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[alsa-devel] [PATCH 0/2] Correct widget and PLL handling
by Jorge Eduardo Candelaria 21 May '10

21 May '10

Fix an issue when using 19.2 Mhz as clock input for TWL6040. Also, Drivers in TWL6040 can be modeled after PGA components of DAPM. This allows for better power management and also ensures that the DACs are enabled before the Drivers, reducing this way a pop-noise heard when powering up components.

3 5

[alsa-devel] [ice1712] Word Clock with Delta1010 E
by Bernhard Walle 21 May '10

21 May '10

Hi, we have a problem with the word clock input of the Delta 1010 E. At first: I modified the source code of envy24control to accept the Delta 1010 E as subdevice so that it allows to set the word clock. I'll post the patch after it works with the driver. However, the problem also exists when I use plain alsamixer or amixer. With the unmodified driver, I'm able to turn word clock on but whenever I record something, the arecord program (or Audacity) doesn't get any samples. With OSS (which we're using now but we're trying to switch to ALSA) my colleague Markus Prosch made a patch with lots of try & error that made the word clock working: http://opensound.hg.sourceforge.net/hgweb/opensound/opensound/rev/6a4bad6e7… When I try to "port" that patch to ALSA, the behaviour changes: The recording works, but the sample rate is too low so that it sounds quite strange. In any case: In the same hardware setup, it works flawlessly with OSS. Also, we had no problem with the older Delta 1010. It seems to be an incompatible change (or bug ;)) in revision E. Are there any devs here that know the hardware better and could help me in debugging that issue? Without any deeper sound driver knowledge and without any data sheet, that's quite hard for me. Thanks for your help! Regards, Bernhard PS: Please keep me in Cc since I'm not on the list.

2 2

[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

1 0

[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

4 7

[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

10 56

[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(-)

2 1

[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

2 1

[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

2 2

[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" }

3 2

[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

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