May 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: add support for Akai MPD16
by Krzysztof Foltman 20 May '10

20 May '10

The decoding/encoding is based on own reverse-engineering. Both control and data ports are handled. Writing to control port supports SysEx events only, as this is the only type of messages that MPD16 recognizes. Signed-off-by: Krzysztof Foltman <wdev(a)foltman.com> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1558c..9c23d89 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -645,6 +645,105 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: + * + * For control port (endpoint 1): + * ============================== + * One or more chunks consisting of first byte of (0x10 | msg_len) and then a + * SysEx message (msg_len=9 bytes long). + * + * For data port (endpoint 2): + * =========================== + * One or more chunks consisting of first byte of (0x20 | msg_len) and then a + * MIDI message (msg_len bytes long) + * + * Messages sent: Active Sense, Note On, Poly Pressure, Control Change. + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + unsigned int len = (unsigned int)buffer_length; + while (pos < len) { + unsigned int port = (buffer[pos] >> 4) - 1; + unsigned int msg_len = buffer[pos] & 0x0f; + pos++; + if (pos + msg_len <= len && port < 2) + snd_usbmidi_input_data(ep, 0, &buffer[pos], msg_len); + pos += msg_len; + } +} + +#define MAX_AKAI_SYSEX_LEN 9 + +static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep, + struct urb *urb) +{ + uint8_t *msg; + int pos, end, count, buf_end; + uint8_t tmp[MAX_AKAI_SYSEX_LEN]; + struct snd_rawmidi_substream *substream = ep->ports[0].substream; + + if (!ep->ports[0].active) + return; + + msg = urb->transfer_buffer + urb->transfer_buffer_length; + buf_end = ep->max_transfer - MAX_AKAI_SYSEX_LEN - 1; + + /* only try adding more data when there's space for at least 1 SysEx */ + while (urb->transfer_buffer_length < buf_end) { + count = snd_rawmidi_transmit_peek(substream, + tmp, MAX_AKAI_SYSEX_LEN); + if (!count) { + ep->ports[0].active = 0; + return; + } + /* try to skip non-SysEx data */ + for (pos = 0; pos < count && tmp[pos] != 0xF0; pos++) + ; + + if (pos > 0) { + snd_rawmidi_transmit_ack(substream, pos); + continue; + } + + /* look for the start or end marker */ + for (end = 1; end < count && tmp[end] < 0xF0; end++) + ; + + /* next SysEx started before the end of current one */ + if (end < count && tmp[end] == 0xF0) { + /* it's incomplete - drop it */ + snd_rawmidi_transmit_ack(substream, end); + continue; + } + /* SysEx complete */ + if (end < count && tmp[end] == 0xF7) { + /* queue it, ack it, and get the next one */ + count = end + 1; + msg[0] = 0x10 | count; + memcpy(&msg[1], tmp, count); + snd_rawmidi_transmit_ack(substream, count); + urb->transfer_buffer_length += count + 1; + msg += count + 1; + continue; + } + /* less than 9 bytes and no end byte - wait for more */ + if (count < MAX_AKAI_SYSEX_LEN) { + ep->ports[0].active = 0; + return; + } + /* 9 bytes and no end marker in sight - malformed, skip it */ + snd_rawmidi_transmit_ack(substream, count); + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_akai_output, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1533,11 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + CONTROL_PORT(0x09e8, 0x0062, 0, "%s Control"), + PORT_INFO(0x09e8, 0x0062, 1, "%s MIDI", 0, + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | + SNDRV_SEQ_PORT_TYPE_HARDWARE), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -2035,6 +2139,12 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 1 is input-only */ + endpoints[1].out_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec9..2fca80b 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef3..f8797f6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4..b45e54c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72..06ebf24 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
by Jarkko Nikula 20 May '10

20 May '10

Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise external widgets doesn't alter the output state. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- I noticed this with a widget and audio map below where state of "Headphone Jack" wasn't changing the codec output. SND_SOC_DAPM_HP("Headphone Jack", NULL), {"Headphone Jack", NULL, "TPA6130A2 Headphone Left"}, {"Headphone Jack", NULL, "TPA6130A2 Headphone Right"}, {"TPA6130A2 Left", NULL, "LLOUT"}, {"TPA6130A2 Right", NULL, "RLOUT"} --- sound/soc/codecs/tpa6130a2.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 100a747..99b70e5 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -359,8 +359,8 @@ static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { 0, 0, tpa6130a2_supply_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Outputs */ - SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL), - SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL), + SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Left"), + SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Right"), }; static const struct snd_soc_dapm_route audio_map[] = { -- 1.7.1

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[alsa-devel] [RFC] disabling ALSA period interrupts
by pl bossart 20 May '10

20 May '10

Howdy, When PulseAudio is used and all PCM is routed through PulseAudio (Fedora, Meego, etc), the notion of ALSA periods isn't very useful. PulseAudio uses a timer to refill buffers and the period interrupts are not used at all. So why not disable them entirely to reduce the number of wakeups? This is possible with a number of existing DMA controllers. The simple patch attached shows a proof-of-concept on HDAudio, it's been working for 5 hours on my Fedora12 laptop without any glitches and powertop does show _zero_ wakeups from the HDAudio controller (except on startup). I am told by my colleagues working in Windows environments that this is what's done on Vista/Windows7 btw. This isn't to say Windows is great but that artificial generation of not-needed interrupts is avoidable. There are probably some cases where you don't want this type of behavior (broken hardware, legacy code with multiple-buffering, disabled timer in PulseAudio), so I think it would make sense to request the disabling of interrupts when hw_params are set, since this is also the time when period sizes are set. I am aware that some changes would be needed in pcm_lib.c, where all the error checks are done. Takashi, Jaroslav, Lennart, what do you think? Feedback and suggestions welcome. Cheers - Pierre

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[alsa-devel] [GIT PULL] ASoC update for 2.6.35
by Liam Girdwood 20 May '10

20 May '10

Hi Takashi, Hope this is not too late. Thanks Liam --- The following changes since commit 266d38c8e3d7f62152b1448fd9a7265f32f32d87: Jarkko Nikula (1): ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (1): ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies sound/soc/codecs/tpa6130a2.c | 52 ++++++++++++++--------------------------- 1 files changed, 18 insertions(+), 34 deletions(-)

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[alsa-devel] [PATCH 11/11] ALSA: hda: Storage class should be before const qualifier
by Tobias Klauser 20 May '10

20 May '10

The C99 specification states in section 6.11.5: The placement of a storage-class specifier other than at the beginning of the declaration specifiers in a declaration is an obsolescent feature. Signed-off-by: Tobias Klauser <tklauser(a)distanz.ch> --- 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 80a183f..53538b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14207,7 +14207,7 @@ enum { ALC269_FIXUP_SONY_VAIO, }; -const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = { +static const struct hda_verb alc269_sony_vaio_fixup_verbs[] = { {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, {} }; -- 1.6.3.3

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[alsa-devel] [PATCHv2] ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
by Jarkko Nikula 20 May '10

20 May '10

These pins are for decoupling capacitors for the internal charge pumps in TPA6130A2 and TPA6140A2 and not for connecting external supply. Thanks to Eduardo Valentin <eduardo.valentin(a)nokia.com> for pointing out the issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen(a)nokia.com> with TPA6140A2. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> Reviewed-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- v2: - I took freedom to add ack and reviewed by lines as Peter and Ilkka gave those to v1 - HPVdd for the TPA6140A2 is also removed as it's for decoupling capacitor as well - No need to use regulator_bulk API as there s only single supply for both supported chips now --- sound/soc/codecs/tpa6130a2.c | 50 ++++++++++++++--------------------------- 1 files changed, 17 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 221bb90..f9b3d6d 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -36,22 +36,11 @@ static struct i2c_client *tpa6130a2_client; -#define TPA6130A2_NUM_SUPPLIES 2 -static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "CPVSS", - "Vdd", -}; - -static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "HPVdd", - "AVdd", -}; - /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; unsigned char regs[TPA6130A2_CACHEREGNUM]; - struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; + struct regulator *supply; int power_gpio; unsigned char power_state; enum tpa_model id; @@ -136,11 +125,10 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_enable(data->supply); if (ret != 0) { dev_err(&tpa6130a2_client->dev, - "Failed to enable supplies: %d\n", ret); + "Failed to enable supply: %d\n", ret); goto exit; } @@ -161,11 +149,10 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_disable(data->supply); if (ret != 0) { dev_err(&tpa6130a2_client->dev, - "Failed to disable supplies: %d\n", ret); + "Failed to disable supply: %d\n", ret); goto exit; } @@ -412,6 +399,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; + const char *regulator; int i, ret; dev = &client->dev; @@ -454,25 +442,21 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, } switch (data->id) { + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); case TPA6130A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + regulator = "Vdd"; break; case TPA6140A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6140a2_supply_names[i];; + regulator = "AVdd"; break; - default: - dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", - pdata->id); - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; } - ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), - data->supplies); - if (ret != 0) { - dev_err(dev, "Failed to request supplies: %d\n", ret); + data->supply = regulator_get(dev, regulator); + if (IS_ERR(data->supply)) { + ret = PTR_ERR(data->supply); + dev_err(dev, "Failed to request supply: %d\n", ret); goto err_regulator; } @@ -495,7 +479,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, return 0; err_power: - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_put(data->supply); err_regulator: if (data->power_gpio >= 0) gpio_free(data->power_gpio); @@ -516,7 +500,7 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_put(data->supply); kfree(data); tpa6130a2_client = NULL; -- 1.7.1

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[alsa-devel] First 50 milliseconds of audio clipping with dmix
by Tyler Conant 20 May '10

20 May '10

I'm trying to use dmix on an OMAP processor. I'm experiencing the first 50 ms of the sample getting clipped. When I aplay directly to the hw:0 I don't get the clipping. The difference I see between the alsa configuration between dmix and hw:0 is the start_threshold, stop_threshold and silence_size. When it's direct to hw:0 the start and stop threshold are equal to the buffer size. For dmix the start threshold is 1 and the stop threshold is 1445199872. Is this dmix which is setting the start and stop or is this aplay. I'm only guessing this is the reason why the first part is clipping. The only bug which seem similar is http://bugs.gentoo.org/show_bug.cgi?id=218641 and it's already been closed out. I'm using alsa lib 1.0.23 with hw:0 Audio_Prompts # aplay -v -D hw:0 Click03.wav Playing WAVE 'Click03.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 22052 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 appl_ptr : 0 hw_ptr : 0 /opt/axonapp/Audio_Prompts # With dmix: /Audio_Prompts # aplay -v Click03.wav Playing WAVE 'Click03.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 22052 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tstamp_mode : ENABLE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 1 stop_threshold : 1445199872 silence_threshold: 0 silence_size : 1445199872 boundary : 1445199872 appl_ptr : 0 hw_ptr : 0 /opt/axonapp/Audio_Prompts # asound.conf: pcm.dmixer { type dmix ipc_key 2048 ipc_key_add_uid true slave { channels 1 pcm hw:0 rate 44100 period_time 125011 period_size 5513 buffer_size 22052 } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "hw:0,0" } pcm.!default { type plug slave.pcm "asymed" }

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[alsa-devel] [PATCH 1/1] ALSA: usb-audio: add support for Akai MPD16
by Krzysztof Foltman 19 May '10

19 May '10

The decoding routine is based on own reverse-engineering. It seems to recognize all the messages that MPD16 sends during normal operation (that is, via standard MIDI messages). Configuration (changing pad sensitivity, slider controller and MIDI notes) is not supported in this version. Signed-off-by: Krzysztof Foltman <wdev(a)foltman.com> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1558c..cce5dbe 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -645,6 +645,33 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: one or more chunks consisting of first byte of + * (0x20 | msg_len) and then a MIDI message (msg_len bytes long) + * + * Messages sent: + * 21 FE (active sense) + * 23 90 xx xx (note on) + * 23 Ax xx xx (polyphonic pressure) + * 23 Bx xx xx (control change) + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + while (pos < (unsigned)buffer_length && (buffer[pos] & 0xF8) == 0x20) { + int msg_len = buffer[pos] & 0x0f; + snd_usbmidi_input_data(ep, 0, &buffer[pos + 1], msg_len); + pos += 1 + msg_len; + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_standard_output, + .output_packet = snd_usbmidi_output_standard_packet, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1461,8 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + EXTERNAL_PORT(0x09e8, 0x0062, 0, "%s MIDI"), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -2035,6 +2064,13 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 0 does not carry MIDI data */ + endpoints[0].out_cables = 0; + endpoints[0].in_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec9..2fca80b 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef3..7ecad05 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4..b45e54c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72..06ebf24 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, -- 1.7.0.4

2 2

[alsa-devel] [GIT PULL] ASoC updates for 2.6.35
by Liam Girdwood 19 May '10

19 May '10

Hi Takashi, The last one for 2.6.35 :) Thanks Liam ---- The following changes since commit 2d4cdd6fc974716555fcbaf7ec1d4dda22784c1b: Peter Ujfalusi (1): ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Jarkko Nikula (1): ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT Jorge Eduardo Candelaria (2): ASoC: SDP4430: Add support for Earphone speaker ASoC: TWL6040: Enable earphone path in codec Liam Girdwood (1): ASoC: sdp4430 - add sdp4430 pcm ops to DAI. Misael Lopez Cruz (1): ASoC: SDP4430: Add sdp4430 machine driver sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/twl6040.c | 21 ++++- sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp4430.c | 233 ++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 266 insertions(+), 3 deletions(-) create mode 100644 sound/soc/omap/sdp4430.c

2 1

[alsa-devel] [PATCH 0/4] McBSP smart idle and DMA op mode updates for ASoC
by Liam Girdwood 19 May '10

19 May '10

This series expands the OMAP mcbsp driver to support changing it's DMA operating mode and smart idle mode from client drivers. It's primarily aimed at lowering the power consumption for OMAP ASoC drivers by providing methods to gate clocks on the mcbsp interface at runtime. I've also added a patch to remove the mcbsp DMA op mode sysfs set functionality. I think DMA op mode is very specific to the mcbsp client driver _only_ and shouldn't really be changed by userspace. Please let me know if you use this feature and I'll drop this patch. Tony, can I get your ack to upstream via ALSA if it's ok Thanks Liam arch/arm/plat-omap/include/plat/mcbsp.h | 17 +++ arch/arm/plat-omap/mcbsp.c | 183 ++++++++++++++++++++++++++----- sound/soc/omap/omap-mcbsp.c | 22 ++++- sound/soc/omap/omap-mcbsp.h | 2 + 4 files changed, 194 insertions(+), 30 deletions(-)

7 31

Re: [alsa-devel] [PATCH v2] NUC900/audio: add nuc900 audio driver support
by Wan ZongShun 19 May '10

19 May '10

Hi Mark, How about this patch? Could it be merged into mainline? Thanks! 在 2010年5月18日下午1:41，Wan ZongShun <mcuos.com(a)gmail.com> 写道： > Dear Mark, > > I have modified some codes according to your comments, > now I submit the patch as v2 again. > > Thanks a lot for your help. > > Signed-off-by: Wan ZongShun<mcuos.com(a)gmail.com> > --- > sound/soc/Kconfig | 1 + > sound/soc/Makefile | 1 + > sound/soc/nuc900/Kconfig | 27 +++ > sound/soc/nuc900/Makefile | 11 + > sound/soc/nuc900/nuc900-ac97.c | 443 +++++++++++++++++++++++++++++++++++++++ > sound/soc/nuc900/nuc900-audio.c | 81 +++++++ > sound/soc/nuc900/nuc900-auido.h | 121 +++++++++++ > sound/soc/nuc900/nuc900-pcm.c | 352 +++++++++++++++++++++++++++++++ > 8 files changed, 1037 insertions(+), 0 deletions(-) > create mode 100644 sound/soc/nuc900/Kconfig > create mode 100644 sound/soc/nuc900/Makefile > create mode 100644 sound/soc/nuc900/nuc900-ac97.c > create mode 100644 sound/soc/nuc900/nuc900-audio.c > create mode 100644 sound/soc/nuc900/nuc900-auido.h > create mode 100644 sound/soc/nuc900/nuc900-pcm.c > > diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig > index b1749bc..6e04fc2 100644 > --- a/sound/soc/Kconfig > +++ b/sound/soc/Kconfig > @@ -30,6 +30,7 @@ source "sound/soc/blackfin/Kconfig" > source "sound/soc/davinci/Kconfig" > source "sound/soc/fsl/Kconfig" > source "sound/soc/imx/Kconfig" > +source "sound/soc/nuc900/Kconfig" > source "sound/soc/omap/Kconfig" > source "sound/soc/pxa/Kconfig" > source "sound/soc/s3c24xx/Kconfig" > diff --git a/sound/soc/Makefile b/sound/soc/Makefile > index 1470141..ccec241 100644 > --- a/sound/soc/Makefile > +++ b/sound/soc/Makefile > @@ -8,6 +8,7 @@ obj-$(CONFIG_SND_SOC) += blackfin/ > obj-$(CONFIG_SND_SOC) += davinci/ > obj-$(CONFIG_SND_SOC) += fsl/ > obj-$(CONFIG_SND_SOC) += imx/ > +obj-$(CONFIG_SND_SOC) += nuc900/ > obj-$(CONFIG_SND_SOC) += omap/ > obj-$(CONFIG_SND_SOC) += pxa/ > obj-$(CONFIG_SND_SOC) += s3c24xx/ > diff --git a/sound/soc/nuc900/Kconfig b/sound/soc/nuc900/Kconfig > new file mode 100644 > index 0000000..a0ed1c6 > --- /dev/null > +++ b/sound/soc/nuc900/Kconfig > @@ -0,0 +1,27 @@ > +## > +## NUC900 series AC97 API > +## > +config SND_SOC_NUC900 > + tristate "SoC Audio for NUC900 series" > + depends on ARCH_W90X900 > + help > + This option enables support for AC97 mode on the NUC900 SoC. > + > +config SND_SOC_NUC900_AC97 > + tristate > + select AC97_BUS > + select SND_AC97_CODEC > + select SND_SOC_AC97_BUS > + > + > +## > +## Boards > +## > +config SND_SOC_NUC900EVB > + tristate "NUC900 AC97 support for demo board" > + depends on SND_SOC_NUC900 > + select SND_SOC_NUC900_AC97 > + select SND_SOC_AC97_CODEC > + help > + Select this option to enable audio (AC97) on the > + NUC900 demoboard. > diff --git a/sound/soc/nuc900/Makefile b/sound/soc/nuc900/Makefile > new file mode 100644 > index 0000000..7e46c71 > --- /dev/null > +++ b/sound/soc/nuc900/Makefile > @@ -0,0 +1,11 @@ > +# NUC900 series audio > +snd-soc-nuc900-pcm-objs := nuc900-pcm.o > +snd-soc-nuc900-ac97-objs := nuc900-ac97.o > + > +obj-$(CONFIG_SND_SOC_NUC900) += snd-soc-nuc900-pcm.o > +obj-$(CONFIG_SND_SOC_NUC900_AC97) += snd-soc-nuc900-ac97.o > + > +# Boards > +snd-soc-nuc900-audio-objs := nuc900-audio.o > + > +obj-$(CONFIG_SND_SOC_NUC900EVB) += snd-soc-nuc900-audio.o > diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c > new file mode 100644 > index 0000000..2c345e2 > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-ac97.c > @@ -0,0 +1,443 @@ > +/* > + * Copyright (c) 2009-2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#include <linux/init.h> > +#include <linux/module.h> > +#include <linux/slab.h> > +#include <linux/device.h> > +#include <linux/delay.h> > +#include <linux/mutex.h> > +#include <linux/suspend.h> > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/initval.h> > +#include <sound/soc.h> > +#include <linux/device.h> > +#include <linux/clk.h> > + > +#include <mach/mfp.h> > + > +#include "nuc900-auido.h" > + > +static DEFINE_MUTEX(ac97_mutex); > +struct nuc900_audio *nuc900_ac97_data; > + > +static int nuc900_checkready(void) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + > + if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY)) > + return -EPERM; > + > + return 0; > +} > + > +/* AC97 controller reads codec register */ > +static unsigned short nuc900_ac97_read(struct snd_ac97 *ac97, > + unsigned short reg) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long timeout = 0x10000, val; > + > + mutex_lock(&ac97_mutex); > + > + val = nuc900_checkready(); > + if (!!val) { > + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); > + goto out; > + } > + > + /* set the R_WB bit and write register index */ > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, R_WB | reg); > + > + /* set the valid frame bit and valid slots */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + val |= (VALID_FRAME | SLOT1_VALID); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val); > + > + udelay(100); > + > + /* polling the AC_R_FINISH */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val &= AC_R_FINISH; > + while (!val && timeout--) > + mdelay(1); > + > + if (!timeout) { > + dev_err(nuc900_audio->dev, "AC97 read register time out !\n"); > + val = -EPERM; > + goto out; > + } > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ; > + val &= ~SLOT1_VALID; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, val); > + > + if (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS1) >> 2 != reg) { > + dev_err(nuc900_audio->dev, > + "R_INDEX of REG_ACTL_ACIS1 not match!\n"); > + } > + > + udelay(100); > + val = (AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS2) & 0xFFFF); > + > +out: > + mutex_unlock(&ac97_mutex); > + return val; > +} > + > +/* AC97 controller writes to codec register */ > +static void nuc900_ac97_write(struct snd_ac97 *ac97, unsigned short reg, > + unsigned short val) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long tmp, timeout = 0x10000; > + > + mutex_lock(&ac97_mutex); > + > + tmp = nuc900_checkready(); > + if (!!tmp) > + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); > + > + /* clear the R_WB bit and write register index */ > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS1, reg); > + > + /* write register value */ > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS2, val); > + > + /* set the valid frame bit and valid slots */ > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp |= SLOT1_VALID | SLOT2_VALID | VALID_FRAME; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + udelay(100); > + > + /* polling the AC_W_FINISH */ > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + tmp &= AC_W_FINISH; > + while (tmp && timeout--) > + mdelay(1); > + > + if (!timeout) > + dev_err(nuc900_audio->dev, "AC97 write register time out !\n"); > + > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp &= ~(SLOT1_VALID | SLOT2_VALID); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + mutex_unlock(&ac97_mutex); > + > +} > + > +static void nuc900_ac97_warm_reset(struct snd_ac97 *ac97) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long val; > + > + mutex_lock(&ac97_mutex); > + > + /* warm reset AC 97 */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val |= AC_W_RES; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + > + udelay(1000); > + > + val = nuc900_checkready(); > + if (!!val) > + dev_err(nuc900_audio->dev, "AC97 codec is not ready\n"); > + > + mutex_unlock(&ac97_mutex); > +} > + > +static void nuc900_ac97_cold_reset(struct snd_ac97 *ac97) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long val; > + > + mutex_lock(&ac97_mutex); > + > + /* reset Audio Controller */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val |= ACTL_RESET_BIT; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val &= (~ACTL_RESET_BIT); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + /* reset AC-link interface */ > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val |= AC_RESET; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + val &= ~AC_RESET; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + udelay(1000); > + > + /* cold reset AC 97 */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val |= AC_C_RES; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + > + udelay(1000); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val &= (~AC_C_RES); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + > + udelay(1000); > + > + mutex_unlock(&ac97_mutex); > + > +} > + > +/* AC97 controller operations */ > +struct snd_ac97_bus_ops soc_ac97_ops = { > + .read = nuc900_ac97_read, > + .write = nuc900_ac97_write, > + .reset = nuc900_ac97_cold_reset, > + .warm_reset = nuc900_ac97_warm_reset, > +} > +EXPORT_SYMBOL_GPL(soc_ac97_ops); > + > +static int nuc900_ac97_trigger(struct snd_pcm_substream *substream, > + int cmd, struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + int ret, stype = SUBSTREAM_TYPE(substream); > + unsigned long val, tmp; > + > + ret = 0; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + if (PCM_TX == stype) { > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp |= (SLOT3_VALID | SLOT4_VALID | VALID_FRAME); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR); > + tmp |= (P_DMA_END_IRQ | P_DMA_MIDDLE_IRQ); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, tmp); > + val |= AC_PLAY; > + } else { > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR); > + tmp |= (R_DMA_END_IRQ | R_DMA_MIDDLE_IRQ); > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, tmp); > + val |= AC_RECORD; > + } > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + if (PCM_TX == stype) { > + tmp = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0); > + tmp &= ~(SLOT3_VALID | SLOT4_VALID); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACOS0, tmp); > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, RESET_PRSR); > + val &= ~AC_PLAY; > + } else { > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, RESET_PRSR); > + val &= ~AC_RECORD; > + } > + > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + > + break; > + default: > + ret = -EINVAL; > + } > + > + return ret; > +} > + > +static int nuc900_ac97_probe(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + unsigned long val; > + > + mutex_lock(&ac97_mutex); > + > + /* enable unit clock */ > + clk_enable(nuc900_audio->clk); > + > + /* enable audio controller and AC-link interface */ > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + val |= (IIS_AC_PIN_SEL | ACLINK_EN); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); > + > + mutex_unlock(&ac97_mutex); > + > + return 0; > +} > + > +static void nuc900_ac97_remove(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + > + clk_disable(nuc900_audio->clk); > +} > + > +static struct snd_soc_dai_ops nuc900_ac97_dai_ops = { > + .trigger = nuc900_ac97_trigger, > +}; > + > +struct snd_soc_dai nuc900_ac97_dai = { > + .name = "nuc900-ac97", > + .probe = nuc900_ac97_probe, > + .remove = nuc900_ac97_remove, > + .ac97_control = 1, > + .playback = { > + .rates = SNDRV_PCM_RATE_8000_48000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels_min = 1, > + .channels_max = 2, > + }, > + .capture = { > + .rates = SNDRV_PCM_RATE_8000_48000, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels_min = 1, > + .channels_max = 2, > + }, > + .ops = &nuc900_ac97_dai_ops, > +} > +EXPORT_SYMBOL_GPL(nuc900_ac97_dai); > + > +static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) > +{ > + struct nuc900_audio *nuc900_audio; > + int ret; > + > + if (nuc900_ac97_data) > + return -EBUSY; > + > + nuc900_audio = kzalloc(sizeof(struct nuc900_audio), GFP_KERNEL); > + if (!nuc900_audio) > + return -ENOMEM; > + > + spin_lock_init(&nuc900_audio->lock); > + > + nuc900_audio->res = platform_get_resource(pdev, IORESOURCE_MEM, 0); > + if (!nuc900_audio->res) { > + ret = -ENODEV; > + goto out0; > + } > + > + if (!request_mem_region(nuc900_audio->res->start, > + resource_size(nuc900_audio->res), pdev->name)) { > + ret = -EBUSY; > + goto out0; > + } > + > + nuc900_audio->mmio = ioremap(nuc900_audio->res->start, > + resource_size(nuc900_audio->res)); > + if (!nuc900_audio->mmio) { > + ret = -ENOMEM; > + goto out1; > + } > + > + nuc900_audio->clk = clk_get(&pdev->dev, NULL); > + if (IS_ERR(nuc900_audio->clk)) { > + ret = PTR_ERR(nuc900_audio->clk); > + goto out2; > + } > + > + nuc900_audio->irq_num = platform_get_irq(pdev, 0); > + if (!nuc900_audio->irq_num) { > + ret = -EBUSY; > + goto out2; > + } > + > + nuc900_ac97_data = nuc900_audio; > + > + nuc900_audio->dev = nuc900_ac97_dai.dev = &pdev->dev; > + > + ret = snd_soc_register_dai(&nuc900_ac97_dai); > + if (ret) > + goto out3; > + > + mfp_set_groupg(nuc900_audio->dev); /* enbale ac97 multifunction pin*/ > + > + return 0; > + > +out3: > + clk_put(nuc900_audio->clk); > +out2: > + iounmap(nuc900_audio->mmio); > +out1: > + release_mem_region(nuc900_audio->res->start, > + resource_size(nuc900_audio->res)); > +out0: > + kfree(nuc900_audio); > + return ret; > +} > + > +static int __devexit nuc900_ac97_drvremove(struct platform_device *pdev) > +{ > + > + snd_soc_unregister_dai(&nuc900_ac97_dai); > + > + clk_put(nuc900_ac97_data->clk); > + iounmap(nuc900_ac97_data->mmio); > + release_mem_region(nuc900_ac97_data->res->start, > + resource_size(nuc900_ac97_data->res)); > + > + nuc900_ac97_data = NULL; > + > + return 0; > +} > + > +static struct platform_driver nuc900_ac97_driver = { > + .driver = { > + .name = "nuc900-audio", > + .owner = THIS_MODULE, > + }, > + .probe = nuc900_ac97_drvprobe, > + .remove = __devexit_p(nuc900_ac97_drvremove), > +}; > + > +static int __init nuc900_ac97_init(void) > +{ > + return platform_driver_register(&nuc900_ac97_driver); > +} > + > +static void __exit nuc900_ac97_exit(void) > +{ > + platform_driver_unregister(&nuc900_ac97_driver); > +} > + > +module_init(nuc900_ac97_init); > +module_exit(nuc900_ac97_exit); > + > +MODULE_AUTHOR("Wan ZongShun <mcuos.com(a)gmail.com>"); > +MODULE_DESCRIPTION("NUC900 AC97 SoC driver!"); > +MODULE_LICENSE("GPL"); > +MODULE_ALIAS("platform:nuc900-ac97"); > + > diff --git a/sound/soc/nuc900/nuc900-audio.c b/sound/soc/nuc900/nuc900-audio.c > new file mode 100644 > index 0000000..b33d5b8 > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-audio.c > @@ -0,0 +1,81 @@ > +/* > + * Copyright (c) 2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#include <linux/module.h> > +#include <linux/moduleparam.h> > +#include <linux/timer.h> > +#include <linux/interrupt.h> > +#include <linux/platform_device.h> > + > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include <sound/soc-dapm.h> > + > +#include "../codecs/ac97.h" > +#include "nuc900-auido.h" > + > +static struct snd_soc_dai_link nuc900evb_ac97_dai = { > + .name = "AC97", > + .stream_name = "AC97 HiFi", > + .cpu_dai = &nuc900_ac97_dai, > + .codec_dai = &ac97_dai, > +}; > + > +static struct snd_soc_card nuc900evb_audio_machine = { > + .name = "NUC900EVB_AC97", > + .dai_link = &nuc900evb_ac97_dai, > + .num_links = 1, > + .platform = &nuc900_soc_platform, > +}; > + > +static struct snd_soc_device nuc900evb_ac97_devdata = { > + .card = &nuc900evb_audio_machine, > + .codec_dev = &soc_codec_dev_ac97, > +}; > + > +static struct platform_device *nuc900evb_asoc_dev; > + > +static int __init nuc900evb_audio_init(void) > +{ > + int ret; > + > + ret = -ENOMEM; > + nuc900evb_asoc_dev = platform_device_alloc("soc-audio", -1); > + if (!nuc900evb_asoc_dev) > + goto out; > + > + /* nuc900 board audio device */ > + platform_set_drvdata(nuc900evb_asoc_dev, &nuc900evb_ac97_devdata); > + > + nuc900evb_ac97_devdata.dev = &nuc900evb_asoc_dev->dev; > + ret = platform_device_add(nuc900evb_asoc_dev); > + > + if (ret) { > + platform_device_put(nuc900evb_asoc_dev); > + nuc900evb_asoc_dev = NULL; > + } > + > +out: > + return ret; > +} > + > +static void __exit nuc900evb_audio_exit(void) > +{ > + platform_device_unregister(nuc900evb_asoc_dev); > +} > + > +module_init(nuc900evb_audio_init); > +module_exit(nuc900evb_audio_exit); > + > +MODULE_LICENSE("GPL"); > +MODULE_DESCRIPTION("NUC900 Series ASoC audio support"); > +MODULE_AUTHOR("Wan ZongShun"); > diff --git a/sound/soc/nuc900/nuc900-auido.h b/sound/soc/nuc900/nuc900-auido.h > new file mode 100644 > index 0000000..95ac4ef > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-auido.h > @@ -0,0 +1,121 @@ > +/* > + * Copyright (c) 2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#ifndef _NUC900_AUDIO_H > +#define _NUC900_AUDIO_H > + > +#include <linux/io.h> > + > +/* Audio Control Registers */ > +#define ACTL_CON 0x00 > +#define ACTL_RESET 0x04 > +#define ACTL_RDSTB 0x08 > +#define ACTL_RDST_LENGTH 0x0C > +#define ACTL_RDSTC 0x10 > +#define ACTL_RSR 0x14 > +#define ACTL_PDSTB 0x18 > +#define ACTL_PDST_LENGTH 0x1C > +#define ACTL_PDSTC 0x20 > +#define ACTL_PSR 0x24 > +#define ACTL_IISCON 0x28 > +#define ACTL_ACCON 0x2C > +#define ACTL_ACOS0 0x30 > +#define ACTL_ACOS1 0x34 > +#define ACTL_ACOS2 0x38 > +#define ACTL_ACIS0 0x3C > +#define ACTL_ACIS1 0x40 > +#define ACTL_ACIS2 0x44 > +#define ACTL_COUNTER 0x48 > + > +/* bit definition of REG_ACTL_CON register */ > +#define R_DMA_IRQ 0x1000 > +#define T_DMA_IRQ 0x0800 > +#define IIS_AC_PIN_SEL 0x0100 > +#define FIFO_TH 0x0080 > +#define ADC_EN 0x0010 > +#define M80_EN 0x0008 > +#define ACLINK_EN 0x0004 > +#define IIS_EN 0x0002 > + > +/* bit definition of REG_ACTL_RESET register */ > +#define W5691_PLAY 0x20000 > +#define ACTL_RESET_BIT 0x10000 > +#define RECORD_RIGHT_CHNNEL 0x08000 > +#define RECORD_LEFT_CHNNEL 0x04000 > +#define PLAY_RIGHT_CHNNEL 0x02000 > +#define PLAY_LEFT_CHNNEL 0x01000 > +#define DAC_PLAY 0x00800 > +#define ADC_RECORD 0x00400 > +#define M80_PLAY 0x00200 > +#define AC_RECORD 0x00100 > +#define AC_PLAY 0x00080 > +#define IIS_RECORD 0x00040 > +#define IIS_PLAY 0x00020 > +#define DAC_RESET 0x00010 > +#define ADC_RESET 0x00008 > +#define M80_RESET 0x00004 > +#define AC_RESET 0x00002 > +#define IIS_RESET 0x00001 > + > +/* bit definition of REG_ACTL_ACCON register */ > +#define AC_BCLK_PU_EN 0x20 > +#define AC_R_FINISH 0x10 > +#define AC_W_FINISH 0x08 > +#define AC_W_RES 0x04 > +#define AC_C_RES 0x02 > + > +/* bit definition of ACTL_RSR register */ > +#define R_FIFO_EMPTY 0x04 > +#define R_DMA_END_IRQ 0x02 > +#define R_DMA_MIDDLE_IRQ 0x01 > + > +/* bit definition of ACTL_PSR register */ > +#define P_FIFO_EMPTY 0x04 > +#define P_DMA_END_IRQ 0x02 > +#define P_DMA_MIDDLE_IRQ 0x01 > + > +/* bit definition of ACTL_ACOS0 register */ > +#define SLOT1_VALID 0x01 > +#define SLOT2_VALID 0x02 > +#define SLOT3_VALID 0x04 > +#define SLOT4_VALID 0x08 > +#define VALID_FRAME 0x10 > + > +/* bit definition of ACTL_ACOS1 register */ > +#define R_WB 0x80 > + > +#define CODEC_READY 0x10 > +#define RESET_PRSR 0x00 > +#define AUDIO_WRITE(addr, val) __raw_writel(val, addr) > +#define AUDIO_READ(addr) __raw_readl(addr) > +#define PCM_TX 0 > +#define PCM_RX 1 > +#define SUBSTREAM_TYPE(substream) \ > + ((substream)->stream == SNDRV_PCM_STREAM_PLAYBACK ? PCM_TX : PCM_RX) > + > +struct nuc900_audio { > + void __iomem *mmio; > + spinlock_t lock; > + dma_addr_t dma_addr[2]; > + unsigned long buffersize[2]; > + unsigned long irq_num; > + struct snd_pcm_substream *substream; > + struct resource *res; > + struct clk *clk; > + struct device *dev; > + > +}; > + > +extern struct nuc900_audio *nuc900_ac97_data; > +extern struct snd_soc_dai nuc900_ac97_dai; > +extern struct snd_soc_platform nuc900_soc_platform; > + > +#endif /*end _NUC900_AUDIO_H */ > diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c > new file mode 100644 > index 0000000..32a503c > --- /dev/null > +++ b/sound/soc/nuc900/nuc900-pcm.c > @@ -0,0 +1,352 @@ > +/* > + * Copyright (c) 2010 Nuvoton technology corporation. > + * > + * Wan ZongShun <mcuos.com(a)gmail.com> > + * > + * This program is free software; you can redistribute it and/or modify > + * it under the terms of the GNU General Public License as published by > + * the Free Software Foundation;version 2 of the License. > + * > + */ > + > +#include <linux/module.h> > +#include <linux/init.h> > +#include <linux/io.h> > +#include <linux/platform_device.h> > +#include <linux/slab.h> > +#include <linux/dma-mapping.h> > + > +#include <sound/core.h> > +#include <sound/pcm.h> > +#include <sound/pcm_params.h> > +#include <sound/soc.h> > + > +#include <mach/hardware.h> > + > +#include "nuc900-auido.h" > + > +static const struct snd_pcm_hardware nuc900_pcm_hardware = { > + .info = SNDRV_PCM_INFO_INTERLEAVED | > + SNDRV_PCM_INFO_BLOCK_TRANSFER | > + SNDRV_PCM_INFO_MMAP | > + SNDRV_PCM_INFO_MMAP_VALID | > + SNDRV_PCM_INFO_PAUSE | > + SNDRV_PCM_INFO_RESUME, > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > + .channels_min = 1, > + .channels_max = 2, > + .buffer_bytes_max = 4*1024, > + .period_bytes_min = 1*1024, > + .period_bytes_max = 4*1024, > + .periods_min = 1, > + .periods_max = 1024, > +}; > + > +static int nuc900_dma_hw_params(struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long flags, stype = SUBSTREAM_TYPE(substream); > + int ret = 0; > + > + spin_lock_irqsave(&nuc900_audio->lock, flags); > + > + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); > + if (ret < 0) > + return ret; > + > + nuc900_audio->substream = substream; > + nuc900_audio->dma_addr[stype] = runtime->dma_addr; > + nuc900_audio->buffersize[stype] = params_buffer_bytes(params); > + > + spin_unlock_irqrestore(&nuc900_audio->lock, flags); > + > + return ret; > +} > + > +static void nuc900_update_dma_register(struct snd_pcm_substream *substream, > + dma_addr_t dma_addr, size_t count) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + void __iomem *mmio_addr, *mmio_len; > + > + if (SUBSTREAM_TYPE(substream) == PCM_TX) { > + mmio_addr = nuc900_audio->mmio + ACTL_PDSTB; > + mmio_len = nuc900_audio->mmio + ACTL_PDST_LENGTH; > + } else { > + mmio_addr = nuc900_audio->mmio + ACTL_RDSTB; > + mmio_len = nuc900_audio->mmio + ACTL_RDST_LENGTH; > + } > + > + AUDIO_WRITE(mmio_addr, dma_addr); > + AUDIO_WRITE(mmio_len, count); > +} > + > +static void nuc900_dma_start(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long val; > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + val |= (T_DMA_IRQ | R_DMA_IRQ); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); > +} > + > +static void nuc900_dma_stop(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long val; > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + val &= ~(T_DMA_IRQ | R_DMA_IRQ); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val); > +} > + > +static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id) > +{ > + struct snd_pcm_substream *substream = dev_id; > + struct nuc900_audio *nuc900_audio = substream->runtime->private_data; > + unsigned long val; > + > + spin_lock(&nuc900_audio->lock); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_CON); > + > + if (val & R_DMA_IRQ) { > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | R_DMA_IRQ); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RSR); > + > + if (val & R_DMA_MIDDLE_IRQ) { > + val |= R_DMA_MIDDLE_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val); > + } > + > + if (val & R_DMA_END_IRQ) { > + val |= R_DMA_END_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RSR, val); > + } > + } else if (val & T_DMA_IRQ) { > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_CON, val | T_DMA_IRQ); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_PSR); > + > + if (val & P_DMA_MIDDLE_IRQ) { > + val |= P_DMA_MIDDLE_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val); > + } > + > + if (val & P_DMA_END_IRQ) { > + val |= P_DMA_END_IRQ; > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_PSR, val); > + } > + } else { > + dev_err(nuc900_audio->dev, "Wrong DMA interrupt status!\n"); > + spin_unlock(&nuc900_audio->lock); > + return IRQ_HANDLED; > + } > + > + spin_unlock(&nuc900_audio->lock); > + > + snd_pcm_period_elapsed(substream); > + > + return IRQ_HANDLED; > +} > + > +static int nuc900_dma_hw_free(struct snd_pcm_substream *substream) > +{ > + snd_pcm_lib_free_pages(substream); > + return 0; > +} > + > +static int nuc900_dma_prepare(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + unsigned long flags, val, stype = SUBSTREAM_TYPE(substream);; > + > + spin_lock_irqsave(&nuc900_audio->lock, flags); > + > + nuc900_update_dma_register(substream, > + nuc900_audio->dma_addr[stype], nuc900_audio->buffersize[stype]); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_RESET); > + > + switch (runtime->channels) { > + case 1: > + if (PCM_TX == stype) { > + val &= ~(PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); > + val |= PLAY_RIGHT_CHNNEL; > + } else { > + val &= ~(RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL); > + val |= RECORD_RIGHT_CHNNEL; > + } > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + break; > + case 2: > + if (PCM_TX == stype) > + val |= (PLAY_LEFT_CHNNEL | PLAY_RIGHT_CHNNEL); > + else > + val |= (RECORD_LEFT_CHNNEL | RECORD_RIGHT_CHNNEL); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_RESET, val); > + break; > + default: > + return -EINVAL; > + } > + spin_unlock_irqrestore(&nuc900_audio->lock, flags); > + return 0; > +} > + > +static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) > +{ > + int ret = 0; > + > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + nuc900_dma_start(substream); > + break; > + > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + nuc900_dma_stop(substream); > + break; > + > + default: > + ret = -EINVAL; > + break; > + } > + > + return ret; > +} > + > +int nuc900_dma_getposition(struct snd_pcm_substream *substream, > + dma_addr_t *src, dma_addr_t *dst) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + > + if (src != NULL) > + *src = AUDIO_READ(nuc900_audio->mmio + ACTL_PDSTC); > + > + if (dst != NULL) > + *dst = AUDIO_READ(nuc900_audio->mmio + ACTL_RDSTC); > + > + return 0; > +} > + > +static snd_pcm_uframes_t nuc900_dma_pointer(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + dma_addr_t src, dst; > + unsigned long res; > + > + nuc900_dma_getposition(substream, &src, &dst); > + > + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) > + res = dst - runtime->dma_addr; > + else > + res = src - runtime->dma_addr; > + > + return bytes_to_frames(substream->runtime, res); > +} > + > +static int nuc900_dma_open(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio; > + > + snd_soc_set_runtime_hwparams(substream, &nuc900_pcm_hardware); > + > + nuc900_audio = nuc900_ac97_data; > + > + if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt, > + IRQF_DISABLED, "nuc900-dma", substream)) > + return -EBUSY; > + > + runtime->private_data = nuc900_audio; > + > + return 0; > +} > + > +static int nuc900_dma_close(struct snd_pcm_substream *substream) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + struct nuc900_audio *nuc900_audio = runtime->private_data; > + > + free_irq(nuc900_audio->irq_num, substream); > + > + return 0; > +} > + > +static int nuc900_dma_mmap(struct snd_pcm_substream *substream, > + struct vm_area_struct *vma) > +{ > + struct snd_pcm_runtime *runtime = substream->runtime; > + > + return dma_mmap_writecombine(substream->pcm->card->dev, vma, > + runtime->dma_area, > + runtime->dma_addr, > + runtime->dma_bytes); > +} > + > +static struct snd_pcm_ops nuc900_dma_ops = { > + .open = nuc900_dma_open, > + .close = nuc900_dma_close, > + .ioctl = snd_pcm_lib_ioctl, > + .hw_params = nuc900_dma_hw_params, > + .hw_free = nuc900_dma_hw_free, > + .prepare = nuc900_dma_prepare, > + .trigger = nuc900_dma_trigger, > + .pointer = nuc900_dma_pointer, > + .mmap = nuc900_dma_mmap, > +}; > + > +static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) > +{ > + snd_pcm_lib_preallocate_free_for_all(pcm); > +} > + > +static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); > +static int nuc900_dma_new(struct snd_card *card, > + struct snd_soc_dai *dai, struct snd_pcm *pcm) > +{ > + if (!card->dev->dma_mask) > + card->dev->dma_mask = &nuc900_pcm_dmamask; > + if (!card->dev->coherent_dma_mask) > + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); > + > + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > + card->dev, 4 * 1024, (4 * 1024) - 1); > + > + return 0; > +} > + > +struct snd_soc_platform nuc900_soc_platform = { > + .name = "nuc900-dma", > + .pcm_ops = &nuc900_dma_ops, > + .pcm_new = nuc900_dma_new, > + .pcm_free = nuc900_dma_free_dma_buffers, > +} > +EXPORT_SYMBOL_GPL(nuc900_soc_platform); > + > +static int __init nuc900_soc_platform_init(void) > +{ > + return snd_soc_register_platform(&nuc900_soc_platform); > +} > + > +static void __exit nuc900_soc_platform_exit(void) > +{ > + snd_soc_unregister_platform(&nuc900_soc_platform); > +} > + > +module_init(nuc900_soc_platform_init); > +module_exit(nuc900_soc_platform_exit); > + > +MODULE_AUTHOR("Wan ZongShun, <mcuos.com(a)gmail.com>"); > +MODULE_DESCRIPTION("nuc900 Audio DMA module"); > +MODULE_LICENSE("GPL"); > -- > 1.6.3.3 > -- *linux-arm-kernel mailing list mail addr:linux-arm-kernel@lists.infradead.org you can subscribe by: http://lists.infradead.org/mailman/listinfo/linux-arm-kernel * linux-arm-NUC900 mailing list mail addr:NUC900@googlegroups.com main web: https://groups.google.com/group/NUC900 you can subscribe it by sending me mail: mcuos.com(a)gmail.com

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[alsa-devel] ASoC: Dummy codec driver?
by Joonyoung Shim 19 May '10

19 May '10

Hi, I have a target(The cpu is S5PC110) which doesn't have audio codec, but it needs the audio device node to play audio at the TV via HDMI. The S5PC110 HDMI transmits itself to TV the audio data from I2S buffer, so i have made dummy codec driver to create audio device node and to transmit audio data to I2S buffer. How do you think about the dummy codec driver?

2 1

[alsa-devel] [PATCH v2 0/2] Add earphone speaker in OMAP4
by Jorge Eduardo Candelaria 19 May '10

19 May '10

Add support for earphone speaker in SDP4430 board. Changes from v1: - Separated changes for codec and machine drivers in two patches. - Added Acked-by tag to patches. Jorge Eduardo Candelaria (2): ASoC: SDP4430: Add support for Earphone speaker ASoC: TWL6040: Enable earphone path in codec sound/soc/codecs/twl6040.c | 21 ++++++++++++++++++++- sound/soc/omap/sdp4430.c | 4 ++++ 2 files changed, 24 insertions(+), 1 deletions(-)

2 3

[alsa-devel] [PATCH 1/2] ASoC: SDP4430: Add sdp4430 machine driver
by Jorge Eduardo Candelaria 19 May '10

19 May '10

From: Misael Lopez Cruz <x0052729(a)ti.com> Add ASoC support for TI SDP4430. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> --- sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp4430.c | 228 ++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/sdp4430.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 4292701..c23c4a6 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -102,6 +102,15 @@ config SND_OMAP_SOC_SDP3430 Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. +config SND_OMAP_SOC_SDP4430 + tristate "SoC Audio support for Texas Instruments SDP4430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP + select SND_OMAP_SOC_MCPDM + select SND_SOC_TWL6040 + help + Say Y if you want to add support for SoC audio on Texas Instruments + SDP4430. + config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3a75755..ba9fc65 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -17,6 +17,7 @@ snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-sdp4430-objs := sdp4430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o @@ -31,6 +32,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c new file mode 100644 index 0000000..83831ee --- /dev/null +++ b/sound/soc/omap/sdp4430.c @@ -0,0 +1,228 @@ +/* + * sdp4430.c -- SoC audio for TI OMAP4430 SDP + * + * Author: Misael Lopez Cruz <x0052729(a)ti.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <plat/hardware.h> +#include <plat/mux.h> + +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" +#include "../codecs/twl6040.h" + +static int twl6040_power_mode; + +static int sdp4430_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->dai->codec_dai; + int clk_id, freq; + int ret; + + if (twl6040_power_mode) { + clk_id = TWL6040_SYSCLK_SEL_HPPLL; + freq = 38400000; + } else { + clk_id = TWL6040_SYSCLK_SEL_LPPLL; + freq = 32768; + } + + /* set the codec mclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq, + SND_SOC_CLOCK_IN); + if (ret) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } +} + +static struct snd_soc_ops sdp4430_ops = { + .hw_params = sdp4430_hw_params, +}; + +static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = twl6040_power_mode; + return 0; +} + +static int sdp4430_set_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (twl6040_power_mode == ucontrol->value.integer.value[0]) + return 0; + + twl6040_power_mode = ucontrol->value.integer.value[0]; + + return 1; +} + +static const char *power_texts[] = {"Low-Power", "High-Performance"}; + +static const struct soc_enum sdp4430_enum[] = { + SOC_ENUM_SINGLE_EXT(2, power_texts), +}; + +static const struct snd_kcontrol_new sdp4430_controls[] = { + SOC_ENUM_EXT("TWL6040 Power Mode", sdp4430_enum[0], + sdp4430_get_power_mode, sdp4430_set_power_mode), +}; + +/* SDP4430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Main Mic Bias"}, + {"SUBMIC", NULL, "Main Mic Bias"}, + {"Main Mic Bias", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, +}; + +static int sdp4430_twl6040_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add SDP4430 specific controls */ + ret = snd_soc_add_controls(codec, sdp4430_controls, + ARRAY_SIZE(sdp4430_controls)); + if (ret) + return ret; + + /* Add SDP4430 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, sdp4430_twl6040_dapm_widgets, + ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP4430 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* SDP4430 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + + /* TWL6040 not connected pins */ + snd_soc_dapm_nc_pin(codec, "AFML"); + snd_soc_dapm_nc_pin(codec, "AFMR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp4430_dai = { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai = &omap_mcpdm_dai, + .codec_dai = &twl6040_dai, + .init = sdp4430_twl6040_init, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sdp4430 = { + .name = "SDP4430", + .platform = &omap_soc_platform, + .dai_link = &sdp4430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp4430_snd_devdata = { + .card = &snd_soc_sdp4430, + .codec_dev = &soc_codec_dev_twl6040, +}; + +static struct platform_device *sdp4430_snd_device; + +static int __init sdp4430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_4430sdp()) { + pr_debug("Not SDP4430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP4430 SoC init\n"); + + sdp4430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp4430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp4430_snd_device, &sdp4430_snd_devdata); + sdp4430_snd_devdata.dev = &sdp4430_snd_device->dev; + + ret = platform_device_add(sdp4430_snd_device); + if (ret) + goto err; + + /* Codec starts in HP mode */ + twl6040_power_mode = 1; + + return 0; + +err: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp4430_snd_device); + return ret; +} +module_init(sdp4430_soc_init); + +static void __exit sdp4430_soc_exit(void) +{ + platform_device_unregister(sdp4430_snd_device); +} +module_exit(sdp4430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); +MODULE_DESCRIPTION("ALSA SoC SDP4430"); +MODULE_LICENSE("GPL"); + -- 1.6.3.3

3 2

[alsa-devel] [PATCH - alsa-lib] tlv: Check dB range only within the control's volume range
by Peter Ujfalusi 19 May '10

19 May '10

The DB_RANGE need to be used on some HW, since the gain on volume control is not continuous, and has to be divided into several sub DB_SCALE ranges. ASoC has a feature to override the HW default volume range, and in this case when the volume range is less than the HW maximum we do not need to go through the whole DB_RANGE, but we need to stop where the kcontrol's maximum tell us. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello, this patch fixes the ASoC related problem, when the volume is limited by the machine driver on a given control, which happen to have DB_RANGE gain mapping. The DB_RANGE has to have _only_ DB_SCALE type of mapping, since with LINEAR, MINMAX, MINMAX_MUTE it is not possible to accurately calculate the maximum dB limit on the control, when the volume is limited. Applications like PulseAudio need to know the actual dB range, and not only the theoretical maximum. src/control/tlv.c | 11 ++++++++--- 1 files changed, 8 insertions(+), 3 deletions(-) diff --git a/src/control/tlv.c b/src/control/tlv.c index cfd9b97..0ff052e 100644 --- a/src/control/tlv.c +++ b/src/control/tlv.c @@ -140,10 +140,13 @@ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax, pos = 2; while (pos + 4 <= len) { long rmin, rmax; - rangemin = (int)tlv[pos]; - rangemax = (int)tlv[pos + 1]; + long submin, submax; + submin = (int)tlv[pos]; + submax = (int)tlv[pos + 1]; + if (rangemax < submax) + submax = rangemax; err = snd_tlv_get_dB_range(tlv + pos + 2, - rangemin, rangemax, + submin, submax, &rmin, &rmax); if (err < 0) return err; @@ -156,6 +159,8 @@ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax, *min = rmin; *max = rmax; } + if (rangemax == submax) + return 0; pos += int_index(tlv[pos + 3]) + 4; } return 0; -- 1.7.1

2 4

[alsa-devel] [PATCH] ASoC: tpa6130a2: Remove CPVSS supply
by Jarkko Nikula 19 May '10

19 May '10

I believe there is no need for the CPVSS supply as this pin is for decoupling capacitor for the internal charge pump. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- sound/soc/codecs/tpa6130a2.c | 40 +++++++++++++++++++++------------------- 1 files changed, 21 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 221bb90..87a5b09 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -36,13 +36,14 @@ static struct i2c_client *tpa6130a2_client; -#define TPA6130A2_NUM_SUPPLIES 2 +#define TPA6130A2_NUM_SUPPLIES 1 +#define TPA6140A2_NUM_SUPPLIES 2 +#define TPA61x0A2_MAX_SUPPLIES 2 static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "CPVSS", "Vdd", }; -static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { +static const char *tpa6140a2_supply_names[TPA6140A2_NUM_SUPPLIES] = { "HPVdd", "AVdd", }; @@ -51,7 +52,8 @@ static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { struct tpa6130a2_data { struct mutex mutex; unsigned char regs[TPA6130A2_CACHEREGNUM]; - struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; + struct regulator_bulk_data supplies[TPA61x0A2_MAX_SUPPLIES]; + int num_supplies; int power_gpio; unsigned char power_state; enum tpa_model id; @@ -136,7 +138,7 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), + ret = regulator_bulk_enable(data->num_supplies, data->supplies); if (ret != 0) { dev_err(&tpa6130a2_client->dev, @@ -161,7 +163,7 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), + ret = regulator_bulk_disable(data->num_supplies, data->supplies); if (ret != 0) { dev_err(&tpa6130a2_client->dev, @@ -412,6 +414,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; + const char **supply_names; int i, ret; dev = &client->dev; @@ -454,23 +457,22 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, } switch (data->id) { + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); case TPA6130A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + data->num_supplies = TPA6130A2_NUM_SUPPLIES; + supply_names = tpa6130a2_supply_names; break; case TPA6140A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6140a2_supply_names[i];; + data->num_supplies = TPA6140A2_NUM_SUPPLIES; + supply_names = tpa6140a2_supply_names; break; - default: - dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", - pdata->id); - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; } + for (i = 0; i < data->num_supplies; i++) + data->supplies[i].supply = supply_names[i]; - ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_bulk_get(dev, data->num_supplies, data->supplies); if (ret != 0) { dev_err(dev, "Failed to request supplies: %d\n", ret); goto err_regulator; @@ -495,7 +497,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, return 0; err_power: - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_bulk_free(data->num_supplies, data->supplies); err_regulator: if (data->power_gpio >= 0) gpio_free(data->power_gpio); @@ -516,7 +518,7 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_bulk_free(data->num_supplies, data->supplies); kfree(data); tpa6130a2_client = NULL; -- 1.7.1

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[alsa-devel] [PATCH 1/1] ALSA: usb-audio: add support for Akai MPD16
by wdev＠foltman.com 18 May '10

18 May '10

From: Krzysztof Foltman <wdev(a)foltman.com> The decoding routine is based on own reverse-engineering. It seems to recognize all the messages that MPD16 sends during normal operation (that is, via standard MIDI messages). Configuration (changing pad sensitivity, slider controller and MIDI notes) is not supported in this version. Signed-off-by: Krzysztof Foltman <wdev(a)foltman.com> diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c1558c..57d642a 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -645,6 +645,34 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = { }; /* + * AKAI MPD16 protocol: one or more chunks consisting of first byte of + * (0x20 | msg_len) and then a MIDI message (msg_len bytes long) + * + * Messages sent: + * 21 FE (active sense) + * 23 90 xx xx (note on) + * 23 Ax xx xx (polyphonic pressure) + * 23 Bx xx xx (control change) + */ +static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep, + uint8_t *buffer, int buffer_length) +{ + unsigned int pos = 0; + while (pos < (unsigned)buffer_length && (buffer[pos] & 0xF8) == 0x20) + { + int msg_len = buffer[pos] & 0x0f; + snd_usbmidi_input_data(ep, 0, &buffer[pos + 1], msg_len); + pos += 1 + msg_len; + } +} + +static struct usb_protocol_ops snd_usbmidi_akai_ops = { + .input = snd_usbmidi_akai_input, + .output = snd_usbmidi_standard_output, + .output_packet = snd_usbmidi_output_standard_packet, +}; + +/* * Novation USB MIDI protocol: number of data bytes is in the first byte * (when receiving) (+1!) or in the second byte (when sending); data begins * at the third byte. @@ -1434,6 +1462,8 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Akai MPD16 */ + EXTERNAL_PORT(0x09e8, 0x0062, 0, "%s MIDI"), /* Access Music Virus TI */ EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, @@ -2035,6 +2065,13 @@ int snd_usbmidi_create(struct snd_card *card, umidi->usb_protocol_ops = &snd_usbmidi_cme_ops; err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; + case QUIRK_MIDI_AKAI: + umidi->usb_protocol_ops = &snd_usbmidi_akai_ops; + err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); + /* endpoint 0 does not carry MIDI data */ + endpoints[0].out_cables = 0; + endpoints[0].in_cables = 0; + break; default: snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type); err = -ENXIO; diff --git a/sound/usb/midi.h b/sound/usb/midi.h index 2089ec9..2fca80b 100644 --- a/sound/usb/midi.h +++ b/sound/usb/midi.h @@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_CME, data is NULL */ +/* for QUIRK_MIDI_AKAI, data is NULL */ + int snd_usbmidi_create(struct snd_card *card, struct usb_interface *iface, struct list_head *midi_list, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 91ddef3..f8797f6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* AKAI devices */ +{ + USB_DEVICE(0x09e8, 0x0062), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "AKAI", + .product_name = "MPD16", + .ifnum = 0, + .type = QUIRK_MIDI_AKAI, + } +}, + /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 136e5b4..b45e54c 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, [QUIRK_MIDI_CME] = create_any_midi_quirk, + [QUIRK_MIDI_AKAI] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d679e72..06ebf24 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -74,6 +74,7 @@ enum quirk_type { QUIRK_MIDI_FASTLANE, QUIRK_MIDI_EMAGIC, QUIRK_MIDI_CME, + QUIRK_MIDI_AKAI, QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, -- 1.7.0.4

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[alsa-devel] [GIT PULL] ASoC updates for 2.6.35
by Liam Girdwood 18 May '10

18 May '10

Hi Takashi, Please pull the following ASoC updates for 2.6.35. We do have Tony's Ack for the OMAP mcbsp stuff (as it simplifies the merging). Thanks Liam --- The following changes since commit 6a2f1ee1f9bb5346644105c9355e9e06f6a847d3: Mark Brown (1): ASoC: Don't restart unconfigured WM8994 FLLs are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-2.6.35 Felipe Balbi (1): ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function Jorge Eduardo Candelaria (2): ARM: McBSP: Fix request for irq in OMAP4 ARM: McBSP: Add support for omap4 in McBSP driver Peter Ujfalusi (2): ASoC: TWL4030: Add control for digimic Left Right swap ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF Sergey Lapin (1): OMAP: McBSP: Add 32-bit mode support arch/arm/mach-omap2/mcbsp.c | 12 ++++-------- arch/arm/plat-omap/mcbsp.c | 34 +++++++++++++++++++--------------- sound/soc/codecs/tlv320dac33.c | 5 ++++- sound/soc/codecs/twl4030.c | 12 ++++++++++++ sound/soc/omap/omap-mcbsp.c | 28 ++++++++++++++++++++++++---- 5 files changed, 63 insertions(+), 28 deletions(-)

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[alsa-devel] ASoC: OMAP4: Add support for Earphone speaker
by Jorge Eduardo Candelaria 18 May '10

18 May '10

Add control to enable earphone driver in TWL6040 codec. This driver is connected to HSDAC Left. Also enable earphone speaker in sdp4430 machine driver. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/codecs/twl6040.c | 21 ++++++++++++++++++++- sound/soc/omap/sdp4430.c | 5 +++++ 2 files changed, 25 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2ae442e..af36346 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -432,6 +432,12 @@ static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); */ static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); +/* + * EPGAIN volume control: + * from -24 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(ep_tlv, -2400, 200, 0); + /* Left analog microphone selection */ static const char *twl6040_amicl_texts[] = {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; @@ -479,6 +485,9 @@ static const struct snd_kcontrol_new hfl_driver_switch_controls = static const struct snd_kcontrol_new hfr_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); +static const struct snd_kcontrol_new ep_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", @@ -491,7 +500,8 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), SOC_DOUBLE_R_TLV("Handsfree Playback Volume", TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), - + SOC_SINGLE_TLV("Earphone Playback Volume", + TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), }; static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { @@ -507,6 +517,7 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HSOR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("EP"), /* Analog input muxes for the capture amplifiers */ SND_SOC_DAPM_MUX("Analog Left Capture Route", @@ -572,6 +583,10 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Earphone Driver", + SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HFDAC Left PGA", @@ -607,6 +622,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, + /* Earphone playback path */ + {"Earphone Driver", "Switch", "HSDAC Left"}, + {"EP", NULL, "Earphone Driver"}, + /* Handsfree playback path */ {"HFDAC Left Playback", "Switch", "HFDAC Left"}, {"HFDAC Right Playback", "Switch", "HFDAC Right"}, diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 83831ee..ab45621 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_SPK("Earphone Spk", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -120,6 +121,10 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Headset Stereophone (Headphone): HSOL, HSOR */ {"Headset Stereophone", NULL, "HSOL"}, {"Headset Stereophone", NULL, "HSOR"}, + + /* Earphone speaker */ + {"Earphone Spk", NULL, "EP"}, + }; static int sdp4430_twl6040_init(struct snd_soc_codec *codec) -- 1.6.3.3

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[alsa-devel] Creating a /dev entry.
by Adish Kuvelker 18 May '10

18 May '10

Hello, I am writing a ALSA driver for the AtlasIII AC'97 Controller. The controller communicates with the AC'97 Codec and the necessary PCM, Sequencer, Control and Timer nodes are created in the /sys. Also I get the necessary information registered in the procfs under /proc/asound. However, I do not see any sound nodes created under /dev. Are these nodes created by the sound driver registration function which creates necessary files under /sys or will I have to use utilities like "snddevices" to create the "snd" node under /dev. I read that "snddevices" would create the necessary entries under /dev/snd, and I reckon that it does not create the entry "snd" under /dev. Although I am using kernel 2.6.19.2, it doesn't have the "udev" enabled. Will I need the "udev" to create these nodes under /dev? Regards; Aadish

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[alsa-devel] RFC:
by Peter Ujfalusi 18 May '10

18 May '10

Hello, I apologize for the long mail, but I feel that I need to explain this in a bit longish way... ASoC core recently gained support for limiting the volume control's range (maximum), which is really useful in embedded systems. The implementation for this is really simple: The codec drivers are implementing the full range the HW can support. Machine drivers can than override the maximum volume on a given control if it is necessary. This works well with most of the controls: Volume controls without dB scale. Volume controls with DECLARE_TLV_DB_SCALE() Problems started to emerge, when the volume is limited on control, which has SNDRV_CTL_TLVT_DB_RANGE (from the sound/soc/codecs/tpa6130a2.c): static const unsigned int tpa6140_tlv[] = { TLV_DB_RANGE_HEAD(3), 0, 8, TLV_DB_SCALE_ITEM(-5900, 400, 0), 9, 16, TLV_DB_SCALE_ITEM(-2500, 200, 0), 17, 31, TLV_DB_SCALE_ITEM(-1000, 100, 0), }; static const struct snd_kcontrol_new tpa6140a2_controls[] = { SOC_SINGLE_EXT_TLV("TPA6140A2 Headphone Playback Volume", TPA6130A2_REG_VOL_MUTE, 1, 0x1f, 0, tpa6130a2_get_volsw, tpa6130a2_put_volsw, tpa6140_tlv), }; The original HW supported range is 0 .. 31 (-59 .. +4 dB). When the machine driver sets the limit to 21 (-6 dB), than the volume range is reported correctly as 0 .. 21, but snd_mixer_selem_get_playback_dB_range() reports range of -59 .. +4 dB. This is not correct, it should report: -59 .. -6 dB. If the control has single DECLARE_TLV_DB_SCALE() after the limiting both volume and dB range is reported correctly. The explanation for this is in the alsa-lib: src/control/tlv.c: /** * \brief Get the dB min/max values * \param tlv the TLV source returned by #snd_tlv_parse_dB_info() * \param rangemin the minimum value of the raw volume * \param rangemax the maximum value of the raw volume * \param min the pointer to store the minimum dB value (in 0.01dB unit) * \param max the pointer to store the maximum dB value (in 0.01dB unit) * \return 0 if successful, or a negative error code */ int snd_tlv_get_dB_range(unsigned int *tlv, long rangemin, long rangemax, long *min, long *max) { int err; switch (tlv[0]) { case SND_CTL_TLVT_DB_RANGE: { unsigned int pos, len; len = int_index(tlv[1]); if (len > MAX_TLV_RANGE_SIZE) return -EINVAL; pos = 2; while (pos + 4 <= len) { long rmin, rmax; rangemin = (int)tlv[pos]; rangemax = (int)tlv[pos + 1]; err = snd_tlv_get_dB_range(tlv + pos + 2, rangemin, rangemax, &rmin, &rmax); if (err < 0) return err; if (pos > 2) { if (rmin < *min) *min = rmin; if (rmax > *max) *max = rmax; } else { *min = rmin; *max = rmax; } pos += int_index(tlv[pos + 3]) + 4; } return 0; } case SND_CTL_TLVT_DB_SCALE: { int step; *min = (int)tlv[2]; step = (tlv[3] & 0xffff); *max = *min + (long)(step * (rangemax - rangemin)); return 0; } case SND_CTL_TLVT_DB_MINMAX: case SND_CTL_TLVT_DB_MINMAX_MUTE: case SND_CTL_TLVT_DB_LINEAR: *min = (int)tlv[2]; *max = (int)tlv[3]; return 0; } return -EINVAL; } In case of DECLARE_TLV_DB_SCALE(), the SND_CTL_TLVT_DB_SCALE case is taken, and the control's maximum, and minimum range are used for the dB scale limit calculation. But, when SNDRV_CTL_TLVT_DB_RANGE is used (tpa6130a2 driver), than that case will dismisses the control's maximum and minimum range, and goes through the whole TLV struct, and reports the HW supported range. If it would stop at the driver reported maximum, than the dB range reported would be correct. So a 'simple' patch to alsa-lib would solve this. But what happens, if we want to limit control, which has DB_MINMAX, DB_MINMAX_MUTE, or DB_LINEAR? There is no way to actually limit those in a way, that the dB scale reported would be correct. We just stretch the steps bigger, like what happens with the SNDRV_CTL_TLVT_DB_RANGE in the tpa6130a2 driver. When looking for users of DB_LINEAR I have found 8 drivers, DB_MINMAX is used by usbmixer only, the DB_MINMAX_MUTE has no users at all. Since both DB_LINEAR and DB_MINMAX can be presented as DB_SCALE, and the effort to convert the drivers should not be that big. In this way, we will only have drivers using DB_SCALE in kernel, we can mark the other DB_ types as deprecated (but keeping the support for out of tree drivers), and the alsa-lib's tlv code can be adjusted to support properly the volume limiting. If there is interest, than the ALSA core could also gain support for platform/configuration dependent volume limiting. If this is too big change to ALSA (which I think it is not, since for the user it makes no difference, and the volume range, dB scale not going to be changed), than probably cleaning up the sound/soc/ to use _only_ DB_SCALE type for volume controls might be the solution, since the volume limiting feature is only in ASoC AFAIK. Either way, the alsa-lib's src/control/tlv.c:snd_tlv_get_dB_range function in case of SND_CTL_TLVT_DB_RANGE needs some fix to handle the situation, when the volume range is _not_ at the HW supported range. To summarize, I'd like to do (or ask help to do) one of the following [1] Replace the use of DB_LINEAR, DB_MINMAX with DB_SCALE in all drivers under sound/ Document, that the now unused DB_ types are deprecated Fix alsa-lib to support the volume limiting in regards of dB scale [2] Replace the use of DB_LINEAR with DB_SCALE in all drivers under sound/soc/ Fix alsa-lib to support the volume limiting in regards of dB scale What do you think? -- Péter

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Re: [alsa-devel] [PATCH] NUC900/audio: add nuc900 audio driver support
by Mark Brown 18 May '10

18 May '10

On Sun, May 16, 2010 at 11:34:01PM +0800, Wan ZongShun wrote: This all looks pretty good. A few comments below but they're all fairly minor and should be easy to address. > index b1749bc..81d4848 100644 > --- a/sound/soc/Kconfig > +++ b/sound/soc/Kconfig > @@ -36,6 +36,7 @@ source "sound/soc/s3c24xx/Kconfig" > source "sound/soc/s6000/Kconfig" > source "sound/soc/sh/Kconfig" > source "sound/soc/txx9/Kconfig" > +source "sound/soc/nuc900/Kconfig" Please keep the Kconfig and Makefile sorted, this helps avoid merge issues. > +#define NUC900_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ > + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ > + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) SNDRV_PCM_RATE_8000_48000. > + if (!timeout) > + dev_err(nuc900_audio->dev, "AC97 read register time out !\n"); > + > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACOS0) ; If the read timed out shouldn't we be returning rather than continuing? > + val = AUDIO_READ(nuc900_audio->mmio + ACTL_ACCON); > + val &= (~AC_C_RES); > + AUDIO_WRITE(nuc900_audio->mmio + ACTL_ACCON, val); > + udelay(1000); > + if (!(AUDIO_READ(nuc900_audio->mmio + ACTL_ACIS0) & CODEC_READY)) > + dev_err(nuc900_audio->dev, "AC97 codec cold reset failed!\n"); What is this actually checking in the hardware? Not all CODECs enable the AC97 link by default after a cold reset, the standard allows them to power up in a low power state which will On the other hand, given that there's no warm reset operation perhaps this isn't a big deal. > +static void nuc900_ac97_remove(struct platform_device *pdev, > + struct snd_soc_dai *dai) > +{ > + struct nuc900_audio *nuc900_audio = nuc900_ac97_data; > + > + /* Enable unit clock */ > + clk_disable(nuc900_audio->clk); Bit rot in the comments here. > +/* bit definition of REG_ACTL_CON register */ > +#define AUDCLK_EN 0x8000 > +#define PFIFO_EN 0x4000 > +#define RFIFO_EN 0x2000 These constants (and the rest) really should be namespaced - they're likely to collide with other definitions in, for example, CODEC drivers used by machines. > +#define IIS_EN 0x0002 Looks like there's I2S support to come? > +static irqreturn_t nuc900_dma_interrupt(int irq, void *dev_id) > +{ > + > + snd_pcm_period_elapsed(substream); > + > + return IRQ_HANDLED; This is done unconditionally - are you sure there can't be any spurious interrupts (eg, error interrupts). It shouldn't cause any harm, of course.

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[alsa-devel] soc multi chip support
by Attila RS 17 May '10

17 May '10

Hi, I have 2 of the same audio codecs connected to an omap (each codec has it's own i2c address and each has its own mcbsp channel). I can configure and use each one individually, but have not found a way to configure both at the same time so that each codec shows up as an alsa device. I did some searching and found that this may not be supported yet. Is that the case or has anyone done this yet or is working on it? Attila

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[alsa-devel] [PATCH 0/2] ASoC: tlv320dac33: small power handling updates
by Peter Ujfalusi 17 May '10

17 May '10

Hello, While testing suspend, it has been noticed, that the driver would try to power off the codec more than once. Since this case handled in the driver, and results no functional defect we can lower the noise in the kernel log. Also avoid the root cause of the multiple off calls by filtering BIAS_OFF request when the codec was already OFF. --- Felipe Balbi (1): ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function Peter Ujfalusi (1): ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF sound/soc/codecs/tlv320dac33.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-)

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[alsa-devel] enhanced Creative X-Fi support
by Sebastian Reinhardt 17 May '10

17 May '10

@Takashi: I own a Creative X-Fi Elite Pro and I use openSuSE. Since few months, the base card is supported by alsa. Big thanks, great work. But I like to use also the external I/O- terminal, delivered with ElitePro- cards. Is there any chance to get this working. I'm no real linux kernel hacker, but I can give You every infomation about the terminal, if You need (I had already disassembled the I/O- box). I do not need the effects, but In like to use the additional in-/outputs and volume knob. To use the ir- control, would be great, too. -- Kind Regaards Sebastian Reinhardt

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