From a5ba6beb839cfa288960c92cd2668a2601c24dda Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Aug 2010 08:08:48 +0200 Subject: ALSA: riptide - Fix detection / load of firmware files The detection and loading of firmeware on riptide driver has been broken due to rewrite of some codes, checking the presense wrongly. This patch fixes the logic again. Reference: kernel bug 16596 https://bugzilla.kernel.org/show_bug.cgi?id=16596 Cc: Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index f64fb7d988cb..ad5202efd7a9 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1224,15 +1224,14 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) firmware.firmware.ASIC, firmware.firmware.CODEC, firmware.firmware.AUXDSP, firmware.firmware.PROG); + if (!chip) + return 1; + for (i = 0; i < FIRMWARE_VERSIONS; i++) { if (!memcmp(&firmware_versions[i], &firmware, sizeof(firmware))) - break; - } - if (i >= FIRMWARE_VERSIONS) - return 0; /* no match */ + return 1; /* OK */ - if (!chip) - return 1; /* OK */ + } snd_printdd("Writing Firmware\n"); if (!chip->fw_entry) { -- cgit v1.2.1 From c3e68fad88143fd1fe8fe640207fb19c0f087dbc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Aug 2010 10:15:57 +0200 Subject: ALSA: hda - Add quirk for Dell Vostro 1220 model=dell-vostro is needed for Dell Vostro 1220 with Coexnat 5067. Reference: Novell bnc#631066 https://bugzilla.novell.com/show_bug.cgi?id=631066 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 31b5d9eeba68..c424952a734e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3049,6 +3049,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), -- cgit v1.2.1 From b2c1e07b81a126e5846dfc3d36f559d861df59f4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 16 Aug 2010 11:46:57 +0100 Subject: ASoC: Remove DSP mode support for WM8776 This is not supported by current hardware revisions. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8776.c | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 4e212ed62ea6..f8154e661524 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -178,13 +178,6 @@ static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_LEFT_J: iface |= 0x0001; break; - /* FIXME: CHECK A/B */ - case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; - break; - case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0007; - break; default: return -EINVAL; } -- cgit v1.2.1 From c69aefabe004d24e6eedf83b6f253647f77dfc43 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 17 Aug 2010 10:39:22 +0200 Subject: ALSA: hda - Fix ALC680 base model capture - Fix capture mixer elements for ALC680 base model - Support auto change ADC for recording from MIC - Cancel capture source assigned in auto mode. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 176 ++++++++++++++++++++++++++++++++++-------- 1 file changed, 144 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2cd1ae809e46..a4dd04524e43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -19030,6 +19030,7 @@ static int patch_alc888(struct hda_codec *codec) /* * ALC680 support */ +#define ALC680_DIGIN_NID ALC880_DIGIN_NID #define ALC680_DIGOUT_NID ALC880_DIGOUT_NID #define alc680_modes alc260_modes @@ -19044,23 +19045,93 @@ static hda_nid_t alc680_adc_nids[3] = { 0x07, 0x08, 0x09 }; +/* + * Analog capture ADC cgange + */ +static int alc680_capture_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int pre_mic, pre_line; + + pre_mic = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); + pre_line = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_LINE]); + + spec->cur_adc_stream_tag = stream_tag; + spec->cur_adc_format = format; + + if (pre_mic || pre_line) { + if (pre_mic) + snd_hda_codec_setup_stream(codec, 0x08, stream_tag, 0, + format); + else + snd_hda_codec_setup_stream(codec, 0x09, stream_tag, 0, + format); + } else + snd_hda_codec_setup_stream(codec, 0x07, stream_tag, 0, format); + return 0; +} + +static int alc680_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_cleanup_stream(codec, 0x07); + snd_hda_codec_cleanup_stream(codec, 0x08); + snd_hda_codec_cleanup_stream(codec, 0x09); + return 0; +} + +static struct hda_pcm_stream alc680_pcm_analog_auto_capture = { + .substreams = 1, /* can be overridden */ + .channels_min = 2, + .channels_max = 2, + /* NID is set in alc_build_pcms */ + .ops = { + .prepare = alc680_capture_pcm_prepare, + .cleanup = alc680_capture_pcm_cleanup + }, +}; + static struct snd_kcontrol_new alc680_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Headphone Playback Volume", 0x4, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x12, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Boost", 0x19, 0, HDA_INPUT), { } }; -static struct snd_kcontrol_new alc680_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT), +static struct hda_bind_ctls alc680_bind_cap_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc680_bind_cap_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x07, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc680_master_capture_mixer[] = { + HDA_BIND_VOL("Capture Volume", &alc680_bind_cap_vol), + HDA_BIND_SW("Capture Switch", &alc680_bind_cap_switch), { } /* end */ }; @@ -19068,25 +19139,73 @@ static struct snd_kcontrol_new alc680_capture_mixer[] = { * generic initialization of ADC, input mixers and output mixers */ static struct hda_verb alc680_init_verbs[] = { - /* Unmute DAC0-1 and set vol = 0 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + { } }; +/* toggle speaker-output according to the hp-jack state */ +static void alc680_base_setup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x16; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x15; + spec->autocfg.input_pins[AUTO_PIN_MIC] = 0x18; + spec->autocfg.input_pins[AUTO_PIN_LINE] = 0x19; +} + +static void alc680_rec_autoswitch(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int present; + hda_nid_t new_adc; + + present = snd_hda_jack_detect(codec, cfg->input_pins[AUTO_PIN_MIC]); + + new_adc = present ? 0x8 : 0x7; + __snd_hda_codec_cleanup_stream(codec, !present ? 0x8 : 0x7, 1); + snd_hda_codec_setup_stream(codec, new_adc, + spec->cur_adc_stream_tag, 0, + spec->cur_adc_format); + +} + +static void alc680_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc_automute_amp(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc680_rec_autoswitch(codec); +} + +static void alc680_inithook(struct hda_codec *codec) +{ + alc_automute_amp(codec); + alc680_rec_autoswitch(codec); +} + /* create input playback/capture controls for the given pin */ static int alc680_new_analog_output(struct alc_spec *spec, hda_nid_t nid, const char *ctlname, int idx) @@ -19197,13 +19316,7 @@ static void alc680_auto_init_hp_out(struct hda_codec *codec) #define alc680_pcm_analog_capture alc880_pcm_analog_capture #define alc680_pcm_analog_alt_capture alc880_pcm_analog_alt_capture #define alc680_pcm_digital_playback alc880_pcm_digital_playback - -static struct hda_input_mux alc680_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x0 }, - }, -}; +#define alc680_pcm_digital_capture alc880_pcm_digital_capture /* * BIOS auto configuration @@ -19218,6 +19331,7 @@ static int alc680_parse_auto_config(struct hda_codec *codec) alc680_ignore); if (err < 0) return err; + if (!spec->autocfg.line_outs) { if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { spec->multiout.max_channels = 2; @@ -19239,8 +19353,6 @@ static int alc680_parse_auto_config(struct hda_codec *codec) add_mixer(spec, spec->kctls.list); add_verb(spec, alc680_init_verbs); - spec->num_mux_defs = 1; - spec->input_mux = &alc680_capture_source; err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -19279,17 +19391,17 @@ static struct snd_pci_quirk alc680_cfg_tbl[] = { static struct alc_config_preset alc680_presets[] = { [ALC680_BASE] = { .mixers = { alc680_base_mixer }, - .cap_mixer = alc680_capture_mixer, + .cap_mixer = alc680_master_capture_mixer, .init_verbs = { alc680_init_verbs }, .num_dacs = ARRAY_SIZE(alc680_dac_nids), .dac_nids = alc680_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc680_adc_nids), - .adc_nids = alc680_adc_nids, - .hp_nid = 0x04, .dig_out_nid = ALC680_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc680_modes), .channel_mode = alc680_modes, - .input_mux = &alc680_capture_source, + .unsol_event = alc680_unsol_event, + .setup = alc680_base_setup, + .init_hook = alc680_inithook, + }, }; @@ -19333,9 +19445,9 @@ static int patch_alc680(struct hda_codec *codec) setup_preset(codec, &alc680_presets[board_config]); spec->stream_analog_playback = &alc680_pcm_analog_playback; - spec->stream_analog_capture = &alc680_pcm_analog_capture; - spec->stream_analog_alt_capture = &alc680_pcm_analog_alt_capture; + spec->stream_analog_capture = &alc680_pcm_analog_auto_capture; spec->stream_digital_playback = &alc680_pcm_digital_playback; + spec->stream_digital_capture = &alc680_pcm_digital_capture; if (!spec->adc_nids) { spec->adc_nids = alc680_adc_nids; -- cgit v1.2.1 From 56385a12d9bb9e173751f74b6c430742018cafc0 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 18 Aug 2010 14:08:17 +0200 Subject: ALSA: emu10k1 - delay the PCM interrupts (add pcm_irq_delay parameter) With some hardware combinations, the PCM interrupts are acknowledged before the period boundary from the emu10k1 chip. The midlevel PCM code gets confused and the playback stream is interrupted. It seems that the interrupt processing shift by 2 samples is enough to fix this issue. This default value does not harm other, non-affected hardware. More information: Kernel bugzilla bug#16300 [A copmile warning fixed by tiwai] Signed-off-by: Jaroslav Kysela Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 4 ++++ sound/pci/emu10k1/emu10k1.c | 4 ++++ sound/pci/emu10k1/emupcm.c | 30 ++++++++++++++++++++++++++---- sound/pci/emu10k1/memory.c | 4 +++- 4 files changed, 37 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a3b2a6479246..134fc6c2e08d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -978,6 +978,10 @@ static int snd_pcm_do_pause(struct snd_pcm_substream *substream, int push) { if (substream->runtime->trigger_master != substream) return 0; + /* some drivers might use hw_ptr to recover from the pause - + update the hw_ptr now */ + if (push) + snd_pcm_update_hw_ptr(substream); /* The jiffies check in snd_pcm_update_hw_ptr*() is done by * a delta betwen the current jiffies, this gives a large enough * delta, effectively to skip the check once. diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 4203782d7cb7..aff8387c45cf 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -52,6 +52,7 @@ static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64}; static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128}; static int enable_ir[SNDRV_CARDS]; static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ +static uint delay_pcm_irq[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard."); @@ -73,6 +74,8 @@ module_param_array(enable_ir, bool, NULL, 0444); MODULE_PARM_DESC(enable_ir, "Enable IR."); module_param_array(subsystem, uint, NULL, 0444); MODULE_PARM_DESC(subsystem, "Force card subsystem model."); +module_param_array(delay_pcm_irq, uint, NULL, 0444); +MODULE_PARM_DESC(delay_pcm_irq, "Delay PCM interrupt by specified number of samples (default 0)."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ @@ -127,6 +130,7 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci, &emu)) < 0) goto error; card->private_data = emu; + emu->delay_pcm_irq = delay_pcm_irq[dev] & 0x1f; if ((err = snd_emu10k1_pcm(emu, 0, NULL)) < 0) goto error; if ((err = snd_emu10k1_pcm_mic(emu, 1, NULL)) < 0) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 55b83ef73c63..622bace148e3 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -332,7 +332,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, evoice->epcm->ccca_start_addr = start_addr + ccis; if (extra) { start_addr += ccis; - end_addr += ccis; + end_addr += ccis + emu->delay_pcm_irq; } if (stereo && !extra) { snd_emu10k1_ptr_write(emu, CPF, voice, CPF_STEREO_MASK); @@ -360,7 +360,9 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, /* Assumption that PT is already 0 so no harm overwriting */ snd_emu10k1_ptr_write(emu, PTRX, voice, (send_amount[0] << 8) | send_amount[1]); snd_emu10k1_ptr_write(emu, DSL, voice, end_addr | (send_amount[3] << 24)); - snd_emu10k1_ptr_write(emu, PSST, voice, start_addr | (send_amount[2] << 24)); + snd_emu10k1_ptr_write(emu, PSST, voice, + (start_addr + (extra ? emu->delay_pcm_irq : 0)) | + (send_amount[2] << 24)); if (emu->card_capabilities->emu_model) pitch_target = PITCH_48000; /* Disable interpolators on emu1010 card */ else @@ -732,6 +734,23 @@ static void snd_emu10k1_playback_stop_voice(struct snd_emu10k1 *emu, struct snd_ snd_emu10k1_ptr_write(emu, IP, voice, 0); } +static inline void snd_emu10k1_playback_mangle_extra(struct snd_emu10k1 *emu, + struct snd_emu10k1_pcm *epcm, + struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) +{ + unsigned int ptr, period_pos; + + /* try to sychronize the current position for the interrupt + source voice */ + period_pos = runtime->status->hw_ptr - runtime->hw_ptr_interrupt; + period_pos %= runtime->period_size; + ptr = snd_emu10k1_ptr_read(emu, CCCA, epcm->extra->number); + ptr &= ~0x00ffffff; + ptr |= epcm->ccca_start_addr + period_pos; + snd_emu10k1_ptr_write(emu, CCCA, epcm->extra->number, ptr); +} + static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -753,6 +772,8 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, /* follow thru */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: + if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) + snd_emu10k1_playback_mangle_extra(emu, epcm, substream, runtime); mix = &emu->pcm_mixer[substream->number]; snd_emu10k1_playback_prepare_voice(emu, epcm->voices[0], 1, 0, mix); snd_emu10k1_playback_prepare_voice(emu, epcm->voices[1], 0, 0, mix); @@ -869,8 +890,9 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream * #endif /* printk(KERN_DEBUG - "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", - ptr, runtime->buffer_size, runtime->period_size); + "ptr = 0x%lx, buffer_size = 0x%lx, period_size = 0x%lx\n", + (long)ptr, (long)runtime->buffer_size, + (long)runtime->period_size); */ return ptr; } diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index ffb1ddb8dc28..957a311514c8 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -310,8 +310,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst if (snd_BUG_ON(!hdr)) return NULL; + idx = runtime->period_size >= runtime->buffer_size ? + (emu->delay_pcm_irq * 2) : 0; mutex_lock(&hdr->block_mutex); - blk = search_empty(emu, runtime->dma_bytes); + blk = search_empty(emu, runtime->dma_bytes + idx); if (blk == NULL) { mutex_unlock(&hdr->block_mutex); return NULL; -- cgit v1.2.1