diff options
Diffstat (limited to 'sound')
28 files changed, 234 insertions, 176 deletions
diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 51a3d381a59e..9890cf2066ff 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -1722,7 +1722,6 @@ printk (KERN_INFO "FKS: es_rec_set_recmask mask = %x\n", mask); right = (value & 0x0000ff00) >> 8; } else { /* Turn it off (3) */ left = 0; - left = 0; right = 0; } sb_common_mixer_set(devc, i + ES_REC_MIXER_RECDIFF, left, right); diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index f7e374ec4414..1b9bf9395cfe 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -625,6 +625,8 @@ static short create_adapter_obj(struct hpi_adapter_obj *pao, control_cache_size, (struct hpi_control_cache_info *) &phw->control_cache[0] ); + if (!phw->p_cache) + pao->has_control_cache = 0; } else pao->has_control_cache = 0; diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 22c5fc625533..2672f6591ceb 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -644,6 +644,8 @@ static u16 create_adapter_obj(struct hpi_adapter_obj *pao, interface->control_cache.size_in_bytes, (struct hpi_control_cache_info *) p_control_cache_virtual); + if (!phw->p_cache) + err = HPI_ERROR_MEMORY_ALLOC; } if (!err) { err = hpios_locked_mem_get_phys_addr(&phw-> diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c index dda4f1c6f658..d67f4d3db911 100644 --- a/sound/pci/asihpi/hpicmn.c +++ b/sound/pci/asihpi/hpicmn.c @@ -571,14 +571,20 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32 { struct hpi_control_cache *p_cache = kmalloc(sizeof(*p_cache), GFP_KERNEL); + if (!p_cache) + return NULL; + p_cache->p_info = + kmalloc(sizeof(*p_cache->p_info) * number_of_controls, + GFP_KERNEL); + if (!p_cache->p_info) { + kfree(p_cache); + return NULL; + } p_cache->cache_size_in_bytes = size_in_bytes; p_cache->control_count = number_of_controls; p_cache->p_cache = (struct hpi_control_cache_single *)pDSP_control_buffer; p_cache->init = 0; - p_cache->p_info = - kmalloc(sizeof(*p_cache->p_info) * p_cache->control_count, - GFP_KERNEL); return p_cache; } diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 3e5ca8fb519f..e377287192aa 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -225,39 +225,25 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) { struct dsp_spos_instance * ins = kzalloc(sizeof(struct dsp_spos_instance), GFP_KERNEL); - if (ins == NULL) + if (ins == NULL) return NULL; /* better to use vmalloc for this big table */ - ins->symbol_table.nsymbols = 0; ins->symbol_table.symbols = vmalloc(sizeof(struct dsp_symbol_entry) * DSP_MAX_SYMBOLS); - ins->symbol_table.highest_frag_index = 0; - - if (ins->symbol_table.symbols == NULL) { + ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL); + ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL); + if (!ins->symbol_table.symbols || !ins->code.data || !ins->modules) { cs46xx_dsp_spos_destroy(chip); goto error; } - + ins->symbol_table.nsymbols = 0; + ins->symbol_table.highest_frag_index = 0; ins->code.offset = 0; ins->code.size = 0; - ins->code.data = kmalloc(DSP_CODE_BYTE_SIZE, GFP_KERNEL); - - if (ins->code.data == NULL) { - cs46xx_dsp_spos_destroy(chip); - goto error; - } - ins->nscb = 0; ins->ntask = 0; - ins->nmodules = 0; - ins->modules = kmalloc(sizeof(struct dsp_module_desc) * DSP_MAX_MODULES, GFP_KERNEL); - - if (ins->modules == NULL) { - cs46xx_dsp_spos_destroy(chip); - goto error; - } /* default SPDIF input sample rate to 48000 khz */ @@ -271,8 +257,8 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) /* set left and right validity bits and default channel status */ - ins->spdif_csuv_default = - ins->spdif_csuv_stream = + ins->spdif_csuv_default = + ins->spdif_csuv_stream = /* byte 0 */ ((unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF & 0xff)) << 24) | /* byte 1 */ ((unsigned int)_wrap_all_bits( ((SNDRV_PCM_DEFAULT_CON_SPDIF >> 8) & 0xff)) << 16) | /* byte 3 */ (unsigned int)_wrap_all_bits( (SNDRV_PCM_DEFAULT_CON_SPDIF >> 24) & 0xff) | @@ -281,6 +267,9 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) return ins; error: + kfree(ins->modules); + kfree(ins->code.data); + vfree(ins->symbol_table.symbols); kfree(ins); return NULL; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 460fb2ef7e39..18af38ebf757 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1166,6 +1166,7 @@ static const char *cs420x_models[CS420X_MODELS] = { static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), + SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb89, "MacBookPro 7,1", CS420X_MBP55), SND_PCI_QUIRK(0x8086, 0x7270, "IMac 27 Inch", CS420X_IMAC27), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 82ebeb9544fe..93fa59cc60ef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5326,6 +5326,82 @@ again: return 0; } +static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) +{ + if (codec->vendor_id != 0x111d7605 && + codec->vendor_id != 0x111d76d1) + return 0; + + switch (codec->subsystem_id) { + case 0x103c1618: + case 0x103c1619: + case 0x103c161a: + case 0x103c161b: + case 0x103c161c: + case 0x103c161d: + case 0x103c161e: + case 0x103c161f: + case 0x103c1620: + case 0x103c1621: + case 0x103c1622: + case 0x103c1623: + + case 0x103c162a: + case 0x103c162b: + + case 0x103c1630: + case 0x103c1631: + + case 0x103c1633: + + case 0x103c1635: + + case 0x103c164f: + + case 0x103c1676: + case 0x103c1677: + case 0x103c1678: + case 0x103c1679: + case 0x103c167a: + case 0x103c167b: + case 0x103c167c: + case 0x103c167d: + case 0x103c167e: + case 0x103c167f: + case 0x103c1680: + case 0x103c1681: + case 0x103c1682: + case 0x103c1683: + case 0x103c1684: + case 0x103c1685: + case 0x103c1686: + case 0x103c1687: + case 0x103c1688: + case 0x103c1689: + case 0x103c168a: + case 0x103c168b: + case 0x103c168c: + case 0x103c168d: + case 0x103c168e: + case 0x103c168f: + case 0x103c1690: + case 0x103c1691: + case 0x103c1692: + + case 0x103c3587: + case 0x103c3588: + case 0x103c3589: + case 0x103c358a: + + case 0x103c3667: + case 0x103c3668: + /* set BTL amp level to 13.43dB for louder speaker output */ + return snd_hda_codec_write_cache(codec, codec->afg, 0, + 0x7F4, 0x14); + } + return 0; +} + static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5452,6 +5528,8 @@ again: AC_VERB_SET_CONNECT_SEL, num_dacs); } + stac92hd83xxx_set_system_btl_amp(codec); + codec->proc_widget_hook = stac92hd_proc_hook; return 0; diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index ef9af3f4ace2..1bd7a540fd49 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -425,7 +425,7 @@ exit: static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream) { struct snd_pcm_substream *substream = lx_stream->stream; - const int is_capture = lx_stream->is_capture; + const unsigned int is_capture = lx_stream->is_capture; int err; @@ -473,7 +473,7 @@ static void lx_trigger_start(struct lx6464es *chip, struct lx_stream *lx_stream) static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream) { - const int is_capture = lx_stream->is_capture; + const unsigned int is_capture = lx_stream->is_capture; int err; snd_printd(LXP "stopping: stopping stream\n"); diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index 51afc048961d..aea621eafbb5 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -60,7 +60,7 @@ struct lx_stream { snd_pcm_uframes_t frame_pos; enum lx_stream_status status; /* free, open, running, draining * pause */ - int is_capture:1; + unsigned int is_capture:1; }; diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 3086b751da4a..617f98b0cbae 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -1152,7 +1152,7 @@ static int lx_interrupt_request_new_buffer(struct lx6464es *chip, struct lx_stream *lx_stream) { struct snd_pcm_substream *substream = lx_stream->stream; - int is_capture = lx_stream->is_capture; + const unsigned int is_capture = lx_stream->is_capture; int err; unsigned long flags; diff --git a/sound/sh/aica.c b/sound/sh/aica.c index a0df401ebb9f..94c6ea7fa7c2 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -188,7 +188,7 @@ static void spu_reset(void) spu_memset(0, 0, 0x200000 / 4); /* Put ARM7 in endless loop */ local_irq_save(flags); - ctrl_outl(0xea000002, SPU_MEMORY_BASE); + __raw_writel(0xea000002, SPU_MEMORY_BASE); local_irq_restore(flags); spu_enable(); } diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94a9d06b9027..3b5690d28b8b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,8 +25,9 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C - select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index c53955fe17b6..de799cd1ba72 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -47,7 +47,7 @@ static int ad73311_probe(struct platform_device *pdev) &soc_codec_dev_ad73311, &ad73311_dai, 1); } -static int ad73311_remove(struct platform_device *pdev) +static int __devexit ad73311_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index e7a40d16df90..bc22ee93a75d 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2051,7 +2051,7 @@ static int max98088_i2c_probe(struct i2c_client *i2c, return ret; } -static int max98088_i2c_remove(struct i2c_client *client) +static int __devexit max98088_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); kfree(i2c_get_clientdata(client)); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d251ff54a2d3..c5ab8c805771 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -58,7 +58,7 @@ (1000000000 / ((rate * 1000) / samples)) #define US_TO_SAMPLES(rate, us) \ - (rate / (1000000 / us)) + (rate / (1000000 / (us < 1000000 ? us : 1000000))) #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate))) @@ -200,7 +200,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, u8 *value) { struct tlv320dac33_priv *dac33 = snd_soc_codec_get_drvdata(codec); - int val; + int val, ret = 0; *value = reg & 0xff; @@ -210,6 +210,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, if (val < 0) { dev_err(codec->dev, "Read failed (%d)\n", val); value[0] = dac33_read_reg_cache(codec, reg); + ret = val; } else { value[0] = val; dac33_write_reg_cache(codec, reg, val); @@ -218,7 +219,7 @@ static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, value[0] = dac33_read_reg_cache(codec, reg); } - return 0; + return ret; } static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, @@ -329,13 +330,18 @@ static void dac33_init_chip(struct snd_soc_codec *codec) dac33_read_reg_cache(codec, DAC33_LINER_TO_RLO_VOL)); } -static inline void dac33_read_id(struct snd_soc_codec *codec) +static inline int dac33_read_id(struct snd_soc_codec *codec) { + int i, ret = 0; u8 reg; - dac33_read(codec, DAC33_DEVICE_ID_MSB, ®); - dac33_read(codec, DAC33_DEVICE_ID_LSB, ®); - dac33_read(codec, DAC33_DEVICE_REV_ID, ®); + for (i = 0; i < 3; i++) { + ret = dac33_read(codec, DAC33_DEVICE_ID_MSB + i, ®); + if (ret < 0) + break; + } + + return ret; } static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) @@ -1076,6 +1082,9 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) /* Number of samples under i2c latency */ dac33->alarm_threshold = US_TO_SAMPLES(rate, dac33->mode1_latency); + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - + dac33->alarm_threshold; + if (dac33->auto_fifo_config) { if (period_size <= dac33->alarm_threshold) /* @@ -1086,6 +1095,8 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) ((dac33->alarm_threshold / period_size) + (dac33->alarm_threshold % period_size ? 1 : 0)); + else if (period_size > nsample_limit) + dac33->nsample = nsample_limit; else dac33->nsample = period_size; } else { @@ -1097,8 +1108,7 @@ static void dac33_calculate_times(struct snd_pcm_substream *substream) */ dac33->nsample_max = substream->runtime->buffer_size - period_size; - nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - - dac33->alarm_threshold; + if (dac33->nsample_max > nsample_limit) dac33->nsample_max = nsample_limit; @@ -1414,9 +1424,15 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) dev_err(codec->dev, "Failed to power up codec: %d\n", ret); goto err_power; } - dac33_read_id(codec); + ret = dac33_read_id(codec); dac33_hard_power(codec, 0); + if (ret < 0) { + dev_err(codec->dev, "Failed to read chip ID: %d\n", ret); + ret = -ENODEV; + goto err_power; + } + /* Check if the IRQ number is valid and request it */ if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 329acc1a2074..ee4fb201de60 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -119,13 +119,13 @@ static int tpa6130a2_power(int power) { struct tpa6130a2_data *data; u8 val; - int ret; + int ret = 0; BUG_ON(tpa6130a2_client == NULL); data = i2c_get_clientdata(tpa6130a2_client); mutex_lock(&data->mutex); - if (power) { + if (power && !data->power_state) { /* Power on */ if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); @@ -153,7 +153,7 @@ static int tpa6130a2_power(int power) val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val &= ~TPA6130A2_SWS; tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); - } else { + } else if (!power && data->power_state) { /* set SWS */ val = tpa6130a2_read(TPA6130A2_REG_CONTROL); val |= TPA6130A2_SWS; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index b4f11724a63f..aca4b1ea10bb 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg) { switch (reg) { case WM8900_REG_ID: - case WM8900_REG_POWER1: return 1; default: return 0; @@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec) return -ENODEV; } - /* Read back from the chip */ - reg = snd_soc_read(codec, WM8900_REG_POWER1); - reg = (reg >> 12) & 0xf; - dev_info(codec->dev, "WM8900 revision %d\n", reg); - wm8900_reset(codec); /* Turn the chip on */ diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 7a1825418ee4..99c046ba46bb 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -665,7 +665,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, return ret; } -static int wm9090_i2c_remove(struct i2c_client *i2c) +static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2cb81538cd91..19ca782ac970 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -123,7 +123,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; break; default: - WARN(1, "Unknown DCS readback method"); + WARN(1, "Unknown DCS readback method\n"); break; } diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index fe15bb26e484..25f27ec1dd6e 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -24,7 +24,6 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <sound/soc.h> -#include <sound/soc-of-simple.h> #include "mpc5200_dma.h" #include "mpc5200_psc_ac97.h" @@ -49,7 +48,7 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", .platform_name = "mpc5200-pcm-audio", - ..codec_name = "wm9712-codec", + .codec_name = "wm9712-codec", }, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index a3bfb2e8b70f..73d0edd8ded9 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec) static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->card->codec; + struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ tosa_ext_control(codec); diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c index b897f7b96d89..f8e0ab82ef59 100644 --- a/sound/soc/sh/sh7760-ac97.c +++ b/sound/soc/sh/sh7760-ac97.c @@ -52,8 +52,8 @@ static int __init sh7760_ac97_init(void) unsigned short ipsel; /* enable both AC97 controllers in pinmux reg */ - ipsel = ctrl_inw(IPSEL); - ctrl_outw(ipsel | (3 << 10), IPSEL); + ipsel = __raw_readw(IPSEL); + __raw_writew(ipsel | (3 << 10), IPSEL); ret = -ENOMEM; sh7760_ac97_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c8f3f507f54..614a8b30d87b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev, { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); + int ret; - strict_strtol(buf, 10, &rtd->pmdown_time); + ret = strict_strtol(buf, 10, &rtd->pmdown_time); + if (ret) + return ret; return count; } diff --git a/sound/usb/card.h b/sound/usb/card.h index 1febf2f23754..ae4251d5abf7 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -62,12 +62,14 @@ struct snd_usb_substream { unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ + int freqshift; /* how much to shift the feedback value to get Q16.16 */ unsigned int freqmax; /* maximum sampling rate, used for buffer management */ unsigned int phase; /* phase accumulator */ unsigned int maxpacksize; /* max packet size in bytes */ unsigned int maxframesize; /* max packet size in frames */ unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ + unsigned int syncmaxsize; /* sync endpoint packet size */ unsigned int fill_max: 1; /* fill max packet size always */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 7dae05d8783e..782f741cd00a 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -60,7 +60,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ - { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi */ + { USB_ID(0x041e, 0x3042), 0, 1, 1, 1, 1, 0x000d }, /* Usb X-Fi S51 */ { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; @@ -183,7 +183,13 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (value > 1) return -EINVAL; changed = value != mixer->audigy2nx_leds[index]; - err = snd_usb_ctl_msg(mixer->chip->dev, + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + !value, 0, NULL, 0, 100); + else + err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, value, index + 2, NULL, 0, 100); @@ -225,8 +231,12 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { + /* USB X-Fi S51 doesn't have a CMSS LED */ + if ((mixer->chip->usb_id == USB_ID(0x041e, 0x3042)) && i == 0) + continue; if (i > 1 && /* Live24ext has 2 LEDs only */ (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, @@ -365,6 +375,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3042) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { if ((err = snd_audigy2nx_controls_create(mixer)) < 0) return err; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f49756c1b837..4132522ac90f 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -237,6 +237,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->datainterval = fmt->datainterval; subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; + subs->syncmaxsize = 0; subs->fill_max = 0; /* we need a sync pipe in async OUT or adaptive IN mode */ @@ -283,6 +284,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; else subs->syncinterval = 3; + subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); } /* always fill max packet size */ @@ -674,8 +676,10 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, if (!needs_knot) return 0; - subs->rate_list.count = count; subs->rate_list.list = kmalloc(sizeof(int) * count, GFP_KERNEL); + if (!subs->rate_list.list) + return -ENOMEM; + subs->rate_list.count = count; subs->rate_list.mask = 0; count = 0; list_for_each_entry(fp, &subs->fmt_list, list) { diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 3c650ab3c91d..961c9a250686 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -132,6 +132,11 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn ? get_full_speed_hz(subs->freqm) : get_high_speed_hz(subs->freqm), subs->freqm >> 16, subs->freqm & 0xffff); + if (subs->freqshift != INT_MIN) + snd_iprintf(buffer, " Feedback Format = %d.%d\n", + (subs->syncmaxsize > 3 ? 32 : 24) + - (16 - subs->freqshift), + 16 - subs->freqshift); } else { snd_iprintf(buffer, " Status: Stop\n"); } diff --git a/sound/usb/urb.c b/sound/usb/urb.c index 8deeaad10f10..e184349aee83 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -225,6 +225,7 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, else subs->freqn = get_usb_high_speed_rate(rate); subs->freqm = subs->freqn; + subs->freqshift = INT_MIN; /* calculate max. frequency */ if (subs->maxpacksize) { /* whatever fits into a max. size packet */ @@ -513,11 +514,10 @@ static int retire_paused_capture_urb(struct snd_usb_substream *subs, /* - * prepare urb for full speed playback sync pipe + * prepare urb for playback sync pipe * * set up the offset and length to receive the current frequency. */ - static int prepare_playback_sync_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) @@ -525,103 +525,78 @@ static int prepare_playback_sync_urb(struct snd_usb_substream *subs, struct snd_urb_ctx *ctx = urb->context; urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); urb->iso_frame_desc[0].offset = 0; return 0; } /* - * prepare urb for high speed playback sync pipe + * process after playback sync complete * - * set up the offset and length to receive the current frequency. - */ - -static int prepare_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * process after full speed playback sync complete - * - * retrieve the current 10.14 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). + * Full speed devices report feedback values in 10.14 format as samples per + * frame, high speed devices in 16.16 format as samples per microframe. + * Because the Audio Class 1 spec was written before USB 2.0, many high speed + * devices use a wrong interpretation, some others use an entirely different + * format. Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. */ static int retire_playback_sync_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { unsigned int f; + int shift; unsigned long flags; - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 3) { - f = combine_triple((u8*)urb->transfer_buffer) << 2; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } - } - - return 0; -} + if (urb->iso_frame_desc[0].status != 0 || + urb->iso_frame_desc[0].actual_length < 3) + return 0; -/* - * process after high speed playback sync complete - * - * retrieve the current 12.13 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). - */ -static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; + f = le32_to_cpup(urb->transfer_buffer); + if (urb->iso_frame_desc[0].actual_length == 3) + f &= 0x00ffffff; + else + f &= 0x0fffffff; + if (f == 0) + return 0; - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); + if (unlikely(subs->freqshift == INT_MIN)) { + /* + * The first time we see a feedback value, determine its format + * by shifting it left or right until it matches the nominal + * frequency value. This assumes that the feedback does not + * differ from the nominal value more than +50% or -25%. + */ + shift = 0; + while (f < subs->freqn - subs->freqn / 4) { + f <<= 1; + shift++; + } + while (f > subs->freqn + subs->freqn / 2) { + f >>= 1; + shift--; } + subs->freqshift = shift; } + else if (subs->freqshift >= 0) + f <<= subs->freqshift; + else + f >>= -subs->freqshift; - return 0; -} - -/* - * process after E-Mu 0202/0404/Tracker Pre high speed playback sync complete - * - * These devices return the number of samples per packet instead of the number - * of samples per microframe. - */ -static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; - - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - f >>= subs->datainterval; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } + if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { + /* + * If the frequency looks valid, set it. + * This value is referred to in prepare_playback_urb(). + */ + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } else { + /* + * Out of range; maybe the shift value is wrong. + * Reset it so that we autodetect again the next time. + */ + subs->freqshift = INT_MIN; } return 0; @@ -878,21 +853,6 @@ static struct snd_urb_ops audio_urb_ops[2] = { }, }; -static struct snd_urb_ops audio_urb_ops_high_speed[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb_hs, - .retire_sync = retire_playback_sync_urb_hs, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb_hs, - .retire_sync = retire_capture_sync_urb, - }, -}; - /* * initialize the substream instance. */ @@ -909,23 +869,9 @@ void snd_usb_init_substream(struct snd_usb_stream *as, subs->direction = stream; subs->dev = as->chip->dev; subs->txfr_quirk = as->chip->txfr_quirk; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { - subs->ops = audio_urb_ops[stream]; - } else { - subs->ops = audio_urb_ops_high_speed[stream]; - switch (as->chip->usb_id) { - case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ - case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ - case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ - subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; - break; - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ - subs->ops.prepare_sync = prepare_playback_sync_urb; - subs->ops.retire_sync = retire_playback_sync_urb; - break; - } - } + subs->ops = audio_urb_ops[stream]; + if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) + subs->ops.prepare_sync = prepare_capture_sync_urb_hs; snd_usb_set_pcm_ops(as->pcm, stream); |