diff options
Diffstat (limited to 'sound')
30 files changed, 199 insertions, 133 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1fe581167b7b..d083225344a0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -739,6 +739,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) runtime->boundary *= 2; + /* clear the buffer for avoiding possible kernel info leaks */ + if (runtime->dma_area && !substream->ops->copy_user) + memset(runtime->dma_area, 0, runtime->dma_bytes); + snd_pcm_timer_resolution_change(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP); diff --git a/sound/firewire/fireface/ff-pcm.c b/sound/firewire/fireface/ff-pcm.c index 4e3bd9a2bec0..bd91c6ecb112 100644 --- a/sound/firewire/fireface/ff-pcm.c +++ b/sound/firewire/fireface/ff-pcm.c @@ -247,7 +247,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&ff->mutex); } - return 0; + return err; } static int pcm_hw_free(struct snd_pcm_substream *substream) diff --git a/sound/firewire/motu/motu-pcm.c b/sound/firewire/motu/motu-pcm.c index 349b4d09e84f..005970931030 100644 --- a/sound/firewire/motu/motu-pcm.c +++ b/sound/firewire/motu/motu-pcm.c @@ -177,18 +177,14 @@ static int pcm_open(struct snd_pcm_substream *substream) err = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, frames_per_period, frames_per_period); - if (err < 0) { - mutex_unlock(&motu->mutex); + if (err < 0) goto err_locked; - } err = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, frames_per_buffer, frames_per_buffer); - if (err < 0) { - mutex_unlock(&motu->mutex); + if (err < 0) goto err_locked; - } } } diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 9124603edabe..67fd3e844dd6 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -285,7 +285,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, mutex_unlock(&oxfw->mutex); } - return 0; + return err; } static int pcm_capture_hw_free(struct snd_pcm_substream *substream) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index f9707fb05efe..682ed39f79b0 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -120,10 +120,8 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev) snd_hdac_stream_updateb(azx_dev, SD_CTL, SD_CTL_DMA_START | SD_INT_MASK, 0); snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ - if (azx_dev->stripe) { + if (azx_dev->stripe) snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0); - azx_dev->stripe = 0; - } azx_dev->running = false; } EXPORT_SYMBOL_GPL(snd_hdac_stream_clear); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 50d4a87a6bb3..f02f5b1568de 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -635,36 +635,30 @@ This function assumes there are no more than 16 in/out busses or pipes Meters is an array [3][16][2] of long. */ static void get_audio_meters(struct echoaudio *chip, long *meters) { - int i, m, n; + unsigned int i, m, n; - m = 0; - n = 0; - for (i = 0; i < num_busses_out(chip); i++, m++) { + for (i = 0 ; i < 96; i++) + meters[i] = 0; + + for (m = 0, n = 0, i = 0; i < num_busses_out(chip); i++, m++) { meters[n++] = chip->comm_page->vu_meter[m]; meters[n++] = chip->comm_page->peak_meter[m]; } - for (; n < 32; n++) - meters[n] = 0; #ifdef ECHOCARD_ECHO3G m = E3G_MAX_OUTPUTS; /* Skip unused meters */ #endif - for (i = 0; i < num_busses_in(chip); i++, m++) { + for (n = 32, i = 0; i < num_busses_in(chip); i++, m++) { meters[n++] = chip->comm_page->vu_meter[m]; meters[n++] = chip->comm_page->peak_meter[m]; } - for (; n < 64; n++) - meters[n] = 0; - #ifdef ECHOCARD_HAS_VMIXER - for (i = 0; i < num_pipes_out(chip); i++, m++) { + for (n = 64, i = 0; i < num_pipes_out(chip); i++, m++) { meters[n++] = chip->comm_page->vu_meter[m]; meters[n++] = chip->comm_page->peak_meter[m]; } #endif - for (; n < 96; n++) - meters[n] = 0; } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 2f3b7a35f2d9..ba56b59b3e17 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -883,7 +883,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr, return -EAGAIN; /* give a chance to retry */ } - dev_WARN(chip->card->dev, + dev_err(chip->card->dev, "azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n", bus->last_cmd[addr]); chip->single_cmd = 1; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 35b4526f0d28..b856b89378ac 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1419,7 +1419,6 @@ static bool atpx_present(void) return true; } } - pci_dev_put(pdev); } return false; } diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index b7a1abb3e231..32ed46464af7 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1809,13 +1809,14 @@ struct scp_msg { static void dspio_clear_response_queue(struct hda_codec *codec) { + unsigned long timeout = jiffies + msecs_to_jiffies(1000); unsigned int dummy = 0; - int status = -1; + int status; /* clear all from the response queue */ do { status = dspio_read(codec, &dummy); - } while (status == 0); + } while (status == 0 && time_before(jiffies, timeout)); } static int dspio_get_response_data(struct hda_codec *codec) @@ -7588,12 +7589,14 @@ static void ca0132_process_dsp_response(struct hda_codec *codec, struct ca0132_spec *spec = codec->spec; codec_dbg(codec, "ca0132_process_dsp_response\n"); + snd_hda_power_up_pm(codec); if (spec->wait_scp) { if (dspio_get_response_data(codec) >= 0) spec->wait_scp = 0; } dspio_clear_response_queue(codec); + snd_hda_power_down_pm(codec); } static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) @@ -7604,11 +7607,10 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) /* Delay enabling the HP amp, to let the mic-detection * state machine run. */ - cancel_delayed_work(&spec->unsol_hp_work); - schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); tbl = snd_hda_jack_tbl_get(codec, cb->nid); if (tbl) tbl->block_report = 1; + schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500)); } static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb) @@ -8454,12 +8456,25 @@ static void ca0132_reboot_notify(struct hda_codec *codec) codec->patch_ops.free(codec); } +#ifdef CONFIG_PM +static int ca0132_suspend(struct hda_codec *codec) +{ + struct ca0132_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->unsol_hp_work); + return 0; +} +#endif + static const struct hda_codec_ops ca0132_patch_ops = { .build_controls = ca0132_build_controls, .build_pcms = ca0132_build_pcms, .init = ca0132_init, .free = ca0132_free, .unsol_event = snd_hda_jack_unsol_event, +#ifdef CONFIG_PM + .suspend = ca0132_suspend, +#endif .reboot_notify = ca0132_reboot_notify, }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78647ee02339..630b1f5c276d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2021,6 +2021,8 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, per_cvt->assigned = 0; hinfo->nid = 0; + azx_stream(get_azx_dev(substream))->stripe = 0; + mutex_lock(&spec->pcm_lock); snd_hda_spdif_ctls_unassign(codec, pcm_idx); clear_bit(pcm_idx, &spec->pcm_in_use); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d6e34b3b3aa..dbfafee97931 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7643,11 +7643,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x1a, 0x90a70130}, {0x1b, 0x90170110}, {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, - {0x12, 0xb7a60130}, - {0x13, 0xb8a61140}, - {0x16, 0x90170110}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, @@ -7841,6 +7836,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x19, 0x40000000}, {0x1a, 0x40000000}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, + {0x19, 0x40000000}, + {0x1a, 0x40000000}), {} }; diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f4ee6798154a..7a5621e5e233 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -96,14 +96,19 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int da7219_clk_enable(struct snd_pcm_substream *substream, - int wclk_rate, int bclk_rate) +static int da7219_clk_enable(struct snd_pcm_substream *substream) { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - clk_set_rate(da7219_dai_wclk, wclk_rate); - clk_set_rate(da7219_dai_bclk, bclk_rate); + /* + * Set wclk to 48000 because the rate constraint of this driver is + * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is + * minimum of 64x the LRCLK sample rate." DA7219 is the only clk + * source so for all codecs we have to limit bclk to 64X lrclk. + */ + clk_set_rate(da7219_dai_wclk, 48000); + clk_set_rate(da7219_dai_bclk, 48000 * 64); ret = clk_prepare_enable(da7219_dai_bclk); if (ret < 0) { dev_err(rtd->dev, "can't enable master clock %d\n", ret); @@ -156,7 +161,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_SP_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) @@ -178,7 +183,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL1; - return 0; + return da7219_clk_enable(substream); } static int cz_max_startup(struct snd_pcm_substream *substream) @@ -199,7 +204,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->play_i2s_instance = I2S_BT_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_dmic0_startup(struct snd_pcm_substream *substream) @@ -220,7 +225,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream) &constraints_rates); machine->cap_i2s_instance = I2S_BT_INSTANCE; - return 0; + return da7219_clk_enable(substream); } static int cz_dmic1_startup(struct snd_pcm_substream *substream) @@ -242,25 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_SP_INSTANCE; machine->capture_channel = CAP_CHANNEL0; - return 0; -} - -static int cz_da7219_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - int wclk, bclk; - - wclk = params_rate(params); - bclk = wclk * params_channels(params) * - snd_pcm_format_width(params_format(params)); - /* ADAU7002 spec: "The ADAU7002 requires a BCLK rate - * that is minimum of 64x the LRCLK sample rate." - * DA7219 is the only clk source so for all codecs - * we have to limit bclk to 64X lrclk. - */ - if (bclk < (wclk * 64)) - bclk = wclk * 64; - return da7219_clk_enable(substream, wclk, bclk); + return da7219_clk_enable(substream); } static void cz_da7219_shutdown(struct snd_pcm_substream *substream) @@ -271,31 +258,26 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream) static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_da7219_cap_ops = { .startup = cz_da7219_cap_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_da7219_shutdown, - .hw_params = cz_da7219_params, }; SND_SOC_DAILINK_DEF(designware1, diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index f8b5b960e597..4eaa2b5b20a5 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -292,7 +292,7 @@ static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; - uinfo->count = FIELD_SIZEOF(struct hdmi_codec_priv, eld); + uinfo->count = sizeof_field(struct hdmi_codec_priv, eld); return 0; } diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index f6bf4cfbea23..e46b6ada13b1 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2103,26 +2103,40 @@ static void max98090_pll_det_disable_work(struct work_struct *work) M98090_IULK_MASK, 0); } -static void max98090_pll_work(struct work_struct *work) +static void max98090_pll_work(struct max98090_priv *max98090) { - struct max98090_priv *max98090 = - container_of(work, struct max98090_priv, pll_work); struct snd_soc_component *component = max98090->component; + unsigned int pll; + int i; if (!snd_soc_component_is_active(component)) return; dev_info_ratelimited(component->dev, "PLL unlocked\n"); + /* + * As the datasheet suggested, the maximum PLL lock time should be + * 7 msec. The workaround resets the codec softly by toggling SHDN + * off and on if PLL failed to lock for 10 msec. Notably, there is + * no suggested hold time for SHDN off. + */ + /* Toggle shutdown OFF then ON */ snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, 0); - msleep(10); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); - /* Give PLL time to lock */ - msleep(10); + for (i = 0; i < 10; ++i) { + /* Give PLL time to lock */ + usleep_range(1000, 1200); + + /* Check lock status */ + pll = snd_soc_component_read32( + component, M98090_REG_DEVICE_STATUS); + if (!(pll & M98090_ULK_MASK)) + break; + } } static void max98090_jack_work(struct work_struct *work) @@ -2259,7 +2273,7 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (active & M98090_ULK_MASK) { dev_dbg(component->dev, "M98090_ULK_MASK\n"); - schedule_work(&max98090->pll_work); + max98090_pll_work(max98090); } if (active & M98090_JDET_MASK) { @@ -2422,7 +2436,6 @@ static int max98090_probe(struct snd_soc_component *component) max98090_pll_det_enable_work); INIT_WORK(&max98090->pll_det_disable_work, max98090_pll_det_disable_work); - INIT_WORK(&max98090->pll_work, max98090_pll_work); /* Enable jack detection */ snd_soc_component_write(component, M98090_REG_JACK_DETECT, @@ -2475,7 +2488,6 @@ static void max98090_remove(struct snd_soc_component *component) cancel_delayed_work_sync(&max98090->jack_work); cancel_delayed_work_sync(&max98090->pll_det_enable_work); cancel_work_sync(&max98090->pll_det_disable_work); - cancel_work_sync(&max98090->pll_work); max98090->component = NULL; } diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 57965cd678b4..a197114b0dad 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1530,7 +1530,6 @@ struct max98090_priv { struct delayed_work jack_work; struct delayed_work pll_det_enable_work; struct work_struct pll_det_disable_work; - struct work_struct pll_work; struct snd_soc_jack *jack; unsigned int dai_fmt; int tdm_slots; diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h index 3af36ec928e9..088b77931727 100644 --- a/sound/soc/codecs/rt5677-spi.h +++ b/sound/soc/codecs/rt5677-spi.h @@ -9,9 +9,25 @@ #ifndef __RT5677_SPI_H__ #define __RT5677_SPI_H__ +#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI) int rt5677_spi_read(u32 addr, void *rxbuf, size_t len); int rt5677_spi_write(u32 addr, const void *txbuf, size_t len); int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw); void rt5677_spi_hotword_detected(void); +#else +static inline int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) +{ + return -EINVAL; +} +static inline int rt5677_spi_write(u32 addr, const void *txbuf, size_t len) +{ + return -EINVAL; +} +static inline int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw) +{ + return -EINVAL; +} +static inline void rt5677_spi_hotword_detected(void){} +#endif #endif /* __RT5677_SPI_H__ */ diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index b1713fffa3eb..ae6f6121bc1b 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -73,6 +73,7 @@ struct rt5682_priv { static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, + {RT5682_I2C_CTRL, 0x000f}, }; static const struct reg_default rt5682_reg[] = { @@ -2474,6 +2475,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) mutex_lock(&rt5682->calibrate_mutex); rt5682_reset(rt5682->regmap); + regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af); usleep_range(15000, 20000); regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af); diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 7d7ea15d73e0..5ffbaddd6e49 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1806,6 +1806,12 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id, switch (clk_id) { case WM8904_CLK_AUTO: + /* We don't have any rate constraints, so just ignore the + * request to disable constraining. + */ + if (!freq) + return 0; + mclk_freq = clk_get_rate(priv->mclk); /* enable FLL if a different sysclk is desired */ if (mclk_freq != freq) { diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3e5c69fbc33a..d9d59f45833f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2788,7 +2788,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, if (target % Fref == 0) { fll_div->theta = 0; - fll_div->lambda = 0; + fll_div->lambda = 1; } else { gcd_fll = gcd(target, fratio * Fref); @@ -2858,7 +2858,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s return -EINVAL; } - if (fll_div.theta || fll_div.lambda) + if (fll_div.theta) fll1 |= WM8962_FLL_FRAC; /* Stop the FLL while we reconfigure */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 10b82bf043d1..55e9f8800b3e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -371,6 +371,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, do { struct asoc_simple_data adata; struct device_node *codec; + struct device_node *plat; struct device_node *np; int num = of_get_child_count(node); @@ -381,6 +382,9 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, ret = -ENODEV; goto error; } + /* get platform */ + plat = of_get_child_by_name(node, is_top ? + PREFIX "plat" : "plat"); /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); @@ -389,6 +393,8 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, /* loop for all CPU/Codec node */ for_each_child_of_node(node, np) { + if (plat == np) + continue; /* * It is DPCM * if it has many CPUs, diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index fbecbb74350b..68bcec5241f7 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -14,6 +14,7 @@ #include <linux/module.h> #include <linux/fs.h> #include <linux/interrupt.h> +#include <linux/io.h> #include <linux/firmware.h> #include <linux/pm_runtime.h> #include <linux/pm_qos.h> diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index dd2b5ad08659..243f683bc02a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -707,13 +707,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_MCLK_EN), }, { + /* Teclast X89 */ .matches = { DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"), DMI_MATCH(DMI_BOARD_NAME, "tPAD"), }, .driver_data = (void *)(BYT_RT5640_IN3_MAP | - BYT_RT5640_MCLK_EN | - BYT_RT5640_SSP0_AIF1), + BYT_RT5640_JD_SRC_JD1_IN4P | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_1P0 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), }, { /* Toshiba Satellite Click Mini L9W-B */ .matches = { diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 5d08ae066738..fb9ba8819706 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -9,45 +9,52 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> -static struct snd_soc_acpi_codecs cml_codecs = { +static struct snd_soc_acpi_codecs rt1011_spk_codecs = { .num_codecs = 1, - .codecs = {"10EC5682"} + .codecs = {"10EC1011"} }; -static struct snd_soc_acpi_codecs cml_spk_codecs = { +static struct snd_soc_acpi_codecs max98357a_spk_codecs = { .num_codecs = 1, .codecs = {"MX98357A"} }; +/* + * The order of the three entries with .id = "10EC5682" matters + * here, because DSDT tables expose an ACPI HID for the MAX98357A + * speaker amplifier which is not populated on the board. + */ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { { - .id = "DLGS7219", - .drv_name = "cml_da7219_max98357a", - .quirk_data = &cml_spk_codecs, + .id = "10EC5682", + .drv_name = "cml_rt1011_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1011_spk_codecs, .sof_fw_filename = "sof-cml.ri", - .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", }, { - .id = "MX98357A", + .id = "10EC5682", .drv_name = "sof_rt5682", - .quirk_data = &cml_codecs, + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &max98357a_spk_codecs, .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg", }, { - .id = "10EC1011", - .drv_name = "cml_rt1011_rt5682", - .quirk_data = &cml_codecs, - .sof_fw_filename = "sof-cml.ri", - .sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg", - }, - { .id = "10EC5682", .drv_name = "sof_rt5682", .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-rt5682.tplg", }, - + { + .id = "DLGS7219", + .drv_name = "cml_da7219_max98357a", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &max98357a_spk_codecs, + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-da7219-max98357a.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 61f230324164..6615ef64c7f5 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -214,10 +214,8 @@ be_err: * This is to ensure there are no pops or clicks in between any music tracks * due to DAPM power cycling. */ -static void close_delayed_work(struct work_struct *work) +static void close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dai; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -929,7 +927,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* DAPM dai link stream work */ - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->close_delayed_work_func = close_delayed_work; rtd->compr = compr; compr->private_data = rtd; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 062653ab03a3..1c84ff1a5bf9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -419,7 +419,8 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) list_del(&rtd->list); - flush_delayed_work(&rtd->delayed_work); + if (delayed_work_pending(&rtd->delayed_work)) + flush_delayed_work(&rtd->delayed_work); snd_soc_pcm_component_free(rtd); /* @@ -435,6 +436,15 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd) device_unregister(rtd->dev); } +static void close_delayed_work(struct work_struct *work) { + struct snd_soc_pcm_runtime *rtd = + container_of(work, struct snd_soc_pcm_runtime, + delayed_work.work); + + if (rtd->close_delayed_work_func) + rtd->close_delayed_work_func(rtd); +} + static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -470,6 +480,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; dev_set_drvdata(dev, rtd); + INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); /* * for rtd->codec_dais diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 76b7ee637e86..01e7bc03d92f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -637,10 +637,8 @@ out: * This is to ensure there are no pops or clicks in between any music tracks * due to DAPM power cycling. */ -static void close_delayed_work(struct work_struct *work) +static void close_delayed_work(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = - container_of(work, struct snd_soc_pcm_runtime, delayed_work.work); struct snd_soc_dai *codec_dai = rtd->codec_dais[0]; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); @@ -660,7 +658,7 @@ static void close_delayed_work(struct work_struct *work) mutex_unlock(&rtd->card->pcm_mutex); } -static void codec2codec_close_delayed_work(struct work_struct *work) +static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { /* * Currently nothing to do for c2c links @@ -2974,10 +2972,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) /* DAPM dai link stream work */ if (rtd->dai_link->params) - INIT_DELAYED_WORK(&rtd->delayed_work, - codec2codec_close_delayed_work); + rtd->close_delayed_work_func = codec2codec_close_delayed_work; else - INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); + rtd->close_delayed_work_func = close_delayed_work; pcm->nonatomic = rtd->dai_link->nonatomic; rtd->pcm = pcm; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 81d2af000a5c..b28613149b0c 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1933,11 +1933,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); - kfree(link->name); - kfree(link->stream_name); - kfree(link->cpus->dai_name); - kfree(link); - return ret; + goto err; + } + + ret = snd_soc_add_dai_link(tplg->comp->card, link); + if (ret < 0) { + dev_err(tplg->comp->dev, "ASoC: adding FE link failed\n"); + goto err; } link->dobj.index = tplg->index; @@ -1945,8 +1947,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, link->dobj.type = SND_SOC_DOBJ_DAI_LINK; list_add(&link->dobj.list, &tplg->comp->dobj_list); - snd_soc_add_dai_link(tplg->comp->card, link); return 0; +err: + kfree(link->name); + kfree(link->stream_name); + kfree(link->cpus->dai_name); + kfree(link); + return ret; } /* create a FE DAI and DAI link from the PCM object */ @@ -2039,6 +2046,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, int size; int i; bool abi_match; + int ret; count = le32_to_cpu(hdr->count); @@ -2080,7 +2088,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ - soc_tplg_pcm_create(tplg, _pcm); + ret = soc_tplg_pcm_create(tplg, _pcm); + if (ret < 0) { + if (!abi_match) + kfree(_pcm); + return ret; + } /* offset by version-specific struct size and * real priv data size diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 2abf80b3eb52..92ef6a796fd5 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -24,7 +24,8 @@ #define DRAM_OFFSET 0x100000 #define DRAM_SIZE (160 * 1024) #define SHIM_OFFSET 0x140000 -#define SHIM_SIZE 0x100 +#define SHIM_SIZE_BYT 0x100 +#define SHIM_SIZE_CHT 0x118 #define MBOX_OFFSET 0x144000 #define MBOX_SIZE 0x1000 #define EXCEPT_OFFSET 0x800 @@ -75,7 +76,7 @@ static const struct snd_sof_debugfs_map byt_debugfs[] = { SOF_DEBUGFS_ACCESS_D0_ONLY}, {"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE, SOF_DEBUGFS_ACCESS_D0_ONLY}, - {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE, + {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_BYT, SOF_DEBUGFS_ACCESS_ALWAYS}, }; @@ -102,7 +103,7 @@ static const struct snd_sof_debugfs_map cht_debugfs[] = { SOF_DEBUGFS_ACCESS_D0_ONLY}, {"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE, SOF_DEBUGFS_ACCESS_D0_ONLY}, - {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE, + {"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_CHT, SOF_DEBUGFS_ACCESS_ALWAYS}, }; @@ -145,33 +146,33 @@ static void byt_dump(struct snd_sof_dev *sdev, u32 flags) struct sof_ipc_dsp_oops_xtensa xoops; struct sof_ipc_panic_info panic_info; u32 stack[BYT_STACK_DUMP_SIZE]; - u32 status, panic, imrd, imrx; + u64 status, panic, imrd, imrx; /* now try generic SOF status messages */ - status = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCD); - panic = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCX); + status = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCD); + panic = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCX); byt_get_registers(sdev, &xoops, &panic_info, stack, BYT_STACK_DUMP_SIZE); snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack, BYT_STACK_DUMP_SIZE); /* provide some context for firmware debug */ - imrx = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRX); - imrd = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRD); + imrx = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRX); + imrd = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRD); dev_err(sdev->dev, - "error: ipc host -> DSP: pending %s complete %s raw 0x%8.8x\n", + "error: ipc host -> DSP: pending %s complete %s raw 0x%llx\n", (panic & SHIM_IPCX_BUSY) ? "yes" : "no", (panic & SHIM_IPCX_DONE) ? "yes" : "no", panic); dev_err(sdev->dev, - "error: mask host: pending %s complete %s raw 0x%8.8x\n", + "error: mask host: pending %s complete %s raw 0x%llx\n", (imrx & SHIM_IMRX_BUSY) ? "yes" : "no", (imrx & SHIM_IMRX_DONE) ? "yes" : "no", imrx); dev_err(sdev->dev, - "error: ipc DSP -> host: pending %s complete %s raw 0x%8.8x\n", + "error: ipc DSP -> host: pending %s complete %s raw 0x%llx\n", (status & SHIM_IPCD_BUSY) ? "yes" : "no", (status & SHIM_IPCD_DONE) ? "yes" : "no", status); dev_err(sdev->dev, - "error: mask DSP: pending %s complete %s raw 0x%8.8x\n", + "error: mask DSP: pending %s complete %s raw 0x%llx\n", (imrd & SHIM_IMRD_BUSY) ? "yes" : "no", (imrd & SHIM_IMRD_DONE) ? "yes" : "no", imrd); diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 9a9a381a908d..432d12bd4937 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -50,8 +50,7 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) while (ext_hdr->hdr.cmd == SOF_IPC_FW_READY) { /* read in ext structure */ - offset += sizeof(*ext_hdr); - snd_sof_dsp_block_read(sdev, bar, offset, + snd_sof_dsp_block_read(sdev, bar, offset + sizeof(*ext_hdr), (void *)((u8 *)ext_data + sizeof(*ext_hdr)), ext_hdr->hdr.size - sizeof(*ext_hdr)); @@ -61,11 +60,15 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset) /* process structure data */ switch (ext_hdr->type) { case SOF_IPC_EXT_DMA_BUFFER: + ret = 0; break; case SOF_IPC_EXT_WINDOW: ret = get_ext_windows(sdev, ext_hdr); break; default: + dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n", + ext_hdr->type, ext_hdr->hdr.size); + ret = 0; break; } diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index d82ab981e840..e20b806ec80f 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3132,7 +3132,9 @@ found: case SOF_DAI_INTEL_SSP: case SOF_DAI_INTEL_DMIC: case SOF_DAI_INTEL_ALH: - /* no resource needs to be released for SSP, DMIC and ALH */ + case SOF_DAI_IMX_SAI: + case SOF_DAI_IMX_ESAI: + /* no resource needs to be released for all cases above */ break; case SOF_DAI_INTEL_HDA: ret = sof_link_hda_unload(sdev, link); |