From 7ad7b218f4aae4f395b3b4cef261572556bbd20a Mon Sep 17 00:00:00 2001 From: Maurus Cuelenaere Date: Tue, 6 Apr 2010 18:12:52 +0200 Subject: [PATCH 01/18] ALSA: hda: Add support for Medion WIM2160 This adds support for the Medion WIM2160 soundcard. There's no PCI quirk added because it has the same PCI id as the Medion MD2. Signed-off-by: Maurus Cuelenaere Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 53 +++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7730dbb9ddb..2971e48e50ad 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -230,6 +230,7 @@ enum { ALC888_ACER_ASPIRE_7730G, ALC883_MEDION, ALC883_MEDION_MD2, + ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, @@ -8455,6 +8456,42 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x08, 0x0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc883_medion_wim2160_verbs[] = { + /* Unmute front mixer */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* Set speaker pin to front mixer */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Init headphone pin */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + + { } /* end */ +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc883_medion_wim2160_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x1a; + spec->autocfg.speaker_pins[0] = 0x15; +} + static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -9164,6 +9201,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC888_ACER_ASPIRE_7730G] = "acer-aspire-7730g", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", + [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", @@ -9818,6 +9856,21 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc883_medion_md2_setup, .init_hook = alc_automute_amp, }, + [ALC883_MEDION_WIM2160] = { + .mixers = { alc883_medion_wim2160_mixer }, + .init_verbs = { alc883_init_verbs, alc883_medion_wim2160_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc_automute_amp_unsol_event, + .setup = alc883_medion_wim2160_setup, + .init_hook = alc_automute_amp, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, From 78e4fd26ef8b85c8cbb6803e18b6b1f970420e06 Mon Sep 17 00:00:00 2001 From: Huang Weiyi Date: Thu, 8 Apr 2010 19:50:08 +0800 Subject: [PATCH 02/18] ASoC: wm2000: remove unused #include Remove unused #include ('s) in sound/soc/codecs/wm2000.c Signed-off-by: Huang Weiyi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 217b02680597..8de866618bf4 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -23,7 +23,6 @@ #include #include -#include #include #include #include From 206b60e189c7cc2b4675687d66f167299a13a4d4 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 8 Apr 2010 11:31:24 +0200 Subject: [PATCH 03/18] ASoC: imx-ssi: honor IMX_SSI_DMA flag When checking if we are DMA capable we have to check for the IMX_SSI_DMA flag which is already set from platform_data instead of setting it again when we want to do DMA. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-ssi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 28e55c7b14b4..1bf9dc88babf 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -655,7 +655,8 @@ static int imx_ssi_probe(struct platform_device *pdev) dai->private_data = ssi; if ((cpu_is_mx27() || cpu_is_mx21()) && - !(ssi->flags & IMX_SSI_USE_AC97)) { + !(ssi->flags & IMX_SSI_USE_AC97) && + (ssi->flags & IMX_SSI_DMA)) { ssi->flags |= IMX_SSI_DMA; platform = imx_ssi_dma_mx2_init(pdev, ssi); } else From 671999cb5d8817611f865f3877f5a5b81372f61e Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 8 Apr 2010 11:31:25 +0200 Subject: [PATCH 04/18] ASoC: imx-pcm-dma-mx2: restart DMA after an error Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index c78c000e2afe..93272966b848 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -70,7 +70,12 @@ static void imx_ssi_dma_callback(int channel, void *data) static void snd_imx_dma_err_callback(int channel, void *data, int err) { - pr_err("DMA error callback called\n"); + struct snd_pcm_substream *substream = data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int ret; pr_err("DMA timeout on channel %d -%s%s%s%s\n", channel, @@ -78,6 +83,14 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err) err & IMX_DMA_ERR_REQUEST ? " request" : "", err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); + + imx_dma_disable(iprtd->dma); + ret = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, + IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_MODE_WRITE : DMA_MODE_READ); + if (!ret) + imx_dma_enable(iprtd->dma); } static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) From 43a3cec01354573517f1348383e0ab6e6067076b Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Thu, 8 Apr 2010 11:31:26 +0200 Subject: [PATCH 05/18] ASoC: imx-ssi: Use a hrtimer in FIQ mode Using a regular timer results in poll times < 1 jiffie with small buffers, so we loaded the timer with the actual jiffie value. We can be more accurate using a hrtimer. Also, we have to call snd_pcm_period_elapsed after playing period_bytes and not runtime->period_size (which is in samples and not in bytes). Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 45 +++++++++++++++++-------------------- 1 file changed, 21 insertions(+), 24 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index d9cb9849b033..64df813b9af8 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -38,20 +38,17 @@ struct imx_pcm_runtime_data { unsigned long offset; unsigned long last_offset; unsigned long size; - struct timer_list timer; - int poll_time; + struct hrtimer hrt; + int poll_time_ns; + struct snd_pcm_substream *substream; }; -static inline void imx_ssi_set_next_poll(struct imx_pcm_runtime_data *iprtd) +static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { - iprtd->timer.expires = jiffies + iprtd->poll_time; -} - -static void imx_ssi_timer_callback(unsigned long data) -{ - struct snd_pcm_substream *substream = (void *)data; + struct imx_pcm_runtime_data *iprtd = + container_of(hrt, struct imx_pcm_runtime_data, hrt); + struct snd_pcm_substream *substream = iprtd->substream; struct snd_pcm_runtime *runtime = substream->runtime; - struct imx_pcm_runtime_data *iprtd = runtime->private_data; struct pt_regs regs; unsigned long delta; @@ -71,16 +68,14 @@ static void imx_ssi_timer_callback(unsigned long data) /* If we've transferred at least a period then report it and * reset our poll time */ - if (delta >= runtime->period_size) { + if (delta >= iprtd->period) { snd_pcm_period_elapsed(substream); iprtd->last_offset = iprtd->offset; - - imx_ssi_set_next_poll(iprtd); } - /* Restart the timer; if we didn't report we'll run on the next tick */ - add_timer(&iprtd->timer); + hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); + return HRTIMER_RESTART; } static struct fiq_handler fh = { @@ -98,8 +93,8 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, iprtd->period = params_period_bytes(params) ; iprtd->offset = 0; iprtd->last_offset = 0; - iprtd->poll_time = HZ / (params_rate(params) / params_period_size(params)); - + iprtd->poll_time_ns = 1000000000 / params_rate(params) * + params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); return 0; @@ -134,8 +129,8 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - imx_ssi_set_next_poll(iprtd); - add_timer(&iprtd->timer); + hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), + HRTIMER_MODE_REL); if (++fiq_enable == 1) enable_fiq(imx_pcm_fiq); @@ -144,7 +139,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - del_timer(&iprtd->timer); + hrtimer_cancel(&iprtd->hrt); if (--fiq_enable == 0) disable_fiq(imx_pcm_fiq); @@ -193,9 +188,10 @@ static int snd_imx_open(struct snd_pcm_substream *substream) iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); runtime->private_data = iprtd; - init_timer(&iprtd->timer); - iprtd->timer.data = (unsigned long)substream; - iprtd->timer.function = imx_ssi_timer_callback; + iprtd->substream = substream; + + hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + iprtd->hrt.function = snd_hrtimer_callback; ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -211,7 +207,8 @@ static int snd_imx_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - del_timer_sync(&iprtd->timer); + hrtimer_cancel(&iprtd->hrt); + kfree(iprtd); return 0; From 531d8791accf1464bc6854ff69d08dd866189d17 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 9 Apr 2010 10:57:33 +0200 Subject: [PATCH 06/18] ALSA: hda - Fix auto-parser of ALC269vb for HP pin NID 0x21 ALC269vb has an alternative HP pin 0x21 in addition. Fix the parser to recognize it. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2971e48e50ad..fbbdfbc8a1ca 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12869,6 +12869,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, dac = 0x02; break; case 0x15: + case 0x21: /* ALC269vb has this pin, too */ dac = 0x03; break; default: From 226b1ec8c18bcb6d1aa448a29b2c8aeae1946228 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Fri, 9 Apr 2010 11:01:20 +0200 Subject: [PATCH 07/18] ALSA: hda - Fix setup for ALC269vb amic and dmic models Corrected HP and mic pins for ALC269vb amic and dmic models. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++------------- 1 file changed, 27 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fbbdfbc8a1ca..9b58f29833e6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13789,6 +13789,18 @@ static void alc269_laptop_unsol_event(struct hda_codec *codec, } } +static void alc269_laptop_amic_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x19; + spec->int_mic.mux_idx = 1; + spec->auto_mic = 1; +} + static void alc269_laptop_dmic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -13801,22 +13813,10 @@ static void alc269_laptop_dmic_setup(struct hda_codec *codec) spec->auto_mic = 1; } -static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +static void alc269vb_laptop_amic_setup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; - spec->autocfg.speaker_pins[0] = 0x14; - spec->ext_mic.pin = 0x18; - spec->ext_mic.mux_idx = 0; - spec->int_mic.pin = 0x12; - spec->int_mic.mux_idx = 6; - spec->auto_mic = 1; -} - -static void alc269_laptop_amic_setup(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -13825,6 +13825,18 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec) spec->auto_mic = 1; } +static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + spec->autocfg.hp_pins[0] = 0x21; + spec->autocfg.speaker_pins[0] = 0x14; + spec->ext_mic.pin = 0x18; + spec->ext_mic.mux_idx = 0; + spec->int_mic.pin = 0x12; + spec->int_mic.mux_idx = 6; + spec->auto_mic = 1; +} + static void alc269_laptop_inithook(struct hda_codec *codec) { alc269_speaker_automute(codec); @@ -14162,7 +14174,7 @@ static struct alc_config_preset alc269_presets[] = { .num_channel_mode = ARRAY_SIZE(alc269_modes), .channel_mode = alc269_modes, .unsol_event = alc269_laptop_unsol_event, - .setup = alc269_laptop_amic_setup, + .setup = alc269vb_laptop_amic_setup, .init_hook = alc269_laptop_inithook, }, [ALC269VB_DMIC] = { From 7f311a46916a3be00a1a8e3f1bdf461d08f1d263 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Apr 2010 17:32:23 +0200 Subject: [PATCH 08/18] ALSA: hda - Fix initial capture source connections of ALC880/260 The widget connections of ADC of ALC880 and ALC2260 aren't initialized, thus it might point to invalid pin. This can be a problem when mode=auto and there is only one input pin. Then user can't change the connection at all. This patch adds the code to initialize the input pin connection of these codecs. Reference: Novell bnc#594363 https://bugzilla.novell.com/show_bug.cgi?id=594363 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b58f29833e6..8d60b1f25ce1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4809,6 +4809,25 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) } } +static void alc880_auto_init_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int c; + + for (c = 0; c < spec->num_adc_nids; c++) { + unsigned int mux_idx; + const struct hda_input_mux *imux; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; + if (!imux->num_items && mux_idx > 0) + imux = &spec->input_mux[0]; + if (imux) + snd_hda_codec_write(codec, spec->adc_nids[c], 0, + AC_VERB_SET_CONNECT_SEL, + imux->items[0].index); + } +} + /* parse the BIOS configuration and set up the alc_spec */ /* return 1 if successful, 0 if the proper config is not found, * or a negative error code @@ -4887,6 +4906,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); + alc880_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } @@ -6398,6 +6418,8 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) } } +#define alc260_auto_init_input_src alc880_auto_init_input_src + /* * generic initialization of ADC, input mixers and output mixers */ @@ -6484,6 +6506,7 @@ static void alc260_auto_init(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); + alc260_auto_init_input_src(codec); if (spec->unsol_event) alc_inithook(codec); } From 29aac005ff4dc8a5f50b80f4e5c4f59b21c0fb50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 10 Apr 2010 21:27:23 +0200 Subject: [PATCH 09/18] ALSA: usb - Fix Oops after usb-midi disconnection usb-midi causes sometimes Oops at snd_usbmidi_output_drain() after disconnection. This is due to the access to the endpoints which have been already released at disconnection while the files are still alive. This patch fixes the problem by checking disconnection state at snd_usbmidi_output_drain() and by releasing urbs but keeping the endpoint instances until really all freed. Tested-by: Tvrtko Ursulin Cc: Signed-off-by: Takashi Iwai --- sound/usb/usbmidi.c | 24 ++++++++++++++++++------ 1 file changed, 18 insertions(+), 6 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 2c59afd99611..9e28b20cb2ce 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -986,6 +986,8 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) DEFINE_WAIT(wait); long timeout = msecs_to_jiffies(50); + if (ep->umidi->disconnected) + return; /* * The substream buffer is empty, but some data might still be in the * currently active URBs, so we have to wait for those to complete. @@ -1123,14 +1125,21 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, * Frees an output endpoint. * May be called when ep hasn't been initialized completely. */ -static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint* ep) +static void snd_usbmidi_out_endpoint_clear(struct snd_usb_midi_out_endpoint *ep) { unsigned int i; for (i = 0; i < OUTPUT_URBS; ++i) - if (ep->urbs[i].urb) + if (ep->urbs[i].urb) { free_urb_and_buffer(ep->umidi, ep->urbs[i].urb, ep->max_transfer); + ep->urbs[i].urb = NULL; + } +} + +static void snd_usbmidi_out_endpoint_delete(struct snd_usb_midi_out_endpoint *ep) +{ + snd_usbmidi_out_endpoint_clear(ep); kfree(ep); } @@ -1262,15 +1271,18 @@ void snd_usbmidi_disconnect(struct list_head* p) usb_kill_urb(ep->out->urbs[j].urb); if (umidi->usb_protocol_ops->finish_out_endpoint) umidi->usb_protocol_ops->finish_out_endpoint(ep->out); + ep->out->active_urbs = 0; + if (ep->out->drain_urbs) { + ep->out->drain_urbs = 0; + wake_up(&ep->out->drain_wait); + } } if (ep->in) for (j = 0; j < INPUT_URBS; ++j) usb_kill_urb(ep->in->urbs[j]); /* free endpoints here; later call can result in Oops */ - if (ep->out) { - snd_usbmidi_out_endpoint_delete(ep->out); - ep->out = NULL; - } + if (ep->out) + snd_usbmidi_out_endpoint_clear(ep->out); if (ep->in) { snd_usbmidi_in_endpoint_delete(ep->in); ep->in = NULL; From 7fa90e873f520dad5ec58f47340996cda083e875 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Apr 2010 08:49:00 +0200 Subject: [PATCH 10/18] ALSA: hda - Enhance fix-up table for Realtek codecs A few enhancement / fixes for fix-up table of some Realtek codecs: - Apply fix-ups only for the auto model - Apply additional verbs after normal init verbs - Add a debug print to show the fix-up application This is basically a preliminary work for the next fix for Sony VAIO. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 35 ++++++++++++++++++++++++++++------- 1 file changed, 28 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8d60b1f25ce1..cff57710d1fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1390,22 +1390,31 @@ struct alc_fixup { static void alc_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, - const struct alc_fixup *fix) + const struct alc_fixup *fix, + int pre_init) { const struct alc_pincfg *cfg; quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); if (!quirk) return; - fix += quirk->value; cfg = fix->pins; - if (cfg) { + if (pre_init && cfg) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply pincfg for %s\n", + codec->chip_name, quirk->name); +#endif for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - if (fix->verbs) + if (!pre_init && fix->verbs) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + snd_printdd(KERN_INFO "hda_codec: %s: Apply fix-verbs for %s\n", + codec->chip_name, quirk->name); +#endif add_verb(codec->spec, fix->verbs); + } } static int alc_read_coef_idx(struct hda_codec *codec, @@ -10439,7 +10448,8 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 1); if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ @@ -10512,6 +10522,9 @@ static int patch_alc882(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + if (board_config == ALC882_AUTO) + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups, 0); + spec->vmaster_nid = 0x0c; codec->patch_ops = alc_patch_ops; @@ -15417,7 +15430,8 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } - alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups); + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 1); if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ @@ -15454,6 +15468,9 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; + if (board_config == ALC861_AUTO) + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) { spec->init_hook = alc861_auto_init; @@ -16388,7 +16405,8 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } - alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups); + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 1); if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ @@ -16436,6 +16454,9 @@ static int patch_alc861vd(struct hda_codec *codec) spec->vmaster_nid = 0x02; + if (board_config == ALC861VD_AUTO) + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups, 0); + codec->patch_ops = alc_patch_ops; if (board_config == ALC861VD_AUTO) From ff818c24c2af370153646d302d831b69b023816f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Apr 2010 08:59:25 +0200 Subject: [PATCH 11/18] ALSA: hda - Add fix-up for Sony VAIO with ALC269 Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF ground or Hi-Z to make the headphone working. Other than that, model=auto works fine, so let's use model=auto with a specific fix-up table. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cff57710d1fb..4b35176d3454 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14077,6 +14077,27 @@ static void alc269_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC269_FIXUP_SONY_VAIO, +}; + +const static struct hda_verb alc269_sony_vaio_fixup_verbs[] = { + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREFGRD}, + {} +}; + +static const struct alc_fixup alc269_fixups[] = { + [ALC269_FIXUP_SONY_VAIO] = { + .verbs = alc269_sony_vaio_fixup_verbs + }, +}; + +static struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + {} +}; + + /* * configuration and preset */ @@ -14136,7 +14157,7 @@ static struct snd_pci_quirk alc269_cfg_tbl[] = { ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005HA", ALC269_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005HA", ALC269_DMIC), - SND_PCI_QUIRK(0x104d, 0x9071, "SONY XTB", ALC269_DMIC), + SND_PCI_QUIRK(0x104d, 0x9071, "Sony VAIO", ALC269_AUTO), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), SND_PCI_QUIRK(0x152d, 0x1778, "Quanta ON1", ALC269_DMIC), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), @@ -14290,6 +14311,9 @@ static int patch_alc269(struct hda_codec *codec) board_config = ALC269_AUTO; } + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 1); + if (board_config == ALC269_AUTO) { /* automatic parse from the BIOS config */ err = alc269_parse_auto_config(codec); @@ -14342,6 +14366,9 @@ static int patch_alc269(struct hda_codec *codec) set_capture_mixer(codec); set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); + if (board_config == ALC269_AUTO) + alc_pick_fixup(codec, alc269_fixup_tbl, alc269_fixups, 0); + spec->vmaster_nid = 0x02; codec->patch_ops = alc_patch_ops; From b68b58fd6a341c2115ff5fb466fe9fc0b581980e Mon Sep 17 00:00:00 2001 From: Philby John Date: Fri, 26 Mar 2010 21:37:51 +0530 Subject: [PATCH 12/18] ALSA: aaci - Fix alignment faults on ARM Cortex introduced by commit 29a4f2d3 The commit 29a4f2d3 used writel() at offset 0x26 which is half-word aligned causing unaligned exceptions on a Cortex-A8. The original patch solved the "aaci-pl041 fpga:04: ac97 read back fail" issue on a soft reset. Reading from any arbitrary aaci register seems to solve this issue. Signed-off-by: Philby John Acked-by: Russell King Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 656e474dca47..91acc9a243ec 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -863,7 +863,6 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; - writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ @@ -1047,7 +1046,11 @@ static int __devinit aaci_probe(struct amba_device *dev, struct amba_id *id) writel(0x1fff, aaci->base + AACI_INTCLR); writel(aaci->maincr, aaci->base + AACI_MAINCR); - + /* + * Fix: ac97 read back fail errors by reading + * from any arbitrary aaci register. + */ + readl(aaci->base + AACI_CSCH1); ret = aaci_probe_ac97(aaci); if (ret) goto out; From b331439dfd41dc813b3557ca5927a3a644f35792 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Apr 2010 14:33:57 +0200 Subject: [PATCH 13/18] ALSA: hda - Fix control element allocations in VIA codec parser The commit 5b0cb1d850c26893b1468b3a519433a1b7a176be ALSA: hda - add more NID->Control mapping breaks the control element allocation by returning a wrong value. Let's fix it. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 9ddc37300f6b..be1295438989 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -476,7 +476,7 @@ static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, knew->name = kstrdup(tmpl->name, GFP_KERNEL); if (!knew->name) return NULL; - return 0; + return knew; } static void via_free_kctls(struct hda_codec *codec) From 3d83e577a8206f0f3822a3840e12f76477142ba2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Apr 2010 14:36:23 +0200 Subject: [PATCH 14/18] ALSA: hda - Avoid invalid "Independent HP" control for VIA codecs Some VIA codecs have no multiple source selection for headphone pins, thus it's useless (and wrong) to create "Independent HP" control on them. This patch adds the check of connections to skip the control in such a case. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 39 +++++++++++++++++++++++---------------- 1 file changed, 23 insertions(+), 16 deletions(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index be1295438989..73453814e098 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1215,14 +1215,13 @@ static struct snd_kcontrol_new via_hp_mixer[2] = { }, }; -static int via_hp_build(struct via_spec *spec) +static int via_hp_build(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; struct snd_kcontrol_new *knew; hda_nid_t nid; - - knew = via_clone_control(spec, &via_hp_mixer[0]); - if (knew == NULL) - return -ENOMEM; + int nums; + hda_nid_t conn[HDA_MAX_CONNECTIONS]; switch (spec->codec_type) { case VT1718S: @@ -1239,6 +1238,14 @@ static int via_hp_build(struct via_spec *spec) break; } + nums = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); + if (nums <= 1) + return 0; + + knew = via_clone_control(spec, &via_hp_mixer[0]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; knew->private_value = nid; @@ -2561,7 +2568,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -3087,7 +3094,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -3654,7 +3661,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -4140,7 +4147,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); return 1; @@ -4510,7 +4517,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } @@ -4930,7 +4937,7 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); @@ -5425,7 +5432,7 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); via_smart51_build(spec); @@ -5781,7 +5788,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } @@ -6000,12 +6007,12 @@ static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, /* Line-Out: PortE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, - "Master Front Playback Volume", + "Front Playback Volume", HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, - "Master Front Playback Switch", + "Front Playback Switch", HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -6130,7 +6137,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - via_hp_build(spec); + via_hp_build(codec); return 1; } From 565a79f74af96ae90dfec411da14dc38d2cd56bc Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 14 Apr 2010 09:17:31 +0200 Subject: [PATCH 15/18] ASoC: imx-ssi: increase minimum periods to 4 Currently the notification of elapsed periods is not very exact. Increase minimum periods to 4 as suggested by Liam Girdwood. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 64df813b9af8..98ab33109527 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -174,7 +174,7 @@ static struct snd_pcm_hardware snd_imx_hardware = { .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 16 * 1024, - .periods_min = 2, + .periods_min = 4, .periods_max = 255, .fifo_size = 0, }; From d1501ea844eefdf925f6b711875b4b2b928fddf8 Mon Sep 17 00:00:00 2001 From: Joerg Schirottke Date: Thu, 15 Apr 2010 08:37:41 +0200 Subject: [PATCH 16/18] ALSA: hda - add a quirk for Clevo M570U laptop Added the matching model for Clevo laptop M570U. Signed-off-by: Joerg Schirottke Tested-by: Maximilian Gerhard Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4b35176d3454..aad1627f56f1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9350,6 +9350,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1558, 0x0571, "Clevo laptop M570U", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), From 8815cd030fdd73932a791d1f06194c8db807cde7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 15 Apr 2010 09:02:41 +0200 Subject: [PATCH 17/18] ALSA: hda - Add position_fix quirk for Biostar mobo The Biostar mobo seems to give a wrong DMA position, resulting in stuttering or skipping sounds on 2.6.34. Since the commit 7b3a177b0d4f92b3431b8dca777313a07533a710, "ALSA: pcm_lib: fix "something must be really wrong" condition", makes the position check more strictly, the DMA position problem is revealed more clearly now. The fix is to use only LPIB for obtaining the position, i.e. passing position_fix=1. This patch adds a static quirk to achieve it as default. Reported-by: Frank Griffin Cc: Eric Piel Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f8fd586ae024..f669442b7c82 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2272,6 +2272,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), {} }; From 8392609969b3b37a4da5cff08161661f7a8c16af Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Wed, 14 Apr 2010 09:17:30 +0200 Subject: [PATCH 18/18] ASoC: imx-ssi: do not call hrtimer_disable in trigger function Doing so causes a deadlock, so just signal the timer to stop using an atomic variable. Signed-off-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-fiq.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 98ab33109527..ecec332121f2 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -41,6 +41,7 @@ struct imx_pcm_runtime_data { struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; + atomic_t running; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) @@ -52,6 +53,9 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct pt_regs regs; unsigned long delta; + if (!atomic_read(&iprtd->running)) + return HRTIMER_NORESTART; + get_fiq_regs(®s); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -129,6 +133,7 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + atomic_set(&iprtd->running, 1); hrtimer_start(&iprtd->hrt, ns_to_ktime(iprtd->poll_time_ns), HRTIMER_MODE_REL); if (++fiq_enable == 1) @@ -139,11 +144,11 @@ static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - hrtimer_cancel(&iprtd->hrt); + atomic_set(&iprtd->running, 0); + if (--fiq_enable == 0) disable_fiq(imx_pcm_fiq); - break; default: return -EINVAL; @@ -190,6 +195,7 @@ static int snd_imx_open(struct snd_pcm_substream *substream) iprtd->substream = substream; + atomic_set(&iprtd->running, 0); hrtimer_init(&iprtd->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); iprtd->hrt.function = snd_hrtimer_callback;