diff options
Diffstat (limited to 'sound')
30 files changed, 336 insertions, 127 deletions
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index ee47924aef0d..827161bc269c 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -117,7 +117,7 @@ destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) conn = &efw->in_conn; amdtp_stream_destroy(stream); - cmp_connection_destroy(&efw->out_conn); + cmp_connection_destroy(conn); } static int diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 4ad3bd7fd445..f1657a4e0621 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -343,7 +343,7 @@ int snd_tscm_stream_init_duplex(struct snd_tscm *tscm) if (err < 0) amdtp_stream_destroy(&tscm->rx_stream); - return 0; + return err; } /* At bus reset, streaming is stopped and some registers are clear. */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9448daff9d8b..7d660ee1d5e8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2230,6 +2230,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), + SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), @@ -6983,6 +6984,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index efe3a44658d5..4576f987a4a5 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -561,9 +561,9 @@ static void nau8825_xtalk_prepare(struct nau8825 *nau8825) nau8825_xtalk_backup(nau8825); /* Config IIS as master to output signal by codec */ regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_MS_MASK | NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_MASTER | - (0x2 << NAU8825_I2S_DRV_SFT) | 0x1); + (0x2 << NAU8825_I2S_LRC_DIV_SFT) | 0x1); /* Ramp up headphone volume to 0dB to get better performance and * avoid pop noise in headphone. */ @@ -657,7 +657,7 @@ static void nau8825_xtalk_clean(struct nau8825 *nau8825) NAU8825_IRQ_RMS_EN, NAU8825_IRQ_RMS_EN); /* Recover default value for IIS */ regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_MS_MASK | NAU8825_I2S_LRC_DIV_MASK | NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_SLAVE); /* Restore value of specific register for cross talk */ nau8825_xtalk_restore(nau8825); @@ -2006,7 +2006,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_INTEGER_MASK, fll_param->fll_int); /* FLL pre-scaler */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL4, - NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); + NAU8825_FLL_REF_DIV_MASK, + fll_param->clk_ref_div << NAU8825_FLL_REF_DIV_SFT); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 5d1704e73241..514fd13c2f46 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -137,7 +137,8 @@ #define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ -#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) +#define NAU8825_FLL_REF_DIV_SFT 10 +#define NAU8825_FLL_REF_DIV_MASK (0x3 << NAU8825_FLL_REF_DIV_SFT) /* FLL5 (0x08) */ #define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) @@ -247,8 +248,8 @@ /* I2S_PCM_CTRL2 (0x1d) */ #define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ -#define NAU8825_I2S_DRV_SFT 12 -#define NAU8825_I2S_DRV_MASK (0x3 << NAU8825_I2S_DRV_SFT) +#define NAU8825_I2S_LRC_DIV_SFT 12 +#define NAU8825_I2S_LRC_DIV_MASK (0x3 << NAU8825_I2S_LRC_DIV_SFT) #define NAU8825_I2S_MS_SFT 3 #define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index b0c264d361bc..43dee1b5779d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3835,6 +3835,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } } + regmap_update_bits(rt5645->regmap, RT5645_ADDA_CLK1, + RT5645_I2S_PD1_MASK, RT5645_I2S_PD1_2); + if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8877b74b0510..bb94d50052d7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -126,6 +126,16 @@ static const struct reg_default aic3x_reg[] = { { 108, 0x00 }, { 109, 0x00 }, }; +static bool aic3x_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC3X_RESET: + return true; + default: + return false; + } +} + static const struct regmap_config aic3x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -133,6 +143,9 @@ static const struct regmap_config aic3x_regmap = { .max_register = DAC_ICC_ADJ, .reg_defaults = aic3x_reg, .num_reg_defaults = ARRAY_SIZE(aic3x_reg), + + .volatile_reg = aic3x_volatile_reg, + .cache_type = REGCACHE_RBTREE, }; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 593b7d1aed46..d72ccef9e238 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1551,7 +1551,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wmfw_region *region; const struct wm_adsp_region *mem; const char *region_name; - char *file, *text; + char *file, *text = NULL; struct wm_adsp_buf *buf; unsigned int reg; int regions = 0; @@ -1700,10 +1700,21 @@ static int wm_adsp_load(struct wm_adsp *dsp) regions, le32_to_cpu(region->len), offset, region_name); + if ((pos + le32_to_cpu(region->len) + sizeof(*region)) > + firmware->size) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %zu\n", + file, regions, region_name, + le32_to_cpu(region->len), firmware->size); + ret = -EINVAL; + goto out_fw; + } + if (text) { memcpy(text, region->data, le32_to_cpu(region->len)); adsp_info(dsp, "%s: %s\n", file, text); kfree(text); + text = NULL; } if (reg) { @@ -1748,6 +1759,7 @@ out_fw: regmap_async_complete(regmap); wm_adsp_buf_free(&buf_list); release_firmware(firmware); + kfree(text); out: kfree(file); @@ -2233,6 +2245,17 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { + if ((pos + le32_to_cpu(blk->len) + sizeof(*blk)) > + firmware->size) { + adsp_err(dsp, + "%s.%d: %s region len %d bytes exceeds file length %zu\n", + file, blocks, region_name, + le32_to_cpu(blk->len), + firmware->size); + ret = -EINVAL; + goto out_fw; + } + buf = wm_adsp_buf_alloc(blk->data, le32_to_cpu(blk->len), &buf_list); diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 2998954a1c74..bdf8398cbc81 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -681,22 +681,19 @@ static int dw_i2s_probe(struct platform_device *pdev) } if (!pdata) { - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); - if (ret == -EPROBE_DEFER) { - dev_err(&pdev->dev, - "failed to register PCM, deferring probe\n"); - return ret; - } else if (ret) { - dev_err(&pdev->dev, - "Could not register DMA PCM: %d\n" - "falling back to PIO mode\n", ret); + if (irq >= 0) { ret = dw_pcm_register(pdev); - if (ret) { - dev_err(&pdev->dev, - "Could not register PIO PCM: %d\n", + dev->use_pio = true; + } else { + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + 0); + dev->use_pio = false; + } + + if (ret) { + dev_err(&pdev->dev, "could not register pcm: %d\n", ret); - goto err_clk_disable; - } + goto err_clk_disable; } } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 50349437d961..fde08660b63b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -224,6 +224,12 @@ struct fsl_ssi_soc_data { * @dbg_stats: Debugging statistics * * @soc: SoC specific data + * + * @fifo_watermark: the FIFO watermark setting. Notifies DMA when + * there are @fifo_watermark or fewer words in TX fifo or + * @fifo_watermark or more empty words in RX fifo. + * @dma_maxburst: max number of words to transfer in one go. So far, + * this is always the same as fifo_watermark. */ struct fsl_ssi_private { struct regmap *regs; @@ -263,6 +269,9 @@ struct fsl_ssi_private { const struct fsl_ssi_soc_data *soc; struct device *dev; + + u32 fifo_watermark; + u32 dma_maxburst; }; /* @@ -1051,21 +1060,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, regmap_write(regs, CCSR_SSI_SRCR, srcr); regmap_write(regs, CCSR_SSI_SCR, scr); - /* - * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't - * use FIFO 1. We program the transmit water to signal a DMA transfer - * if there are only two (or fewer) elements left in the FIFO. Two - * elements equals one frame (left channel, right channel). This value, - * however, depends on the depth of the transmit buffer. - * - * We set the watermark on the same level as the DMA burstsize. For - * fiq it is probably better to use the biggest possible watermark - * size. - */ - if (ssi_private->use_dma) - wm = ssi_private->fifo_depth - 2; - else - wm = ssi_private->fifo_depth; + wm = ssi_private->fifo_watermark; regmap_write(regs, CCSR_SSI_SFCSR, CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) | @@ -1373,12 +1368,8 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); - /* - * We have burstsize be "fifo_depth - 2" to match the SSI - * watermark setting in fsl_ssi_startup(). - */ - ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2; - ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.maxburst = ssi_private->dma_maxburst; + ssi_private->dma_params_rx.maxburst = ssi_private->dma_maxburst; ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; @@ -1543,6 +1534,47 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + /* + * Set the watermark for transmit FIFO 0 and receive FIFO 0. We don't + * use FIFO 1 but set the watermark appropriately nontheless. + * We program the transmit water to signal a DMA transfer + * if there are N elements left in the FIFO. For chips with 15-deep + * FIFOs, set watermark to 8. This allows the SSI to operate at a + * high data rate without channel slipping. Behavior is unchanged + * for the older chips with a fifo depth of only 8. A value of 4 + * might be appropriate for the older chips, but is left at + * fifo_depth-2 until sombody has a chance to test. + * + * We set the watermark on the same level as the DMA burstsize. For + * fiq it is probably better to use the biggest possible watermark + * size. + */ + switch (ssi_private->fifo_depth) { + case 15: + /* + * 2 samples is not enough when running at high data + * rates (like 48kHz @ 16 bits/channel, 16 channels) + * 8 seems to split things evenly and leave enough time + * for the DMA to fill the FIFO before it's over/under + * run. + */ + ssi_private->fifo_watermark = 8; + ssi_private->dma_maxburst = 8; + break; + case 8: + default: + /* + * maintain old behavior for older chips. + * Keeping it the same because I don't have an older + * board to test with. + * I suspect this could be changed to be something to + * leave some more space in the fifo. + */ + ssi_private->fifo_watermark = ssi_private->fifo_depth - 2; + ssi_private->dma_maxburst = ssi_private->fifo_depth - 2; + break; + } + dev_set_drvdata(&pdev->dev, ssi_private); if (ssi_private->soc->imx) { diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 744e5eab2298..5c7219fb3aa8 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -142,7 +142,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, * for Jack detection and button press */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_RCCLK, - 0, + 48000 * 512, SND_SOC_CLOCK_IN); if (!ret) { if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && priv->mclk) @@ -835,10 +835,20 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) if ((byt_rt5640_quirk & BYT_RT5640_MCLK_EN) && (is_valleyview())) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); if (IS_ERR(priv->mclk)) { + ret_val = PTR_ERR(priv->mclk); + dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(priv->mclk)); - return PTR_ERR(priv->mclk); + "Failed to get MCLK from pmc_plt_clk_3: %d\n", + ret_val); + + /* + * Fall back to bit clock usage for -ENOENT (clock not + * available likely due to missing dependencies), bail + * for all other errors, including -EPROBE_DEFER + */ + if (ret_val != -ENOENT) + return ret_val; + byt_rt5640_quirk &= ~BYT_RT5640_MCLK_EN; } } diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 55dc9f27d4b2..e12520e142ff 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -254,6 +254,9 @@ static int skl_pcm_open(struct snd_pcm_substream *substream, snd_pcm_set_sync(substream); mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); + if (!mconfig) + return -EINVAL; + skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); return 0; diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 8fc3178bc79c..b30bd384c8d3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -515,6 +515,9 @@ EXPORT_SYMBOL_GPL(skl_sst_init_fw); void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) { + + if (ctx->dsp->fw) + release_firmware(ctx->dsp->fw); skl_clear_module_table(ctx->dsp); skl_freeup_uuid_list(ctx); skl_ipc_free(&ctx->ipc); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index a002ab892772..b42f301c6b96 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -119,23 +119,33 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * Set SAIF clock * * The SAIF clock should be either 384*fs or 512*fs. - * If MCLK is used, the SAIF clk ratio need to match mclk ratio. - * For 32x mclk, set saif clk as 512*fs. - * For 48x mclk, set saif clk as 384*fs. + * If MCLK is used, the SAIF clk ratio needs to match mclk ratio. + * For 256x, 128x, 64x, and 32x sub-rates, set saif clk as 512*fs. + * For 192x, 96x, and 48x sub-rates, set saif clk as 384*fs. * * If MCLK is not used, we just set saif clk to 512*fs. */ clk_prepare_enable(master_saif->clk); if (master_saif->mclk_in_use) { - if (mclk % 32 == 0) { + switch (mclk / rate) { + case 32: + case 64: + case 128: + case 256: + case 512: scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; ret = clk_set_rate(master_saif->clk, 512 * rate); - } else if (mclk % 48 == 0) { + break; + case 48: + case 96: + case 192: + case 384: scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; ret = clk_set_rate(master_saif->clk, 384 * rate); - } else { - /* SAIF MCLK should be either 32x or 48x */ + break; + default: + /* SAIF MCLK should be a sub-rate of 512x or 384x */ clk_disable_unprepare(master_saif->clk); return -EINVAL; } @@ -299,6 +309,16 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) return -EBUSY; } + /* If SAIF1 is configured as slave, the clk gate needs to be cleared + * before the register can be written. + */ + if (saif->id != saif->master_id) { + __raw_writel(BM_SAIF_CTRL_SFTRST, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + __raw_writel(BM_SAIF_CTRL_CLKGATE, + saif->base + SAIF_CTRL + MXS_CLR_ADDR); + } + scr0 = __raw_readl(saif->base + SAIF_CTRL); scr0 = scr0 & ~BM_SAIF_CTRL_BITCLK_EDGE & ~BM_SAIF_CTRL_LRCLK_POLARITY \ & ~BM_SAIF_CTRL_JUSTIFY & ~BM_SAIF_CTRL_DELAY; diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index cda656e4afc6..9104c98deeb7 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -37,8 +37,12 @@ int samsung_asoc_dma_platform_register(struct device *dev, dma_filter_fn filter, pcm_conf->prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; pcm_conf->compat_filter_fn = filter; - pcm_conf->chan_names[SNDRV_PCM_STREAM_PLAYBACK] = tx; - pcm_conf->chan_names[SNDRV_PCM_STREAM_CAPTURE] = rx; + if (dev->of_node) { + pcm_conf->chan_names[SNDRV_PCM_STREAM_PLAYBACK] = tx; + pcm_conf->chan_names[SNDRV_PCM_STREAM_CAPTURE] = rx; + } else { + flags |= SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME; + } return devm_snd_dmaengine_pcm_register(dev, pcm_conf, flags); } diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index e00974bc5616..85324e61cbd5 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1305,6 +1305,8 @@ static int samsung_i2s_probe(struct platform_device *pdev) } pri_dai->dma_playback.addr = regs_base + I2STXD; pri_dai->dma_capture.addr = regs_base + I2SRXD; + pri_dai->dma_playback.chan_name = "tx"; + pri_dai->dma_capture.chan_name = "rx"; pri_dai->dma_playback.addr_width = 4; pri_dai->dma_capture.addr_width = 4; pri_dai->quirks = quirks; @@ -1329,6 +1331,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) sec_dai->lock = &pri_dai->spinlock; sec_dai->variant_regs = pri_dai->variant_regs; sec_dai->dma_playback.addr = regs_base + I2STXDS; + sec_dai->dma_playback.chan_name = "tx-sec"; if (!np) { sec_dai->dma_playback.filter_data = i2s_pdata->dma_play_sec; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 6d0b8897fa6c..0a4718207e6e 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -35,10 +35,12 @@ #include <linux/platform_data/asoc-s3c.h> static struct snd_dmaengine_dai_dma_data s3c2412_i2s_pcm_stereo_out = { + .chan_name = "tx", .addr_width = 4, }; static struct snd_dmaengine_dai_dma_data s3c2412_i2s_pcm_stereo_in = { + .chan_name = "rx", .addr_width = 4, }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 07f5091b33e8..91e6871e5413 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,10 +31,12 @@ #include "s3c24xx-i2s.h" static struct snd_dmaengine_dai_dma_data s3c24xx_i2s_pcm_stereo_out = { + .chan_name = "tx", .addr_width = 2, }; static struct snd_dmaengine_dai_dma_data s3c24xx_i2s_pcm_stereo_in = { + .chan_name = "rx", .addr_width = 2, }; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4bd68de76130..47b370cb2d3b 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -363,8 +363,6 @@ struct rsnd_mod *rsnd_mod_next(int *iterator, if (!mod) continue; - (*iterator)++; - return mod; } @@ -1030,10 +1028,8 @@ static int __rsnd_kctrl_new(struct rsnd_mod *mod, return -ENOMEM; ret = snd_ctl_add(card, kctrl); - if (ret < 0) { - snd_ctl_free_one(kctrl); + if (ret < 0) return ret; - } cfg->update = update; cfg->card = card; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index b90df77662df..7410ec0174db 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -374,10 +374,10 @@ struct rsnd_mod *rsnd_mod_next(int *iterator, int array_size); #define for_each_rsnd_mod(iterator, pos, io) \ for (iterator = 0; \ - (pos = rsnd_mod_next(&iterator, io, NULL, 0));) + (pos = rsnd_mod_next(&iterator, io, NULL, 0)); iterator++) #define for_each_rsnd_mod_arrays(iterator, pos, io, array, size) \ for (iterator = 0; \ - (pos = rsnd_mod_next(&iterator, io, array, size));) + (pos = rsnd_mod_next(&iterator, io, array, size)); iterator++) #define for_each_rsnd_mod_array(iterator, pos, io, array) \ for_each_rsnd_mod_arrays(iterator, pos, io, array, ARRAY_SIZE(array)) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 3a8f65bd1bf9..42db48db09ba 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -390,6 +390,9 @@ static int rsnd_src_init(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); + /* reset sync convert_rate */ + src->sync.val = 0; + rsnd_mod_power_on(mod); rsnd_src_activation(mod); @@ -398,9 +401,6 @@ static int rsnd_src_init(struct rsnd_mod *mod, rsnd_src_status_clear(mod); - /* reset sync convert_rate */ - src->sync.val = 0; - return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 530a4dba0709..78fa42589fdf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1749,6 +1749,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) component->init = aux_dev->init; component->auxiliary = 1; + list_add(&component->card_aux_list, &card->aux_comp_list); return 0; @@ -1759,16 +1760,14 @@ err_defer: static int soc_probe_aux_devices(struct snd_soc_card *card) { - struct snd_soc_component *comp; + struct snd_soc_component *comp, *tmp; int order; int ret; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - list_for_each_entry(comp, &card->component_dev_list, card_list) { - if (!comp->auxiliary) - continue; - + list_for_each_entry_safe(comp, tmp, &card->aux_comp_list, + card_aux_list) { if (comp->driver->probe_order == order) { ret = soc_probe_component(card, comp); if (ret < 0) { @@ -1777,6 +1776,7 @@ static int soc_probe_aux_devices(struct snd_soc_card *card) comp->name, ret); return ret; } + list_del(&comp->card_aux_list); } } } @@ -3110,6 +3110,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; component->suspend = component->driver->suspend; component->resume = component->driver->resume; + component->pcm_new = component->driver->pcm_new; + component->pcm_free= component->driver->pcm_free; dapm = &component->dapm; dapm->dev = dev; @@ -3292,6 +3294,21 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) platform->driver->remove(platform); } +static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_platform *platform = rtd->platform; + + return platform->driver->pcm_new(rtd); +} + +static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_platform *platform = rtd->platform; + + platform->driver->pcm_free(pcm); +} + /** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform @@ -3315,6 +3332,10 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->component.probe = snd_soc_platform_drv_probe; if (platform_drv->remove) platform->component.remove = snd_soc_platform_drv_remove; + if (platform_drv->pcm_new) + platform->component.pcm_new = snd_soc_platform_drv_pcm_new; + if (platform_drv->pcm_free) + platform->component.pcm_free = snd_soc_platform_drv_pcm_free; #ifdef CONFIG_DEBUG_FS platform->component.debugfs_prefix = "platform"; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 27dd02e57b31..dcef67a9bd48 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -363,6 +363,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, snd_soc_dapm_new_control_unlocked(widget->dapm, &template); kfree(name); + if (IS_ERR(data->widget)) { + ret = PTR_ERR(data->widget); + goto err_data; + } if (!data->widget) { ret = -ENOMEM; goto err_data; @@ -397,6 +401,10 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, data->widget = snd_soc_dapm_new_control_unlocked( widget->dapm, &template); kfree(name); + if (IS_ERR(data->widget)) { + ret = PTR_ERR(data->widget); + goto err_data; + } if (!data->widget) { ret = -ENOMEM; goto err_data; @@ -3403,11 +3411,22 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); w = snd_soc_dapm_new_control_unlocked(dapm, widget); + /* Do not nag about probe deferrals */ + if (IS_ERR(w)) { + int ret = PTR_ERR(w); + + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, + "ASoC: Failed to create DAPM control %s (%d)\n", + widget->name, ret); + goto out_unlock; + } if (!w) dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", widget->name); +out_unlock: mutex_unlock(&dapm->card->dapm_mutex); return w; } @@ -3430,6 +3449,8 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->regulator = devm_regulator_get(dapm->dev, w->name); if (IS_ERR(w->regulator)) { ret = PTR_ERR(w->regulator); + if (ret == -EPROBE_DEFER) + return ERR_PTR(ret); dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", w->name, ret); return NULL; @@ -3448,6 +3469,8 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); + if (ret == -EPROBE_DEFER) + return ERR_PTR(ret); dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", w->name, ret); return NULL; @@ -3566,6 +3589,16 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm, mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { w = snd_soc_dapm_new_control_unlocked(dapm, widget); + if (IS_ERR(w)) { + ret = PTR_ERR(w); + /* Do not nag about probe deferrals */ + if (ret == -EPROBE_DEFER) + break; + dev_err(dapm->dev, + "ASoC: Failed to create DAPM control %s (%d)\n", + widget->name, ret); + break; + } if (!w) { dev_err(dapm->dev, "ASoC: Failed to create DAPM control %s\n", @@ -3842,6 +3875,15 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); + if (IS_ERR(w)) { + ret = PTR_ERR(w); + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(card->dev, + "ASoC: Failed to create %s widget (%d)\n", + link_name, ret); + goto outfree_kcontrol_news; + } if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); @@ -3893,6 +3935,16 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); + if (IS_ERR(w)) { + int ret = PTR_ERR(w); + + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, + "ASoC: Failed to create %s widget (%d)\n", + dai->driver->playback.stream_name, ret); + return ret; + } if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->playback.stream_name); @@ -3912,6 +3964,16 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); + if (IS_ERR(w)) { + int ret = PTR_ERR(w); + + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, + "ASoC: Failed to create %s widget (%d)\n", + dai->driver->playback.stream_name, ret); + return ret; + } if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->capture.stream_name); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 17eb14935577..d53786498b61 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -263,6 +263,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; struct device *dev = rtd->platform->dev; + struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; size_t prealloc_buffer_size; size_t max_buffer_size; @@ -282,6 +283,13 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (!pcm->chan[i] && + (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) + pcm->chan[i] = dma_request_slave_channel(dev, + dma_data->chan_name); + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); @@ -350,7 +358,9 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, const char *name; struct dma_chan *chan; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) + if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || + !dev->of_node) return 0; if (config && config->dma_dev) { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e7a1eaa2772f..efc5831f205d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1055,7 +1055,6 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai; int i, ret; @@ -1071,12 +1070,6 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, } } - if (platform->driver->bespoke_trigger) { - ret = platform->driver->bespoke_trigger(substream, cmd); - if (ret < 0) - return ret; - } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->bespoke_trigger) { ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); if (ret < 0) @@ -1116,13 +1109,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) } delay += codec_delay; - /* - * None of the existing platform drivers implement delay(), so - * for now the codec_dai of first multicodec entry is used - */ - if (platform->driver->delay) - delay += platform->driver->delay(substream, rtd->codec_dais[0]); - runtime->delay = delay; return offset; @@ -2184,9 +2170,11 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: fe->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP; break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PAUSED; + break; } out: @@ -2640,12 +2628,25 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) return ret; } +static void soc_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_soc_component *component; + + list_for_each_entry(component, &rtd->card->component_dev_list, + card_list) { + if (component->pcm_free) + component->pcm_free(pcm); + } +} + /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_component *component; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; @@ -2754,17 +2755,18 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (capture) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops); - if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd); - if (ret < 0) { - dev_err(platform->dev, - "ASoC: pcm constructor failed: %d\n", - ret); - return ret; + list_for_each_entry(component, &rtd->card->component_dev_list, card_list) { + if (component->pcm_new) { + ret = component->pcm_new(rtd); + if (ret < 0) { + dev_err(component->dev, + "ASoC: pcm constructor failed: %d\n", + ret); + return ret; + } } } - - pcm->private_free = platform->driver->pcm_free; + pcm->private_free = soc_pcm_free; out: dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name, @@ -2872,15 +2874,6 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -int snd_soc_platform_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_platform *platform) -{ - if (platform->driver->ops && platform->driver->ops->trigger) - return platform->driver->ops->trigger(substream, cmd); - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_platform_trigger); - #ifdef CONFIG_DEBUG_FS static const char *dpcm_state_string(enum snd_soc_dpcm_state state) { diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 65670b2b408c..01e8bb0910b2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -514,13 +514,12 @@ static void remove_widget(struct snd_soc_component *comp, == SND_SOC_TPLG_TYPE_MIXER) kfree(kcontrol->tlv.p); - snd_ctl_remove(card, kcontrol); - /* Private value is used as struct soc_mixer_control * for volume mixers or soc_bytes_ext for bytes * controls. */ kfree((void *)kcontrol->private_value); + snd_ctl_remove(card, kcontrol); } kfree(w->kcontrol_news); } @@ -1556,6 +1555,15 @@ widget: widget = snd_soc_dapm_new_control(dapm, &template); else widget = snd_soc_dapm_new_control_unlocked(dapm, &template); + if (IS_ERR(widget)) { + ret = PTR_ERR(widget); + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(tplg->dev, + "ASoC: failed to create widget %s controls (%d)\n", + w->name, ret); + goto hdr_err; + } if (widget == NULL) { dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n", w->name); diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index f24d19526603..4237323ef594 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -694,10 +694,10 @@ static int sun4i_i2s_probe(struct platform_device *pdev) } i2s->playback_dma_data.addr = res->start + SUN4I_I2S_FIFO_TX_REG; - i2s->playback_dma_data.maxburst = 4; + i2s->playback_dma_data.maxburst = 8; i2s->capture_dma_data.addr = res->start + SUN4I_I2S_FIFO_RX_REG; - i2s->capture_dma_data.maxburst = 4; + i2s->capture_dma_data.maxburst = 8; pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 15d1d5c63c3c..c90607ebe155 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -384,6 +384,9 @@ static void snd_complete_urb(struct urb *urb) if (unlikely(atomic_read(&ep->chip->shutdown))) goto exit_clear; + if (unlikely(!test_bit(EP_FLAG_RUNNING, &ep->flags))) + goto exit_clear; + if (usb_pipeout(ep->pipe)) { retire_outbound_urb(ep, ctx); /* can be stopped during retire callback */ @@ -534,6 +537,11 @@ static int wait_clear_urbs(struct snd_usb_endpoint *ep) alive, ep->ep_num); clear_bit(EP_FLAG_STOPPING, &ep->flags); + ep->data_subs = NULL; + ep->sync_slave = NULL; + ep->retire_data_urb = NULL; + ep->prepare_data_urb = NULL; + return 0; } @@ -912,9 +920,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, /** * snd_usb_endpoint_start: start an snd_usb_endpoint * - * @ep: the endpoint to start - * @can_sleep: flag indicating whether the operation is executed in - * non-atomic context + * @ep: the endpoint to start * * A call to this function will increment the use count of the endpoint. * In case it is not already running, the URBs for this endpoint will be @@ -924,7 +930,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, * * Returns an error if the URB submission failed, 0 in all other cases. */ -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep) +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) { int err; unsigned int i; @@ -938,8 +944,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep) /* just to be sure */ deactivate_urbs(ep, false); - if (can_sleep) - wait_clear_urbs(ep); ep->active_mask = 0; ep->unlink_mask = 0; @@ -1020,10 +1024,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep) if (--ep->use_count == 0) { deactivate_urbs(ep, false); - ep->data_subs = NULL; - ep->sync_slave = NULL; - ep->retire_data_urb = NULL; - ep->prepare_data_urb = NULL; set_bit(EP_FLAG_STOPPING, &ep->flags); } } diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6428392d8f62..584f295d7c77 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -18,7 +18,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, bool can_sleep); +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep); void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 34c6d4f2c0b6..9aa5b1855481 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -218,7 +218,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } -static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) +static int start_endpoints(struct snd_usb_substream *subs) { int err; @@ -231,7 +231,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) dev_dbg(&subs->dev->dev, "Starting data EP @%p\n", ep); ep->data_subs = subs; - err = snd_usb_endpoint_start(ep, can_sleep); + err = snd_usb_endpoint_start(ep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); return err; @@ -260,7 +260,7 @@ static int start_endpoints(struct snd_usb_substream *subs, bool can_sleep) dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; - err = snd_usb_endpoint_start(ep, can_sleep); + err = snd_usb_endpoint_start(ep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); return err; @@ -850,7 +850,7 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) - ret = start_endpoints(subs, true); + ret = start_endpoints(subs); unlock: snd_usb_unlock_shutdown(subs->stream->chip); @@ -1666,7 +1666,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream switch (cmd) { case SNDRV_PCM_TRIGGER_START: - err = start_endpoints(subs, false); + err = start_endpoints(subs); if (err < 0) return err; |