zeng Zhaoming <b32542(a)freescale.com> Remove unnecessary headers: - mach/hardware.h in sgtl5000.c - linux/i2c.h in sgtl5000.h Signed-off-by: Zeng Zhaoming <b32542(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/sgtl5000.h | 2 -- 2 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9eb206e..b528f97 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -27,7 +27,6 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/initval.h> -#include <mach/hardware.h> #include "sgtl5000.h" diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index c29312a..eec3ab3 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -11,8 +11,6 @@ #ifndef _SGTL5000_H #define _SGTL5000_H -#include <linux/i2c.h> - /* * Register values. */ -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/2] acore: deprecated function in wrappers.c
by Grzegorz Daniluk 28 Feb '11

28 Feb '11

This patch replace the call of pci_find_device() to pci_get_device(). The function pci_find_device() was deprecated for quite a long time, and now it is no longer in the kernel. Signed-off-by: Grzegorz Daniluk <g.daniluk(a)elproma.com.pl> --- diff -uprN a/acore/wrappers.c b/acore/wrappers.c --- a/acore/wrappers.c 2011-01-27 12:09:20.000000000 +0000 +++ b/acore/wrappers.c 2011-02-25 13:41:46.000000000 +0000 @@ -197,7 +197,7 @@ EXPORT_SYMBOL(snd_compat_devfs_mk_cdev); int snd_pci_dev_present(const struct pci_device_id *ids) { while (ids->vendor || ids->subvendor) { - if (pci_find_device(ids->vendor, ids->subvendor, NULL)) + if (pci_get_device(ids->vendor, ids->subvendor, NULL)) return 1; ids++; }

2 5

[alsa-devel] [PATCH v20 0/3] ASoC/MFD/V4L2: WL1273 FM Radio driver
by Matti J. Aaltonen 28 Feb '11

28 Feb '11

Hello. And thanks for the comments. If we now move quickly enough we can get this thing in before it becomes deprecated... Samuel wrote: > for Mauro to take this one you'd have to provide a > diff against the already existing wl1273-core. I've made these patches against the existing stuff. >> + * Copyright (C) 2010 Nokia Corporation >2011. Changed. >> +} > I'm confused with this one: Isn't WL1273_VOLUME_SET a command ? Also, > how can reading from it set the volume ? It cannot... so I've changed it (back) to write. >> + i2c_set_clientdata(client, NULL) >Not needed. Removed. >> +err: >> + i2c_set_clientdata(client, NULL); >Ditto. Ditto. Cheers, Matti Matti J. Aaltonen (3): MFD: Wl1273 FM radio core: Add I2C IO functions. V4L2: Wl1273 FM Radio: Remove I2C IO functions. ASoC: WL1273 FM radio: Access I2C IO functions through pointers. drivers/media/radio/radio-wl1273.c | 360 +++++++++++------------------------- drivers/mfd/Kconfig | 2 +- drivers/mfd/wl1273-core.c | 149 +++++++++++++++- include/linux/mfd/wl1273-core.h | 2 + sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wl1273.c | 11 +- 6 files changed, 264 insertions(+), 262 deletions(-)

1 3

[alsa-devel] [PATCH - powermac] Allow input from microphone on Tumbler
by Risto Suominen 28 Feb '11

28 Feb '11

From: Risto Suominen <Risto.Suominen(a)gmail.com> Allow input from microphone on remaining PowerBooks and iBooks with Tumbler. Tested only with PowerBook G4 Gigabit Ethernet (PowerBook3,3). Signed-off-by: Risto Suominen <Risto.Suominen(a)gmail.com> --- Kernel version 2.6.32 (probably any 2.6), ALSA 1.0.22.1. The patch is available here, too: http://ristosu.wippiespace.com/pub/alsa-pmac-1.0.22.1-p1.diff --- a/alsa-kernel/ppc/pmac.c.org 2009-12-28 16:21:33.000000000 +0200 +++ b/alsa-kernel/ppc/pmac.c 2011-02-27 21:42:53.000000000 +0200 @@ -1034,7 +1034,11 @@ static int __devinit snd_pmac_detect(str if (of_device_is_compatible(sound, "tumbler")) { chip->model = PMAC_TUMBLER; chip->can_capture = machine_is_compatible("PowerMac4,2") - || machine_is_compatible("PowerBook4,1"); + || machine_is_compatible("PowerBook3,2") + || machine_is_compatible("PowerBook3,3") + || machine_is_compatible("PowerBook4,1") + || machine_is_compatible("PowerBook4,2") + || machine_is_compatible("PowerBook4,3"); chip->can_duplex = 0; // chip->can_byte_swap = 0; /* FIXME: check this */ chip->num_freqs = ARRAY_SIZE(tumbler_freqs);

2 1

[alsa-devel] [[PATCH] ALSA 1/1] Fix alsa-lib to regard 3D control - Depth as Playback Controls
by Raymond Yau 28 Feb '11

28 Feb '11

Fix alsa-lib to regard 3D control - Depth as Playback Controls This allow those mixer or recording application to find those controls which are really used for capturing

2 1

[alsa-devel] Problem when using silence 'trick'
by Bryan Ischo 28 Feb '11

28 Feb '11

Hi ALSA developers! I am working on my first project that uses sound under Linux, and I've chosen to implement directly to the ALSA API. I am amazed that I was able to get sound working in so few lines of code with ALSA. I like the API and its performance alot but I have a problem! My program, early in its development, wanted to play just one sound stream at a time; and it is a sound stream which normally delivers audio at a sufficent rate to avoid XRUN but in some circumstances the thread generating audio cannot keep up and it falls behind, resulting in XRUN. Well I found that in cases where the audio thread is 'almost' keeping up, the standard behavior of allowing an XRUN to occur, detecting it on the next call to snd_pcm_writei, and 'fixing' it by calling snd_pcm_prepare() and then proceeding to continue delivering audio frames, had the following unfortunate problems: 1. The audio sounds crackly and awful (which is to be expected to some degree with XRUN) 2. The time taken to re-prepare the ALSA device is counterproductive; it takes even more time away from the thread that is delivering audio and thus makes the case where the audio is very marginally behind 'real time' much worse. I thought, if only I could tell ALSA to just play silence if it has run out of frames of audio instead of XRUNning, and then I can add real frames of audio back in as soon as they are available and pick up the audio stream where it left off. Some digging into the ALSA API and I found that someone on the ALSA team had already beaten me to the thought! The API docs kind of hint at how you would go about doing this, and I found this technique to be very successful: 1. Use snd_pcm_sw_params_set_stop_threshold to set the stop threshold to 0. Now, no XRUN - ALSA will keep looping in the circular buffer and play whatever is there; and if you haven't updated quickly enough, you just get looping snippets of audio. 2. Use snd_pcm_sw_params_set_silence_threshold to 0 and snd_pcm_sw_params_set_silence_size to set the silence 'size' to the boundary value obtained by calling snd_pcm_sw_params_get_boundary(). Now ALSA is instructed to fill the already-played portions of the circular buffer with silence, which means that even when looping due to XRUN, no sound is played over and over, instead silence is played. Hey presto - works perfectly! Then later on I needed to play more than one sound simultaneously. It turns out that I had been opening the "plughw:0,0" device originally and this had been working great with my sound trick. The problem is, you cannot use plughw:0,0 and open it more than once to play more than one sound simultaneously; the second open will fail. So after some digging I found that the device I really should have been using all along is called "default" (hey, you'd think it would be obvious, but somehow I never found ALSA documentation that told me to do this). Now the real problem: when I use the "default" device with my silence technique, I have no problem playing multiple audio streams simultaneously by opening the "default" device multiple times and simultaneously playing streams into the different devices. BUT, whenever I cannot deliver audio fast enough to a stream, and the silence trick takes over and the stream plays some silence - boom, the device goes silent forever. I don't get any errors from any ALSA function, I just get silence no matter what I write into the device after that point. This is in sharp contrast to the plughw:0,0 device, which does play silence if I don't deliver audio fast enough but immediately plays whatever audio I do end up delivering to it when I finally have audio to deliver. Am I using an invalid technique with this silence trick? Is this a known and expected behavior of the "default" device? Or could something else be going on? Thanks for any advice you can give! Best wishes, Bryan

2 2

[alsa-devel] Tascam M-164UF module request/information
by Dan Swain 28 Feb '11

28 Feb '11

Dear ALSA developers, After talking with nevyn on the ALSA IRC channel on Freenode, I have been recommended to submit the device information to this list so that the developers can take it further! The pastebin for this device is at: http://alsa.pastebin.com/YsMDfyxi (a lsusb -v -d 0644: output) The device shows up in dmesg, but so far ALSA is unaware of the device. I don't have much programming experience, but if I can offer any help, please let me know and I'll do what I can :-) Many Thanks, Dan PS: Official information for the device can be found at: http://www.tascam.de/en/m-164.html

1 0

[alsa-devel] [PATCH 17/17]sound:core:seq:seq_ports.c Remove one to many n's in a word.
by Justin P. Mattock 27 Feb '11

27 Feb '11

The Patch below removes one to many "n's" in a word.. Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> CC: alsa-devel(a)alsa-project.org --- sound/core/seq/seq_ports.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 3bf7d73..e12bcd9 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -412,7 +412,7 @@ int snd_seq_get_port_info(struct snd_seq_client_port * port, * initialization or termination of devices (see seq_midi.c). * * If callback_all option is set, the callback function is invoked - * at each connnection/disconnection. + * at each connection/disconnection. */ static int subscribe_port(struct snd_seq_client *client, -- 1.7.4.1

2 1

Re: [alsa-devel] [PATCH] intel8x0m: schedule timeout before warm reset check
by Takashi Iwai 27 Feb '11

27 Feb '11

At Sun, 27 Feb 2011 01:23:58 +0100, Paul Bolle wrote: > > At every resume a laptop I use prints this message (at KERN_ERR level): > ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? > [0x2] > > The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what > seems to be happening is that the register involved indicated a warm > reset for some time (as the ICH_AC97WARM bit was set) but by the time > the warning is printed, and that same register is checked again, that > bit is already cleared and only the ICH_AC97COLD bit is still set. > > It turns out a warm reset needs some time to settle, but it is currently > checked right away. The test therefore fails the first time it is done > and schedule_timeout_uninterruptible() will be called. Once we return > from that jiffies is already (far) past end_time on this laptop, so we > exit the loop, print a warning, and exit the function while the warm > reset actually succeeded. > > One way to fix this is to call schedule_timeout_uninterruptible() at the > start of the loop. (This might punish the "finish cold reset" case, but > on this laptop such a cold reset apparently never happens, so I can't > say for sure.) > > While we're at it drop the extra single tick from end_time, as it looks > rather silly. > > Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> > --- > The question I asked in the previous message ("snd_intel8x0m: error > message because return value is unused?") made little sense. I probably > got quite confused by my own efforts to debug this. It'd be better to put a more proper delay time than tick, e.g. usleep_range() or whatever. schedule_timeout(1) can be a different length depending on HZ. Could you figure out? thanks, Takashi > > sound/pci/intel8x0m.c | 4 ++-- > 1 files changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c > index 13cec1e..9abff7b 100644 > --- a/sound/pci/intel8x0m.c > +++ b/sound/pci/intel8x0m.c > @@ -894,11 +894,11 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) > /* finish cold or do warm reset */ > cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; > iputdword(chip, ICHREG(GLOB_CNT), cnt); > - end_time = (jiffies + (HZ / 4)) + 1; > + end_time = jiffies + HZ / 4; > do { > + schedule_timeout_uninterruptible(1); > if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0) > goto __ok; > - schedule_timeout_uninterruptible(1); > } while (time_after_eq(end_time, jiffies)); > snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n", > igetdword(chip, ICHREG(GLOB_CNT))); > -- > 1.7.4.1 > > > >

1 0

[alsa-devel] ASoC update for 2.6.38
by Mark Brown 26 Feb '11

26 Feb '11

The following changes since commit 4bfc4e2508234f9149fd33fae853e99fb9e4a75b: ASoC: correct pxa AC97 DAI names (2011-02-25 11:42:18 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 Eric Bénard (1): eukrea-tlv320: fix platform_name sound/soc/imx/eukrea-tlv320.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-)

2 1

Re: [alsa-devel] [PATCH v5] ASoC: Add Freescale SGTL5000 codec support
by Timur Tabi 26 Feb '11

26 Feb '11

On Fri, Feb 25, 2011 at 9:56 AM, Zeng Zhaoming <zengzm.kernel(a)gmail.com> wrote: > I have not remove void* since my toolchain ( gcc 4.4.4) complains > about the type mismatch, I will test > it with other version toolchain to confirm. That's because you declared sgtl5000_regs[] as "const", which it isn't. The compiler is trying to tell you that you made a mistake. -- Timur Tabi Linux kernel developer at Freescale

1 0

Re: [alsa-devel] [PATCH v5] ASoC: Add Freescale SGTL5000 codec support
by Timur Tabi 25 Feb '11

25 Feb '11

On Fri, Feb 25, 2011 at 9:56 AM, Zeng Zhaoming <zengzm.kernel(a)gmail.com> wrote: >>> + memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), >>> + sgtl5000_dap_regs, >>> + SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); >> >> As has been pointed out on previous revisions of the patch there is no >> need to cast away from void in C and it can be actively harmful. >> > I have not remove void* since my toolchain ( gcc 4.4.4) complains > about the type mismatch, I will test > it with other version toolchain to confirm. It would probably be simpler if you did this instead: memcpy((void *)sgtl5000_regs + SGTL5000_DAP_REG_OFFSET, IMHO, you should be using a struct instead of an array for stgtl5000_regs. -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] ASoC updates for 2.6.39
by Mark Brown 25 Feb '11

25 Feb '11

The following changes since commit fadddc8753ccfab26ee57f3205d6926fe4be1350: ASoC: Add kerneldoc for jack_status_check callback (2011-02-18 09:14:24 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.39 Dimitris Papastamos (1): ASoC: soc-dapm: Include quotes around contents in debugfs entries Dmitry Eremin-Solenikov (1): ASoC: correct pxa AC97 DAI names Justin P. Mattock (1): ASoC: remove one to many l's in the word Mark Brown (15): ASoC: Mark WM8958 microphone detection registers readable ASoC: Hook wm_hubs micbiases up to CLK_SYS ASoC: Fix WM8958 default microphone detection argument ordering ASoC: Correct definition of WM8903_VMID_RES_5K ASoC: Hook wm_hubs micbiases up to CLK_SYS Merge branch 'for-2.6.38' into for-2.6.39 ASoC: Remove card from snd_soc_dapm_set_bias_level() ASoC: Run bias level changes for all DAPM contexts in parallel ASoC: Mark WM8958 microphone bias registers as readable ASoC: Support WM8958 direct microphone detection IRQ ASoC: Support configuration of WM8958 microphone bias analogue parameters ASoC: Simplify default WM8958 jack detection code ASoC: Pass the jack to jack notifiers ASoC: Remove -codec suffix from WM9081 driver Merge branch 'for-2.6.38' into for-2.6.39 Stephen Warren (1): ASoC: Tegra: Move utilities to separate module Zeng Zhaoming (1): ASoC: Add Freescale SGTL5000 codec support include/linux/mfd/wm8994/pdata.h | 12 +- include/linux/mfd/wm8994/registers.h | 2 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sgtl5000.c | 1512 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 402 +++++++++ sound/soc/codecs/wm8903.h | 2 +- sound/soc/codecs/wm8994-tables.c | 10 +- sound/soc/codecs/wm8994.c | 105 ++-- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm_hubs.c | 3 + sound/soc/pxa/e740_wm9705.c | 4 +- sound/soc/pxa/e750_wm9705.c | 4 +- sound/soc/pxa/e800_wm9712.c | 4 +- sound/soc/pxa/em-x270.c | 4 +- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/palm27x.c | 4 +- sound/soc/pxa/tosa.c | 4 +- sound/soc/pxa/zylonite.c | 4 +- sound/soc/samsung/dma.c | 2 +- sound/soc/soc-dapm.c | 134 ++-- sound/soc/soc-jack.c | 2 +- sound/soc/tegra/Makefile | 3 +- sound/soc/tegra/tegra_asoc_utils.c | 6 + 24 files changed, 2091 insertions(+), 145 deletions(-) create mode 100644 sound/soc/codecs/sgtl5000.c create mode 100644 sound/soc/codecs/sgtl5000.h

2 1

[alsa-devel] [PATCH] ASOC: missing files in alsa-driver
by Grzegorz Daniluk 25 Feb '11

25 Feb '11

This patch adds few files missing in soc/codecs and soc/sh. Without them the compilation of alsa-driver with those soc-s is impossible. Signed-off-by: Grzegorz Daniluk <g.daniluk(a)elproma.com.pl> --- diff -uprN a/soc/codecs/da7210.c b/soc/codecs/da7210.c --- a/soc/codecs/da7210.c 1970-01-01 00:00:00.000000000 +0000 +++ b/soc/codecs/da7210.c 2011-02-25 13:48:26.000000000 +0000 @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/soc/codecs/da7210.c" diff -uprN a/soc/sh/fsi-da7210.c b/soc/sh/fsi-da7210.c --- a/soc/sh/fsi-da7210.c 1970-01-01 00:00:00.000000000 +0000 +++ b/soc/sh/fsi-da7210.c 2011-02-25 14:38:05.485218100 +0000 @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/soc/sh/fsi-da7210.c" diff -uprN a/soc/sh/fsi.c b/soc/sh/fsi.c --- a/soc/sh/fsi.c 1970-01-01 00:00:00.000000000 +0000 +++ b/soc/sh/fsi.c 2011-02-25 13:51:21.000000000 +0000 @@ -0,0 +1,2 @@ +#include "adriver.h" +#include "../../alsa-kernel/soc/sh/fsi.c"

1 0

[alsa-devel] [PATCH] eukrea-tlv320: fix platform_name
by Eric Bénard 25 Feb '11

25 Feb '11

commit f0fba2ad1b6b53d5360125c41953b7afcd6deff0 included a mistake on the name of the platform in the snd_soc_dai_link structure. Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/eukrea-tlv320.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index e20c9e1..1e9bcca 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -79,7 +79,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-pcm-audio.0", + .platform_name = "imx-fiq-pcm-audio.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, -- 1.7.4

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[alsa-devel] [PATCH] ASoC: Move WM2000 to dev_pm_ops
by Mark Brown 25 Feb '11

25 Feb '11

There's a general move to use dev_pm_ops rather than bus specific functions in order to facilitate work on the PM core. Do this conversion to WM2000. The driver ought to be updated to work better in a multi-component model but the mechanical conversion ensures that we avoid blocking PM core work until that happens. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 14 +++++++------- 1 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 80ddf4f..a3b9cbb 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -836,24 +836,25 @@ static void wm2000_i2c_shutdown(struct i2c_client *i2c) } #ifdef CONFIG_PM -static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg) +static int wm2000_i2c_suspend(struct device *dev) { + struct i2c_client *i2c = to_i2c_client(dev); struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); return wm2000_anc_transition(wm2000, ANC_OFF); } -static int wm2000_i2c_resume(struct i2c_client *i2c) +static int wm2000_i2c_resume(struct device *dev) { + struct i2c_client *i2c = to_i2c_client(dev); struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev); return wm2000_anc_set_mode(wm2000); } -#else -#define wm2000_i2c_suspend NULL -#define wm2000_i2c_resume NULL #endif +static SIMPLE_DEV_PM_OPS(wm2000_pm, wm2000_i2c_suspend, wm2000_i2c_resume); + static const struct i2c_device_id wm2000_i2c_id[] = { { "wm2000", 0 }, { } @@ -864,11 +865,10 @@ static struct i2c_driver wm2000_i2c_driver = { .driver = { .name = "wm2000", .owner = THIS_MODULE, + .pm = &wm2000_pm, }, .probe = wm2000_i2c_probe, .remove = __devexit_p(wm2000_i2c_remove), - .suspend = wm2000_i2c_suspend, - .resume = wm2000_i2c_resume, .shutdown = wm2000_i2c_shutdown, .id_table = wm2000_i2c_id, }; -- 1.7.2.3

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[alsa-devel] [PATCH] eukrea-tlv320: add MBIMXSD51 support
by Eric Bénard 25 Feb '11

25 Feb '11

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 3 ++- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 642270a..9eeb8f0 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -44,7 +44,8 @@ config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ - || MACH_EUKREA_MBIMXSD35_BASEBOARD + || MACH_EUKREA_MBIMXSD35_BASEBOARD \ + || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 1e9bcca..75fb4b8 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -98,7 +98,8 @@ static int __init eukrea_tlv320_init(void) int ret; if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd() - && !machine_is_eukrea_cpuimx35sd()) + && !machine_is_eukrea_cpuimx35sd() + && !machine_is_eukrea_cpuimx51sd()) /* return happy. We might run on a totally different machine */ return 0; -- 1.7.4

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[alsa-devel] [PATCH v5] ASoC: Add Freescale SGTL5000 codec support
by zhaoming.zeng＠freescale.com 25 Feb '11

25 Feb '11

From: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> Add Freescale SGTL5000 codec support. Supported features: - line-in and mic input - headphone and line-out output - line-in bypass ADC and DAC to headphone - 16, 20, 24, 32 bit audio - 8 ~ 96k sample rates Signed-off-by: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> --- Changes since v4: 1. Fix regulator resource leak in sgtl5000_enable_regulators(). 2. Fix memory leak in ldo_regulator_register() when desc.name malloc failed. 3. Del snd_soc_dapm_free() in sgtl5000_remove(). 4. Correct the typos. 5. Replace pr_err() with dev_err in sgtl5000_set_clock(). 6. Fix code style. Thanks for the review of Timur, Mark, Sascha and Arnaud. --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sgtl5000.c | 1512 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sgtl5000.h | 402 ++++++++++++ 4 files changed, 1921 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e239345..c04da18 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 + select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2602 if I2C @@ -182,6 +183,10 @@ config SND_SOC_MAX98088 config SND_SOC_PCM3008 tristate +#Freescale sgtl5000 codec +config SND_SOC_SGTL5000 + tristate + config SND_SOC_SN95031 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ae10507..3bbb08c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-objs := spdif_transciever.o @@ -103,6 +104,7 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c new file mode 100644 index 0000000..9eb206e --- /dev/null +++ b/sound/soc/codecs/sgtl5000.c @@ -0,0 +1,1512 @@ +/* + * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/regulator/driver.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/consumer.h> +#include <sound/core.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <mach/hardware.h> + +#include "sgtl5000.h" + +#define SGTL5000_DAP_REG_OFFSET 0x0100 +#define SGTL5000_MAX_REG_OFFSET 0x013A + +/* default value of sgtl5000 registers except DAP */ +static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = { + 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ + 0x0000, /* 0x0002, CHIP_DIG_POWER. */ + 0x0008, /* 0x0004, CHIP_CKL_CTRL */ + 0x0010, /* 0x0006, CHIP_I2S_CTRL */ + 0x0000, /* 0x0008, reserved */ + 0x0008, /* 0x000A, CHIP_SSS_CTRL */ + 0x0000, /* 0x000C, reserved */ + 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ + 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ + 0x0000, /* 0x0012, reserved */ + 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ + 0x0000, /* 0x0016, reserved */ + 0x0000, /* 0x0018, reserved */ + 0x0000, /* 0x001A, reserved */ + 0x0000, /* 0x001E, reserved */ + 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ + 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ + 0x0111, /* 0x0024, CHIP_ANN_CTRL */ + 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ + 0x0000, /* 0x0028, CHIP_REF_CTRL */ + 0x0000, /* 0x002A, CHIP_MIC_CTRL */ + 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ + 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ + 0x7060, /* 0x0030, CHIP_ANA_POWER */ + 0x5000, /* 0x0032, CHIP_PLL_CTRL */ + 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ + 0x0000, /* 0x0036, CHIP_ANA_STATUS */ + 0x0000, /* 0x0038, reserved */ + 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ + 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ + 0x0000, /* reserved */ +}; + +/* default value of dap registers */ +static const u16 sgtl5000_dap_regs[] = { + 0x0000, /* 0x0100, DAP_CONTROL */ + 0x0000, /* 0x0102, DAP_PEQ */ + 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ + 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ + 0x0000, /* 0x0108, DAP_AUDIO_EQ */ + 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ + 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ + 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ + 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ + 0x0000, /* 0x0112, reserved */ + 0x0000, /* 0x0114, reserved */ + 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ + 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ + 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ + 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ + 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ + 0x8000, /* 0x0120, DAP_MAIN_CHAN */ + 0x0000, /* 0x0122, DAP_MIX_CHAN */ + 0x0510, /* 0x0124, DAP_AVC_CTRL */ + 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ + 0x0028, /* 0x0128, DAP_AVC_ATTACK */ + 0x0050, /* 0x012A, DAP_AVC_DECAY */ + 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ + 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ + 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ + 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ + 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ + 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ + 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ + 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ +}; + +/* regulator supplies for sgtl5000, VDDD is an optional external supply */ +enum sgtl5000_regulator_supplies { + VDDA, + VDDIO, + VDDD, + SGTL5000_SUPPLY_NUM +}; + +/* vddd is optional supply */ +static const char *supply_names[SGTL5000_SUPPLY_NUM] = { + "VDDA", + "VDDIO", + "VDDD" +}; + +#define LDO_CONSUMER_NAME "VDDD_LDO" +#define LDO_VOLTAGE 1200000 + +static struct regulator_consumer_supply ldo_consumer[] = { + REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL), +}; + +struct regulator_init_data ldo_init_data = { + .constraints = { + .min_uV = 850000, + .max_uV = 1600000, + .valid_modes_mask = REGULATOR_MODE_NORMAL, + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + }, + .num_consumer_supplies = 1, + .consumer_supplies = &ldo_consumer[0], +}; + +/* + * sgtl5000 internal ldo regulator, + * enabled when VDDD not provided + */ +struct ldo_regulator { + struct regulator_desc desc; + struct regulator_dev *dev; + int voltage; + void *codec_data; + bool enabled; +}; + +/* sgtl5000 private structure in codec */ +struct sgtl5000_priv { + int sysclk; /* sysclk rate */ + int master; /* i2s master or not */ + int fmt; /* i2s data format */ + struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM]; + struct ldo_regulator *ldo; +}; + +/* + * mic_bias power on/off share the same register bits with + * output impedance of mic bias, when power on mic bias, we + * need reclaim it to impedance value. + * 0x0 = Powered off + * 0x1 = 2Kohm + * 0x2 = 4Kohm + * 0x3 = 8Kohm + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias resistor to 4Kohm */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k); + break; + + case SND_SOC_DAPM_PRE_PMD: + /* + * SGTL5000_BIAS_R_8k as mask to clean the two bits + * of mic bias and output impedance + */ + snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL, + SGTL5000_BIAS_R_8k, 0); + break; + } + return 0; +} + +/* + * using codec assist to small pop, hp_powerup or lineout_powerup + * should stay setting until vag_powerup is fully ramped down, + * vag fully ramped down require 400ms. + */ +static int small_pop_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + break; + default: + break; + } + + return 0; +} + +/* input sources for ADC */ +static const char *adc_mux_text[] = { + "MIC_IN", "LINE_IN" +}; + +static const struct soc_enum adc_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); + +static const struct snd_kcontrol_new adc_mux = +SOC_DAPM_ENUM("Capture Mux", adc_enum); + +/* input sources for DAC */ +static const char *dac_mux_text[] = { + "DAC", "LINE_IN" +}; + +static const struct soc_enum dac_enum = +SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = +SOC_DAPM_ENUM("Headphone Mux", dac_enum); + +static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("LINE_IN"), + SND_SOC_DAPM_INPUT("MIC_IN"), + + SND_SOC_DAPM_OUTPUT("HP_OUT"), + SND_SOC_DAPM_OUTPUT("LINE_OUT"), + + SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, + small_pop_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + + /* aif for i2s input */ + SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", + 0, SGTL5000_CHIP_DIG_POWER, + 0, 0), + + /* aif for i2s output */ + SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture", + 0, SGTL5000_CHIP_DIG_POWER, + 1, 0), + + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + + SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), +}; + +/* routes for sgtl5000 */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ + {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + + {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ + {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + + {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ + {"LO", NULL, "DAC"}, /* dac --> line_out */ + + {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ + {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ + + {"LINE_OUT", NULL, "LO"}, + {"HP_OUT", NULL, "HP"}, +}; + +/* custom function to fetch info of PCM playback volume */ +static int dac_info_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xfc - 0x3c; + return 0; +} + +/* + * custom function to get of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * register value map to userspace value + * + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + * ------------------------------ + * userspace value 0xc0 0 + */ +static int dac_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg; + int l; + int r; + + reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL); + + /* get left channel volume */ + l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT; + + /* get right channel volume */ + r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT; + + /* make sure value fall in (0x3c,0xfc) */ + l = clamp(l, 0x3c, 0xfc); + r = clamp(r, 0x3c, 0xfc); + + /* invert it and map to userspace value */ + l = 0xfc - l; + r = 0xfc - r; + + ucontrol->value.integer.value[0] = l; + ucontrol->value.integer.value[1] = r; + + return 0; +} + +/* + * custom function to put of PCM playback volume + * + * dac volume register + * 15-------------8-7--------------0 + * | R channel vol | L channel vol | + * ------------------------------- + * + * PCM volume with 0.5017 dB steps from 0 to -90 dB + * + * register values map to dB + * 0x3B and less = Reserved + * 0x3C = 0 dB + * 0x3D = -0.5 dB + * 0xF0 = -90 dB + * 0xFC and greater = Muted + * + * userspace value map to register value + * + * userspace value 0xc0 0 + * ------------------------------ + * register value 0x3c(0dB) 0xf0(-90dB)0xfc + */ +static int dac_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int reg; + int l; + int r; + + l = ucontrol->value.integer.value[0]; + r = ucontrol->value.integer.value[1]; + + /* make sure userspace volume fall in (0, 0xfc-0x3c) */ + l = clamp(l, 0, 0xfc - 0x3c); + r = clamp(r, 0, 0xfc - 0x3c); + + /* invert it, get the value can be set to register */ + l = 0xfc - l; + r = 0xfc - r; + + /* shift to get the register value */ + reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT | + r << SGTL5000_DAC_VOL_RIGHT_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg); + + return 0; +} + +static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0); + +/* tlv for mic gain, 0db 20db 30db 40db */ +static const unsigned int mic_gain_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0), +}; + +/* tlv for hp volume, -51.5db to 12.0db, step .5db */ +static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); + +static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { + /* SOC_DOUBLE_S8_TLV with invert */ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = dac_info_volsw, + .get = dac_get_volsw, + .put = dac_put_volsw, + }, + + SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), + SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)", + SGTL5000_CHIP_ANA_ADC_CTRL, + 8, 2, 0, capture_6db_attenuate), + SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + + SOC_DOUBLE_TLV("Headphone Playback Volume", + SGTL5000_CHIP_ANA_HP_CTRL, + 0, 8, + 0x7f, 1, + headphone_volume), + SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL, + 5, 1, 0), + + SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, + 0, 4, 0, mic_gain_tlv), +}; + +/* mute the codec used by alsa core */ +static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + adcdac_ctrl, mute ? adcdac_ctrl : 0); + + return 0; +} + +/* set codec format */ +static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + u16 i2sctl = 0; + + sgtl5000->master = 0; + /* + * i2s clock and frame master setting. + * ONLY support: + * - clock and frame slave, + * - clock and frame master + */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2sctl |= SGTL5000_I2S_MASTER; + sgtl5000->master = 1; + break; + default: + return -EINVAL; + } + + /* setting i2s data format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + i2sctl |= SGTL5000_I2S_MODE_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + case SND_SOC_DAIFMT_I2S: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + break; + case SND_SOC_DAIFMT_RIGHT_J: + i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_LRPOL; + break; + case SND_SOC_DAIFMT_LEFT_J: + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_LRALIGN; + break; + default: + return -EINVAL; + } + + sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + i2sctl |= SGTL5000_I2S_SCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl); + + return 0; +} + +/* set codec sysclk */ +static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case SGTL5000_SYSCLK: + sgtl5000->sysclk = freq; + break; + default: + return -EINVAL; + } + + return 0; +} + +/* + * set clock according to i2s frame clock, + * sgtl5000 provide 2 clock sources. + * 1. sys_mclk. sample freq can only configure to + * 1/256, 1/384, 1/512 of sys_mclk. + * 2. pll. can derive any audio clocks. + * + * clock setting rules: + * 1. in slave mode, only sys_mclk can use. + * 2. as constraint by sys_mclk, sample freq should + * set to 32k, 44.1k and above. + * 3. using sys_mclk prefer to pll to save power. + */ +static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int clk_ctl = 0; + int sys_fs; /* sample freq */ + + /* + * sample freq should be divided by frame clock, + * if frame clock lower than 44.1khz, sample feq should set to + * 32khz or 44.1khz. + */ + switch (frame_rate) { + case 8000: + case 16000: + sys_fs = 32000; + break; + case 11025: + case 22050: + sys_fs = 44100; + break; + default: + sys_fs = frame_rate; + break; + } + + /* set divided factor of frame clock */ + switch (sys_fs / frame_rate) { + case 4: + clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT; + break; + case 2: + clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT; + break; + case 1: + clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT; + break; + default: + return -EINVAL; + } + + /* set the sys_fs according to frame rate */ + switch (sys_fs) { + case 32000: + clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT; + break; + case 44100: + clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT; + break; + case 48000: + clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT; + break; + case 96000: + clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT; + break; + default: + dev_err(codec->dev, "frame rate %d not supported\n", + frame_rate); + return -EINVAL; + } + + /* + * calculate the divider of mclk/sample_freq, + * factor of freq =96k can only be 256, since mclk in range (12m,27m) + */ + switch (sgtl5000->sysclk / sys_fs) { + case 256: + clk_ctl |= SGTL5000_MCLK_FREQ_256FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 384: + clk_ctl |= SGTL5000_MCLK_FREQ_384FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + case 512: + clk_ctl |= SGTL5000_MCLK_FREQ_512FS << + SGTL5000_MCLK_FREQ_SHIFT; + break; + default: + /* if mclk not satisify the divider, use pll */ + if (sgtl5000->master) { + clk_ctl |= SGTL5000_MCLK_FREQ_PLL << + SGTL5000_MCLK_FREQ_SHIFT; + } else { + dev_err(codec->dev, + "PLL not supported in slave mode\n"); + return -EINVAL; + } + } + + /* if using pll, please check manual 6.4.2 for detail */ + if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) { + u64 out, t; + int div2; + int pll_ctl; + unsigned int in, int_div, frac_div; + + if (sgtl5000->sysclk > 17000000) { + div2 = 1; + in = sgtl5000->sysclk / 2; + } else { + div2 = 0; + in = sgtl5000->sysclk; + } + if (sys_fs == 44100) + out = 180633600; + else + out = 196608000; + t = do_div(out, in); + int_div = out; + t *= 2048; + do_div(t, in); + frac_div = t; + pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT | + frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT; + + snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl); + if (div2) + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + SGTL5000_INPUT_FREQ_DIV2); + else + snd_soc_update_bits(codec, + SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INPUT_FREQ_DIV2, + 0); + + /* power up pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + } else { + /* power down pll */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, + 0); + } + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + + return 0; +} + +/* + * Set PCM DAI bit size and sample rate. + * input: params_rate, params_fmt + */ +static int sgtl5000_pcm_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_codec *codec = rtd->codec; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + int channels = params_channels(params); + int i2s_ctl = 0; + int stereo; + int ret; + + /* sysclk should already set */ + if (!sgtl5000->sysclk) { + dev_err(codec->dev, "%s: set sysclk first!\n", __func__); + return -EFAULT; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stereo = SGTL5000_DAC_STEREO; + else + stereo = SGTL5000_ADC_STEREO; + + /* set mono to save power */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo, + channels == 1 ? 0 : stereo); + + /* set codec clock base on lrclk */ + ret = sgtl5000_set_clock(codec, params_rate(params)); + if (ret) + return ret; + + /* set i2s data format */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S24_LE: + i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J) + return -EINVAL; + i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT; + i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS << + SGTL5000_I2S_SCLKFREQ_SHIFT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl); + + return 0; +} + +static int ldo_regulator_is_enabled(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->enabled; +} + +static int ldo_regulator_enable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + int reg; + + if (ldo_regulator_is_enabled(dev)) + return 0; + + /* set regulator value firstly */ + reg = (1600 - ldo->voltage / 1000) / 50; + reg = clamp(reg, 0x0, 0xf); + + /* amend the voltage value, unit: uV */ + ldo->voltage = (1600 - reg * 50) * 1000; + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, reg); + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + SGTL5000_LINEREG_D_POWERUP); + + /* when internal ldo enabled, simple digital power can be disabled */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + + ldo->enabled = 1; + return 0; +} + +static int ldo_regulator_disable(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data; + + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINEREG_D_POWERUP, + 0); + + /* clear voltage info */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0); + + ldo->enabled = 0; + + return 0; +} + +static int ldo_regulator_get_voltage(struct regulator_dev *dev) +{ + struct ldo_regulator *ldo = rdev_get_drvdata(dev); + + return ldo->voltage; +} + +static struct regulator_ops ldo_regulator_ops = { + .is_enabled = ldo_regulator_is_enabled, + .enable = ldo_regulator_enable, + .disable = ldo_regulator_disable, + .get_voltage = ldo_regulator_get_voltage, +}; + +static int ldo_regulator_register(struct snd_soc_codec *codec, + struct regulator_init_data *init_data, + int voltage) +{ + struct ldo_regulator *ldo; + + ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL); + + if (!ldo) { + dev_err(codec->dev, "failed to allocate ldo_regulator\n"); + return -ENOMEM; + } + + ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL); + if (!ldo->desc.name) { + kfree(ldo); + dev_err(codec->dev, "failed to allocate decs name memory\n"); + return -ENOMEM; + } + + ldo->desc.type = REGULATOR_VOLTAGE; + ldo->desc.owner = THIS_MODULE; + ldo->desc.ops = &ldo_regulator_ops; + ldo->desc.n_voltages = 1; + + ldo->codec_data = codec; + ldo->voltage = voltage; + + ldo->dev = regulator_register(&ldo->desc, codec->dev, + init_data, ldo); + if (IS_ERR(ldo->dev)) { + dev_err(codec->dev, "failed to register regulator\n"); + kfree(ldo->desc.name); + kfree(ldo); + + return PTR_ERR(ldo->dev); + } + + return 0; +} + +static int ldo_regulator_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + struct ldo_regulator *ldo = sgtl5000->ldo; + + if (!ldo) + return 0; + + regulator_unregister(ldo->dev); + kfree(ldo->desc.name); + kfree(ldo); + + return 0; +} + +/* + * set dac bias + * common state changes: + * startup: + * off --> standby --> prepare --> on + * standby --> prepare --> on + * + * stop: + * on --> prepare --> standby + */ +static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + return ret; + udelay(10); + } + + break; + case SND_SOC_BIAS_OFF: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +struct snd_soc_dai_ops sgtl5000_ops = { + .hw_params = sgtl5000_pcm_hw_params, + .digital_mute = sgtl5000_digital_mute, + .set_fmt = sgtl5000_set_dai_fmt, + .set_sysclk = sgtl5000_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver sgtl5000_dai = { + .name = "sgtl5000", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + /* + * only support 8~48K + 96K, + * TODO modify hw_param to support more + */ + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000, + .formats = SGTL5000_FORMATS, + }, + .ops = &sgtl5000_ops, + .symmetric_rates = 1, +}; + +static int sgtl5000_volatile_register(struct snd_soc_codec *codec, + unsigned int reg) +{ + switch (reg) { + case SGTL5000_CHIP_ID: + case SGTL5000_CHIP_ADCDAC_CTRL: + case SGTL5000_CHIP_ANA_STATUS: + return 1; + } + + return 0; +} + +#ifdef CONFIG_SUSPEND +static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +/* + * restore all sgtl5000 registers, + * since a big hole between dap and regular registers, + * we will restore them respectively. + */ +static int sgtl5000_restore_regs(struct snd_soc_codec *codec) +{ + u16 *cache = codec->reg_cache; + int i; + int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1; + + /* restore regular registers */ + for (i = 0; i < regular_regs; i++) { + int reg = i << 1; + + /* this regs depends on the others */ + if (reg == SGTL5000_CHIP_ANA_POWER || + reg == SGTL5000_CHIP_CLK_CTRL || + reg == SGTL5000_CHIP_LINREG_CTRL || + reg == SGTL5000_CHIP_LINE_OUT_CTRL || + reg == SGTL5000_CHIP_CLK_CTRL) + continue; + + snd_soc_write(codec, reg, cache[i]); + } + + /* restore dap registers */ + for (i = SGTL5000_DAP_REG_OFFSET >> 1; + i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { + int reg = i << 1; + + snd_soc_write(codec, reg, cache[i]); + } + + /* + * restore power and other regs according + * to set_power() and set_clock() + */ + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, + cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, + cache[SGTL5000_CHIP_ANA_POWER >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, + cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, + cache[SGTL5000_CHIP_REF_CTRL >> 1]); + + snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + return 0; +} + +static int sgtl5000_resume(struct snd_soc_codec *codec) +{ + /* Bring the codec back up to standby to enable regulators */ + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Restore registers by cached in memory */ + sgtl5000_restore_regs(codec); + return 0; +} +#else +#define sgtl5000_suspend NULL +#define sgtl5000_resume NULL +#endif /* CONFIG_SUSPEND */ + +/* + * sgtl5000 has 3 internal power supplies: + * 1. VAG, normally set to vdda/2 + * 2. chargepump, set to different value + * according to voltage of vdda and vddio + * 3. line out VAG, normally set to vddio/2 + * + * and should be set according to: + * 1. vddd provided by external or not + * 2. vdda and vddio voltage value. > 3.1v or not + * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd. + */ +static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) +{ + int vddd; + int vdda; + int vddio; + u16 ana_pwr; + u16 lreg_ctrl; + int vag; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + vdda = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); + vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); + vddd = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); + + vdda = vdda / 1000; + vddio = vddio / 1000; + vddd = vddd / 1000; + + if (vdda <= 0 || vddio <= 0 || vddd < 0) { + dev_err(codec->dev, "regulator voltage not set correctly\n"); + + return -EINVAL; + } + + /* according to datasheet, maximum voltage of supplies */ + if (vdda > 3600 || vddio > 3600 || vddd > 1980) { + dev_err(codec->dev, + "exceed max voltage vdda %dmv vddio %dma vddd %dma\n", + vdda, vddio, vddd); + + return -EINVAL; + } + + /* reset value */ + ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER); + ana_pwr |= SGTL5000_DAC_STEREO | + SGTL5000_ADC_STEREO | + SGTL5000_REFTOP_POWERUP; + lreg_ctrl = snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL); + + if (vddio < 3100 && vdda < 3100) { + /* enable internal oscillator used for charge pump */ + snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL, + SGTL5000_INT_OSC_EN, + SGTL5000_INT_OSC_EN); + /* Enable VDDC charge pump */ + ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP; + } else if (vddio >= 3100 && vdda >= 3100) { + /* + * if vddio and vddd > 3.1v, + * charge pump should be clean before set ana_pwr + */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VDDC_CHRGPMP_POWERUP, 0); + + /* VDDC use VDDIO rail */ + lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD; + lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO << + SGTL5000_VDDC_MAN_ASSN_SHIFT; + } + + snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr); + + /* set voltage to register */ + snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, + (0x1 << 4) - 1, 0x8); + + /* + * if vddd linear reg has been enabled, + * simple digital supply should be clear to get + * proper VDDD voltage. + */ + if (ana_pwr & SGTL5000_LINEREG_D_POWERUP) + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP, + 0); + else + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_LINREG_SIMPLE_POWERUP | + SGTL5000_STARTUP_POWERUP, + 0); + + /* + * set ADC/DAC VAG to vdda / 2, + * should stay in range (0.8v, 1.575v) + */ + vag = vdda / 2; + if (vag <= SGTL5000_ANA_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP * + (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT)) + vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT; + else + vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + vag << SGTL5000_ANA_GND_SHIFT, + vag << SGTL5000_ANA_GND_SHIFT); + + /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */ + vag = vddio / 2; + if (vag <= SGTL5000_LINE_OUT_GND_BASE) + vag = 0; + else if (vag >= SGTL5000_LINE_OUT_GND_BASE + + SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX) + vag = SGTL5000_LINE_OUT_GND_MAX; + else + vag = (vag - SGTL5000_LINE_OUT_GND_BASE) / + SGTL5000_LINE_OUT_GND_STP; + + snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT, + vag << SGTL5000_LINE_OUT_GND_SHIFT | + SGTL5000_LINE_OUT_CURRENT_360u << + SGTL5000_LINE_OUT_CURRENT_SHIFT); + + return 0; +} + +static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) +{ + u16 reg; + int ret; + int rev; + int i; + int external_vddd = 0; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) + sgtl5000->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (!ret) + external_vddd = 1; + else { + /* set internal ldo to 1.2v */ + int voltage = LDO_VOLTAGE; + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) { + dev_err(codec->dev, + "Failed to register vddd internal supplies: %d\n", + ret); + return ret; + } + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + + /* wait for all power rails bring up */ + udelay(10); + + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; + goto err_regulator_disable; + } + + rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision %d\n", rev); + + /* + * workaround for revision 0x11 and later, + * roll back to use internal LDO + */ + if (external_vddd && rev >= 0x11) { + int voltage = LDO_VOLTAGE; + /* disable all regulator first */ + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + /* free VDDD regulator */ + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + + ret = ldo_regulator_register(codec, &ldo_init_data, voltage); + if (ret) + return ret; + + sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME; + + ret = regulator_bulk_get(codec->dev, + ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) { + ldo_regulator_remove(codec); + dev_err(codec->dev, + "Failed to request supplies: %d\n", ret); + + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (ret) + goto err_regulator_free; + + /* wait for all power rails bring up */ + udelay(10); + } + + return 0; + +err_regulator_disable: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); +err_regulator_free: + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + if (external_vddd) + ldo_regulator_remove(codec); + return ret; + +} + +static int sgtl5000_probe(struct snd_soc_codec *codec) +{ + int ret; + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + /* setup i2c data ops */ + ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + ret = sgtl5000_enable_regulators(codec); + if (ret) + return ret; + + /* power up sgtl5000 */ + ret = sgtl5000_set_power_regs(codec); + if (ret) + goto err; + + /* enable small pop, introduce 400ms delay in turning off */ + snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL, + SGTL5000_SMALL_POP, + SGTL5000_SMALL_POP); + + /* disable short cut detector */ + snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0); + + /* + * set i2s as default input of sound switch + * TODO: add sound switch to control and dapm widge. + */ + snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL, + SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); + snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER, + SGTL5000_ADC_EN | SGTL5000_DAC_EN); + + /* enable dac volume ramp by default */ + snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_VOL_RAMP_EN | + SGTL5000_DAC_MUTE_RIGHT | + SGTL5000_DAC_MUTE_LEFT); + + snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f); + + snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL, + SGTL5000_HP_ZCD_EN | + SGTL5000_ADC_ZCD_EN); + + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + + /* + * disable DAP + * TODO: + * Enable DAP in kcontrol and dapm. + */ + snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + + /* leading to standby state */ + ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (ret) + goto err; + + snd_soc_add_controls(codec, sgtl5000_snd_controls, + ARRAY_SIZE(sgtl5000_snd_controls)); + + snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets, + ARRAY_SIZE(sgtl5000_dapm_widgets)); + + snd_soc_dapm_add_routes(&codec->dapm, audio_map, + ARRAY_SIZE(audio_map)); + + snd_soc_dapm_new_widgets(&codec->dapm); + + return 0; + +err: + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + ldo_regulator_remove(codec); + + return ret; +} + +static int sgtl5000_remove(struct snd_soc_codec *codec) +{ + struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); + + sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF); + + regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), + sgtl5000->supplies); + ldo_regulator_remove(codec); + + return 0; +} + +struct snd_soc_codec_driver sgtl5000_driver = { + .probe = sgtl5000_probe, + .remove = sgtl5000_remove, + .suspend = sgtl5000_suspend, + .resume = sgtl5000_resume, + .set_bias_level = sgtl5000_set_bias_level, + .reg_cache_size = ARRAY_SIZE(sgtl5000_regs), + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, + .reg_cache_default = sgtl5000_regs, + .volatile_register = sgtl5000_volatile_register, +}; + +static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct sgtl5000_priv *sgtl5000; + int ret; + + sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + if (!sgtl5000) + return -ENOMEM; + + /* + * copy DAP default values to default value array. + * sgtl5000 register space has a big hole, merge it + * at init phase makes life easy. + * FIXME: should we drop 'const' of sgtl5000_regs? + */ + memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), + sgtl5000_dap_regs, + SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); + + i2c_set_clientdata(client, sgtl5000); + + ret = snd_soc_register_codec(&client->dev, + &sgtl5000_driver, &sgtl5000_dai, 1); + if (ret) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + kfree(sgtl5000); + return ret; + } + + return 0; +} + +static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) +{ + struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + kfree(sgtl5000); + return 0; +} + +static const struct i2c_device_id sgtl5000_id[] = { + {"sgtl5000", 0}, + {}, +}; + +MODULE_DEVICE_TABLE(i2c, sgtl5000_id); + +static struct i2c_driver sgtl5000_i2c_driver = { + .driver = { + .name = "sgtl5000", + .owner = THIS_MODULE, + }, + .probe = sgtl5000_i2c_probe, + .remove = __devexit_p(sgtl5000_i2c_remove), + .id_table = sgtl5000_id, +}; + +static int __init sgtl5000_modinit(void) +{ + return i2c_add_driver(&sgtl5000_i2c_driver); +} +module_init(sgtl5000_modinit); + +static void __exit sgtl5000_exit(void) +{ + i2c_del_driver(&sgtl5000_i2c_driver); +} +module_exit(sgtl5000_exit); + +MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng(a)freescale.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h new file mode 100644 index 0000000..c29312a --- /dev/null +++ b/sound/soc/codecs/sgtl5000.h @@ -0,0 +1,402 @@ +/* + * sgtl5000.h - SGTL5000 audio codec interface + * + * Copyright 2010-2011 Freescale Semiconductor, Inc. + * + * 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. + */ + +#ifndef _SGTL5000_H +#define _SGTL5000_H + +#include <linux/i2c.h> + +/* + * Register values. + */ +#define SGTL5000_CHIP_ID 0x0000 +#define SGTL5000_CHIP_DIG_POWER 0x0002 +#define SGTL5000_CHIP_CLK_CTRL 0x0004 +#define SGTL5000_CHIP_I2S_CTRL 0x0006 +#define SGTL5000_CHIP_SSS_CTRL 0x000a +#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e +#define SGTL5000_CHIP_DAC_VOL 0x0010 +#define SGTL5000_CHIP_PAD_STRENGTH 0x0014 +#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020 +#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022 +#define SGTL5000_CHIP_ANA_CTRL 0x0024 +#define SGTL5000_CHIP_LINREG_CTRL 0x0026 +#define SGTL5000_CHIP_REF_CTRL 0x0028 +#define SGTL5000_CHIP_MIC_CTRL 0x002a +#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c +#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e +#define SGTL5000_CHIP_ANA_POWER 0x0030 +#define SGTL5000_CHIP_PLL_CTRL 0x0032 +#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034 +#define SGTL5000_CHIP_ANA_STATUS 0x0036 +#define SGTL5000_CHIP_SHORT_CTRL 0x003c +#define SGTL5000_CHIP_ANA_TEST2 0x003a +#define SGTL5000_DAP_CTRL 0x0100 +#define SGTL5000_DAP_PEQ 0x0102 +#define SGTL5000_DAP_BASS_ENHANCE 0x0104 +#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106 +#define SGTL5000_DAP_AUDIO_EQ 0x0108 +#define SGTL5000_DAP_SURROUND 0x010a +#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c +#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e +#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110 +#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116 +#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118 +#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a +#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c +#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e +#define SGTL5000_DAP_MAIN_CHAN 0x0120 +#define SGTL5000_DAP_MIX_CHAN 0x0122 +#define SGTL5000_DAP_AVC_CTRL 0x0124 +#define SGTL5000_DAP_AVC_THRESHOLD 0x0126 +#define SGTL5000_DAP_AVC_ATTACK 0x0128 +#define SGTL5000_DAP_AVC_DECAY 0x012a +#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c +#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e +#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130 +#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132 +#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134 +#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136 +#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138 +#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a + +/* + * Field Definitions. + */ + +/* + * SGTL5000_CHIP_ID + */ +#define SGTL5000_PARTID_MASK 0xff00 +#define SGTL5000_PARTID_SHIFT 8 +#define SGTL5000_PARTID_WIDTH 8 +#define SGTL5000_PARTID_PART_ID 0xa0 +#define SGTL5000_REVID_MASK 0x00ff +#define SGTL5000_REVID_SHIFT 0 +#define SGTL5000_REVID_WIDTH 8 + +/* + * SGTL5000_CHIP_DIG_POWER + */ +#define SGTL5000_ADC_EN 0x0040 +#define SGTL5000_DAC_EN 0x0020 +#define SGTL5000_DAP_POWERUP 0x0010 +#define SGTL5000_I2S_OUT_POWERUP 0x0002 +#define SGTL5000_I2S_IN_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_CLK_CTRL + */ +#define SGTL5000_RATE_MODE_MASK 0x0030 +#define SGTL5000_RATE_MODE_SHIFT 4 +#define SGTL5000_RATE_MODE_WIDTH 2 +#define SGTL5000_RATE_MODE_DIV_1 0 +#define SGTL5000_RATE_MODE_DIV_2 1 +#define SGTL5000_RATE_MODE_DIV_4 2 +#define SGTL5000_RATE_MODE_DIV_6 3 +#define SGTL5000_SYS_FS_MASK 0x000c +#define SGTL5000_SYS_FS_SHIFT 2 +#define SGTL5000_SYS_FS_WIDTH 2 +#define SGTL5000_SYS_FS_32k 0x0 +#define SGTL5000_SYS_FS_44_1k 0x1 +#define SGTL5000_SYS_FS_48k 0x2 +#define SGTL5000_SYS_FS_96k 0x3 +#define SGTL5000_MCLK_FREQ_MASK 0x0003 +#define SGTL5000_MCLK_FREQ_SHIFT 0 +#define SGTL5000_MCLK_FREQ_WIDTH 2 +#define SGTL5000_MCLK_FREQ_256FS 0x0 +#define SGTL5000_MCLK_FREQ_384FS 0x1 +#define SGTL5000_MCLK_FREQ_512FS 0x2 +#define SGTL5000_MCLK_FREQ_PLL 0x3 + +/* + * SGTL5000_CHIP_I2S_CTRL + */ +#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100 +#define SGTL5000_I2S_SCLKFREQ_SHIFT 8 +#define SGTL5000_I2S_SCLKFREQ_WIDTH 1 +#define SGTL5000_I2S_SCLKFREQ_64FS 0x0 +#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */ +#define SGTL5000_I2S_MASTER 0x0080 +#define SGTL5000_I2S_SCLK_INV 0x0040 +#define SGTL5000_I2S_DLEN_MASK 0x0030 +#define SGTL5000_I2S_DLEN_SHIFT 4 +#define SGTL5000_I2S_DLEN_WIDTH 2 +#define SGTL5000_I2S_DLEN_32 0x0 +#define SGTL5000_I2S_DLEN_24 0x1 +#define SGTL5000_I2S_DLEN_20 0x2 +#define SGTL5000_I2S_DLEN_16 0x3 +#define SGTL5000_I2S_MODE_MASK 0x000c +#define SGTL5000_I2S_MODE_SHIFT 2 +#define SGTL5000_I2S_MODE_WIDTH 2 +#define SGTL5000_I2S_MODE_I2S_LJ 0x0 +#define SGTL5000_I2S_MODE_RJ 0x1 +#define SGTL5000_I2S_MODE_PCM 0x2 +#define SGTL5000_I2S_LRALIGN 0x0002 +#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */ + +/* + * SGTL5000_CHIP_SSS_CTRL + */ +#define SGTL5000_DAP_MIX_LRSWAP 0x4000 +#define SGTL5000_DAP_LRSWAP 0x2000 +#define SGTL5000_DAC_LRSWAP 0x1000 +#define SGTL5000_I2S_OUT_LRSWAP 0x0400 +#define SGTL5000_DAP_MIX_SEL_MASK 0x0300 +#define SGTL5000_DAP_MIX_SEL_SHIFT 8 +#define SGTL5000_DAP_MIX_SEL_WIDTH 2 +#define SGTL5000_DAP_MIX_SEL_ADC 0x0 +#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1 +#define SGTL5000_DAP_SEL_MASK 0x00c0 +#define SGTL5000_DAP_SEL_SHIFT 6 +#define SGTL5000_DAP_SEL_WIDTH 2 +#define SGTL5000_DAP_SEL_ADC 0x0 +#define SGTL5000_DAP_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_MASK 0x0030 +#define SGTL5000_DAC_SEL_SHIFT 4 +#define SGTL5000_DAC_SEL_WIDTH 2 +#define SGTL5000_DAC_SEL_ADC 0x0 +#define SGTL5000_DAC_SEL_I2S_IN 0x1 +#define SGTL5000_DAC_SEL_DAP 0x3 +#define SGTL5000_I2S_OUT_SEL_MASK 0x0003 +#define SGTL5000_I2S_OUT_SEL_SHIFT 0 +#define SGTL5000_I2S_OUT_SEL_WIDTH 2 +#define SGTL5000_I2S_OUT_SEL_ADC 0x0 +#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1 +#define SGTL5000_I2S_OUT_SEL_DAP 0x3 + +/* + * SGTL5000_CHIP_ADCDAC_CTRL + */ +#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000 +#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000 +#define SGTL5000_DAC_VOL_RAMP_EN 0x0200 +#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100 +#define SGTL5000_DAC_MUTE_RIGHT 0x0008 +#define SGTL5000_DAC_MUTE_LEFT 0x0004 +#define SGTL5000_ADC_HPF_FREEZE 0x0002 +#define SGTL5000_ADC_HPF_BYPASS 0x0001 + +/* + * SGTL5000_CHIP_DAC_VOL + */ +#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00 +#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8 +#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8 +#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff +#define SGTL5000_DAC_VOL_LEFT_SHIFT 0 +#define SGTL5000_DAC_VOL_LEFT_WIDTH 8 + +/* + * SGTL5000_CHIP_PAD_STRENGTH + */ +#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300 +#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8 +#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0 +#define SGTL5000_PAD_I2S_SCLK_SHIFT 6 +#define SGTL5000_PAD_I2S_SCLK_WIDTH 2 +#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030 +#define SGTL5000_PAD_I2S_DOUT_SHIFT 4 +#define SGTL5000_PAD_I2S_DOUT_WIDTH 2 +#define SGTL5000_PAD_I2C_SDA_MASK 0x000c +#define SGTL5000_PAD_I2C_SDA_SHIFT 2 +#define SGTL5000_PAD_I2C_SDA_WIDTH 2 +#define SGTL5000_PAD_I2C_SCL_MASK 0x0003 +#define SGTL5000_PAD_I2C_SCL_SHIFT 0 +#define SGTL5000_PAD_I2C_SCL_WIDTH 2 + +/* + * SGTL5000_CHIP_ANA_ADC_CTRL + */ +#define SGTL5000_ADC_VOL_M6DB 0x0100 +#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0 +#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4 +#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4 +#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f +#define SGTL5000_ADC_VOL_LEFT_SHIFT 0 +#define SGTL5000_ADC_VOL_LEFT_WIDTH 4 + +/* + * SGTL5000_CHIP_ANA_HP_CTRL + */ +#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00 +#define SGTL5000_HP_VOL_RIGHT_SHIFT 8 +#define SGTL5000_HP_VOL_RIGHT_WIDTH 7 +#define SGTL5000_HP_VOL_LEFT_MASK 0x007f +#define SGTL5000_HP_VOL_LEFT_SHIFT 0 +#define SGTL5000_HP_VOL_LEFT_WIDTH 7 + +/* + * SGTL5000_CHIP_ANA_CTRL + */ +#define SGTL5000_LINE_OUT_MUTE 0x0100 +#define SGTL5000_HP_SEL_MASK 0x0040 +#define SGTL5000_HP_SEL_SHIFT 6 +#define SGTL5000_HP_SEL_WIDTH 1 +#define SGTL5000_HP_SEL_DAC 0x0 +#define SGTL5000_HP_SEL_LINE_IN 0x1 +#define SGTL5000_HP_ZCD_EN 0x0020 +#define SGTL5000_HP_MUTE 0x0010 +#define SGTL5000_ADC_SEL_MASK 0x0004 +#define SGTL5000_ADC_SEL_SHIFT 2 +#define SGTL5000_ADC_SEL_WIDTH 1 +#define SGTL5000_ADC_SEL_MIC 0x0 +#define SGTL5000_ADC_SEL_LINE_IN 0x1 +#define SGTL5000_ADC_ZCD_EN 0x0002 +#define SGTL5000_ADC_MUTE 0x0001 + +/* + * SGTL5000_CHIP_LINREG_CTRL + */ +#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040 +#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6 +#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1 +#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0 +#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1 +#define SGTL5000_VDDC_ASSN_OVRD 0x0020 +#define SGTL5000_LINREG_VDDD_MASK 0x000f +#define SGTL5000_LINREG_VDDD_SHIFT 0 +#define SGTL5000_LINREG_VDDD_WIDTH 4 + +/* + * SGTL5000_CHIP_REF_CTRL + */ +#define SGTL5000_ANA_GND_MASK 0x01f0 +#define SGTL5000_ANA_GND_SHIFT 4 +#define SGTL5000_ANA_GND_WIDTH 5 +#define SGTL5000_ANA_GND_BASE 800 /* mv */ +#define SGTL5000_ANA_GND_STP 25 /*mv */ +#define SGTL5000_BIAS_CTRL_MASK 0x000e +#define SGTL5000_BIAS_CTRL_SHIFT 1 +#define SGTL5000_BIAS_CTRL_WIDTH 3 +#define SGTL5000_SMALL_POP 0x0001 + +/* + * SGTL5000_CHIP_MIC_CTRL + */ +#define SGTL5000_BIAS_R_MASK 0x0200 +#define SGTL5000_BIAS_R_SHIFT 8 +#define SGTL5000_BIAS_R_WIDTH 2 +#define SGTL5000_BIAS_R_off 0x0 +#define SGTL5000_BIAS_R_2K 0x1 +#define SGTL5000_BIAS_R_4k 0x2 +#define SGTL5000_BIAS_R_8k 0x3 +#define SGTL5000_BIAS_VOLT_MASK 0x0070 +#define SGTL5000_BIAS_VOLT_SHIFT 4 +#define SGTL5000_BIAS_VOLT_WIDTH 3 +#define SGTL5000_MIC_GAIN_MASK 0x0003 +#define SGTL5000_MIC_GAIN_SHIFT 0 +#define SGTL5000_MIC_GAIN_WIDTH 2 + +/* + * SGTL5000_CHIP_LINE_OUT_CTRL + */ +#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00 +#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8 +#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4 +#define SGTL5000_LINE_OUT_CURRENT_180u 0x0 +#define SGTL5000_LINE_OUT_CURRENT_270u 0x1 +#define SGTL5000_LINE_OUT_CURRENT_360u 0x3 +#define SGTL5000_LINE_OUT_CURRENT_450u 0x7 +#define SGTL5000_LINE_OUT_CURRENT_540u 0xf +#define SGTL5000_LINE_OUT_GND_MASK 0x003f +#define SGTL5000_LINE_OUT_GND_SHIFT 0 +#define SGTL5000_LINE_OUT_GND_WIDTH 6 +#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */ +#define SGTL5000_LINE_OUT_GND_STP 25 +#define SGTL5000_LINE_OUT_GND_MAX 0x23 + +/* + * SGTL5000_CHIP_LINE_OUT_VOL + */ +#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00 +#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8 +#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5 +#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f +#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0 +#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5 + +/* + * SGTL5000_CHIP_ANA_POWER + */ +#define SGTL5000_DAC_STEREO 0x4000 +#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000 +#define SGTL5000_STARTUP_POWERUP 0x1000 +#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800 +#define SGTL5000_PLL_POWERUP 0x0400 +#define SGTL5000_LINEREG_D_POWERUP 0x0200 +#define SGTL5000_VCOAMP_POWERUP 0x0100 +#define SGTL5000_VAG_POWERUP 0x0080 +#define SGTL5000_ADC_STEREO 0x0040 +#define SGTL5000_REFTOP_POWERUP 0x0020 +#define SGTL5000_HP_POWERUP 0x0010 +#define SGTL5000_DAC_POWERUP 0x0008 +#define SGTL5000_CAPLESS_HP_POWERUP 0x0004 +#define SGTL5000_ADC_POWERUP 0x0002 +#define SGTL5000_LINE_OUT_POWERUP 0x0001 + +/* + * SGTL5000_CHIP_PLL_CTRL + */ +#define SGTL5000_PLL_INT_DIV_MASK 0xf800 +#define SGTL5000_PLL_INT_DIV_SHIFT 11 +#define SGTL5000_PLL_INT_DIV_WIDTH 5 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_SHIFT 0 +#define SGTL5000_PLL_FRAC_DIV_WIDTH 11 + +/* + * SGTL5000_CHIP_CLK_TOP_CTRL + */ +#define SGTL5000_INT_OSC_EN 0x0800 +#define SGTL5000_INPUT_FREQ_DIV2 0x0008 + +/* + * SGTL5000_CHIP_ANA_STATUS + */ +#define SGTL5000_HP_LRSHORT 0x0200 +#define SGTL5000_CAPLESS_SHORT 0x0100 +#define SGTL5000_PLL_LOCKED 0x0010 + +/* + * SGTL5000_CHIP_SHORT_CTRL + */ +#define SGTL5000_LVLADJR_MASK 0x7000 +#define SGTL5000_LVLADJR_SHIFT 12 +#define SGTL5000_LVLADJR_WIDTH 3 +#define SGTL5000_LVLADJL_MASK 0x0700 +#define SGTL5000_LVLADJL_SHIFT 8 +#define SGTL5000_LVLADJL_WIDTH 3 +#define SGTL5000_LVLADJC_MASK 0x0070 +#define SGTL5000_LVLADJC_SHIFT 4 +#define SGTL5000_LVLADJC_WIDTH 3 +#define SGTL5000_LR_SHORT_MOD_MASK 0x000c +#define SGTL5000_LR_SHORT_MOD_SHIFT 2 +#define SGTL5000_LR_SHORT_MOD_WIDTH 2 +#define SGTL5000_CM_SHORT_MOD_MASK 0x0003 +#define SGTL5000_CM_SHORT_MOD_SHIFT 0 +#define SGTL5000_CM_SHORT_MOD_WIDTH 2 + +/* + *SGTL5000_CHIP_ANA_TEST2 + */ +#define SGTL5000_MONO_DAC 0x1000 + +/* + * SGTL5000_DAP_CTRL + */ +#define SGTL5000_DAP_MIX_EN 0x0010 +#define SGTL5000_DAP_EN 0x0001 + +#define SGTL5000_SYSCLK 0x00 +#define SGTL5000_LRCLK 0x01 + +#endif -- 1.7.0.4

3 2

[alsa-devel] ASoC updates for 2.6.38
by Mark Brown 25 Feb '11

25 Feb '11

The following changes since commit 3017358a75917b5ed5ad361c02ba2a7e257d3b2a: ASoC: Ensure supplies are maintained for force enabled widgets (2011-02-13 19:49:47 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.38 More of the usual small fixes. Dmitry Eremin-Solenikov (1): ASoC: correct pxa AC97 DAI names Mark Brown (3): ASoC: Fix WM8958 default microphone detection argument ordering ASoC: Correct definition of WM8903_VMID_RES_5K ASoC: Hook wm_hubs micbiases up to CLK_SYS sound/soc/codecs/wm8903.h | 2 +- sound/soc/codecs/wm8994.c | 5 ++--- sound/soc/codecs/wm_hubs.c | 3 +++ sound/soc/pxa/e740_wm9705.c | 4 ++-- sound/soc/pxa/e750_wm9705.c | 4 ++-- sound/soc/pxa/e800_wm9712.c | 4 ++-- sound/soc/pxa/em-x270.c | 4 ++-- sound/soc/pxa/mioa701_wm9713.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/pxa/zylonite.c | 4 ++-- 11 files changed, 22 insertions(+), 20 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] CHROMIUM: ASoC: Tegra: Move utilities to separate module
by Stephen Warren 25 Feb '11

25 Feb '11

The utilities will be required by every machine driver. Including the utility object directly into every machine driver causes a build failure if the modules are actually built into the kernel, since each will define the symbols exported by the utility file. Solve this by moving the utility object into a separate module. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/Makefile | 3 ++- sound/soc/tegra/tegra_asoc_utils.c | 6 ++++++ 2 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index dfd2ab9..fd183d3 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -2,13 +2,14 @@ snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-i2s-objs := tegra_i2s.o +snd-soc-tegra-utils-objs += tegra_asoc_utils.o +obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-utils.o obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-das.o obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o # Tegra machine Support snd-soc-tegra-harmony-objs := harmony.o -snd-soc-tegra-harmony-objs += tegra_asoc_utils.o obj-$(CONFIG_SND_TEGRA_SOC_HARMONY) += snd-soc-tegra-harmony.o diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index cb4fc13..52f0a3f 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -101,6 +101,7 @@ int tegra_asoc_utils_set_rate(struct tegra_asoc_utils_data *data, int srate, return 0; } +EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_rate); int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, struct device *dev) @@ -139,6 +140,7 @@ err_put_pll_a: err: return ret; } +EXPORT_SYMBOL_GPL(tegra_asoc_utils_init); void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data) { @@ -146,4 +148,8 @@ void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data) clk_put(data->clk_pll_a_out0); clk_put(data->clk_pll_a); } +EXPORT_SYMBOL_GPL(tegra_asoc_utils_fini); +MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); +MODULE_DESCRIPTION("Tegra ASoC utility code"); +MODULE_LICENSE("GPL"); -- 1.7.1

3 4

[alsa-devel] [PATCH] ASoC: soc-dapm: Include quotes around contents in debugfs entries
by Dimitris Papastamos 25 Feb '11

25 Feb '11

Sometimes the name of the control switch of a dapm route contains spaces which makes it impossible to distinguish it from the source widget. Add quotes around the names of the widgets to makes these parsable. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 200ae7c..b8e6ab7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1242,7 +1242,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, - " in %s %s\n", + " in \"%s\" \"%s\"\n", p->name ? p->name : "static", p->source->name); } @@ -1252,7 +1252,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, - " out %s %s\n", + " out \"%s\" \"%s\"\n", p->name ? p->name : "static", p->sink->name); } -- 1.7.4.1

3 2

[alsa-devel] [PATCH 20/21]sound:pci:hda:patch_realtek.c remove one to many l's in the word.
by Justin P. Mattock 25 Feb '11

25 Feb '11

The patch below removes an extra "l" in the word. Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3328a25..c5208c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4705,7 +4705,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2), SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG), SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG), - SND_PCI_QUIRK(0x1584, 0x9054, "Uniwlll", ALC880_F1734), + SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734), SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL), SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53), SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810), -- 1.7.4.1

3 2

[alsa-devel] [PATCH 19/21]sound:pci:emu10k1:emu10k1_main.c remove one to many l's in the word.
by Justin P. Mattock 25 Feb '11

25 Feb '11

The patch below removes an extra "l" in the word. Signed-off-by: Justin P. Mattock <justinmattock(a)gmail.com> --- sound/pci/emu10k1/emu10k1_main.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 66c7fb3..5e619a8 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -926,7 +926,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_IN, 0x19); /* Unknown. */ snd_emu1010_fpga_write(emu, EMU_HANA_MIDI_OUT, 0x0c); - /* IRQ Enable: Alll on */ + /* IRQ Enable: All on */ /* snd_emu1010_fpga_write(emu, 0x09, 0x0f ); */ /* IRQ Enable: All off */ snd_emu1010_fpga_write(emu, EMU_HANA_IRQ_ENABLE, 0x00); -- 1.7.4.1

1 0

[alsa-devel] [PATCH 1/3] pxa: Enable pxa-pcm-audio on pxa210/pxa25x platform
by Dmitry Eremin-Solenikov 24 Feb '11

24 Feb '11

pxa25x platforms were left out of major ASoC Update patch. Since f0fba2ad1b a registration of pxa-pcm-audio device is required for ASoC to function on pxa platforms. Register one also for pxa210/pxa25x. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov(a)gmail.com> Cc: Eric Miao <eric.y.miao(a)gmail.com> --- arch/arm/mach-pxa/pxa25x.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/arch/arm/mach-pxa/pxa25x.c b/arch/arm/mach-pxa/pxa25x.c index fbc5b77..b166b1d 100644 --- a/arch/arm/mach-pxa/pxa25x.c +++ b/arch/arm/mach-pxa/pxa25x.c @@ -347,6 +347,7 @@ static struct platform_device *pxa25x_devices[] __initdata = { &pxa25x_device_assp, &pxa25x_device_pwm0, &pxa25x_device_pwm1, + &pxa_device_asoc_platform, }; static struct sys_device pxa25x_sysdev[] = { -- 1.7.2.3

3 4

[alsa-devel] : Question ASoC driver without Codec driver?
by Bensaid, Selma 24 Feb '11

24 Feb '11

Hi, I'm trying to find out if it's possible to implement an ASoC driver without a codec driver. I'm implementing a ASoC driver for BT/FM WL1273, where the FM WL1273 control interface is based on V4L2 since since we use the Combined Interface Mode (The host controls both the BT and the FM Radio through the BT HCI host interface). Thanks for your help. Selma. --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

4 3

[alsa-devel] ALSA: hda - 4930g add internal lfe slider
by Łukasz Wojniłowicz 24 Feb '11

24 Feb '11

Hello, Lately I sent patch that switched lfe with side in mixer for acer-aspire-4930g. Then I connected 5.1 speaker system and noticed that lfe slider wasn't working and that old lfe slider worked. What I'm doing now is: - reverting old patch - adding internal lfe slider - removing side as it is superfluous (ALC888S-VC is 7.1 but in fact laptop can only do 5.1 and it is so in drivers for MS Windows) Cheers, Łukasz

2 1

[alsa-devel] [PATCH] ASoC: Pass the jack to jack notifiers
by Mark Brown 23 Feb '11

23 Feb '11

We're currently not passing anything and this will make the card and so on more discoverable. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-jack.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 1382251..fcab80b 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -101,7 +101,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, NULL); + blocking_notifier_call_chain(&jack->notifier, status, jack); snd_soc_dapm_sync(dapm); -- 1.7.2.3

2 1