diff options
Diffstat (limited to 'sound')
31 files changed, 184 insertions, 51 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 6163bf3e8177..2272aee12871 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -500,7 +500,8 @@ static int snd_compress_check_input(struct snd_compr_params *params) { /* first let's check the buffer parameter's */ if (params->buffer.fragment_size == 0 || - params->buffer.fragments > INT_MAX / params->buffer.fragment_size) + params->buffer.fragments > INT_MAX / params->buffer.fragment_size || + params->buffer.fragments == 0) return -EINVAL; /* now codec parameters */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6bda8f6c5f84..cdff5f976480 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/mutex.h> #include <linux/device.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/minors.h> #include <sound/pcm.h> @@ -125,6 +126,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -EFAULT; if (stream < 0 || stream > 1) return -EINVAL; + stream = array_index_nospec(stream, 2); if (get_user(subdevice, &info->subdevice)) return -EFAULT; mutex_lock(®ister_mutex); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5bc7ddf8fc70..3ce2b8771762 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1849,8 +1849,6 @@ int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg) { switch (cmd) { - case SNDRV_PCM_IOCTL1_INFO: - return 0; case SNDRV_PCM_IOCTL1_RESET: return snd_pcm_lib_ioctl_reset(substream, arg); case SNDRV_PCM_IOCTL1_CHANNEL_INFO: diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 0ad194002c0c..9b6dcdea4431 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -214,11 +214,7 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info) info->subdevices_avail = pstr->substream_count - pstr->substream_opened; strlcpy(info->subname, substream->name, sizeof(info->subname)); runtime = substream->runtime; - /* AB: FIXME!!! This is definitely nonsense */ - if (runtime) { - info->sync = runtime->sync; - substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_INFO, info); - } + return 0; } diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 091290d1f3ea..3a0361458597 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -382,7 +382,7 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */ SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal), /* Apogee Electronics, Ensemble */ - SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal), + SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal), /* ESI, Quatafire610 */ SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal), /* AcousticReality, eARMasterOne */ @@ -422,7 +422,19 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Focusrite, SaffirePro 26 I/O */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec), /* Focusrite, SaffirePro 10 I/O */ - SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec), + { + // The combination of vendor_id and model_id is the same as the + // same as the one of Liquid Saffire 56. + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VEN_FOCUSRITE, + .model_id = 0x000006, + .specifier_id = 0x00a02d, + .version = 0x010001, + .driver_data = (kernel_ulong_t)&saffirepro_10_spec, + }, /* Focusrite, Saffire(no label and LE) */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH, &saffire_spec), diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff5693d..718d5e3b7806 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index d2951ed4bf71..1984291ebd07 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -899,6 +899,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; + if (!ins) + return 0; + snd_info_free_entry(ins->proc_sym_info_entry); ins->proc_sym_info_entry = NULL; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 50b216fc369f..5d422d65e62b 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -36,6 +36,7 @@ #include <linux/init.h> #include <linux/mutex.h> #include <linux/moduleparam.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/tlv.h> @@ -1000,6 +1001,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); if (ipcm->channels > 32) return -EINVAL; pcm = &emu->fx8010.pcm[ipcm->substream]; @@ -1046,6 +1049,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; + ipcm->substream = array_index_nospec(ipcm->substream, + EMU10K1_FX8010_PCM_COUNT); pcm = &emu->fx8010.pcm[ipcm->substream]; mutex_lock(&emu->fx8010.lock); spin_lock_irq(&emu->reg_lock); diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 6efadbfb3fe3..7ea201c05e5d 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -109,7 +109,8 @@ static int hda_codec_driver_probe(struct device *dev) err = snd_hda_codec_build_controls(codec); if (err < 0) goto error_module; - if (codec->card->registered) { + /* only register after the bus probe finished; otherwise it's racy */ + if (!codec->bus->bus_probing && codec->card->registered) { err = snd_card_register(codec->card); if (err < 0) goto error_module; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 776dffa88aee..171e11be938d 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -68,6 +68,7 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ + unsigned int bus_probing :1; /* during probing process */ int primary_dig_out_type; /* primary digital out PCM type */ unsigned int mixer_assigned; /* codec addr for mixer name */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f964743b104c..74c9600876d6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2100,6 +2100,7 @@ static int azx_probe_continue(struct azx *chip) int val; int err; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; /* Request display power well for the HDA controller or codec. For @@ -2200,6 +2201,7 @@ i915_power_fail: if (err < 0) hda->init_failed = 1; complete_all(&hda->probe_wait); + to_hda_bus(bus)->bus_probing = 0; return err; } diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 17fd81736d3d..039fbbb1e53c 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -249,10 +249,12 @@ static int hda_tegra_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct hdac_bus *bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_stop_chip(chip); + synchronize_irq(bus->irq); azx_enter_link_reset(chip); hda_tegra_disable_clocks(hda); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index aea3cc2abe3a..40dd46556452 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -853,6 +853,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0467e5ba82e0..5d8ac2d798df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4792,6 +4792,13 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec, } } +static void alc_fixup_disable_mic_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -4891,6 +4898,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DISABLE_MIC_VREF, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC295_FIXUP_DISABLE_DAC3, ALC280_FIXUP_HP_HEADSET_MIC, @@ -5546,6 +5554,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DISABLE_MIC_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_mic_vref, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -5555,7 +5569,7 @@ static const struct hda_fixup alc269_fixups[] = { {} }, .chained = true, - .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + .chain_id = ALC225_FIXUP_DISABLE_MIC_VREF }, [ALC280_FIXUP_HP_HEADSET_MIC] = { .type = HDA_FIXUP_FUNC, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7c8941b8b2de..dd6c9e6a1d53 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -30,6 +30,7 @@ #include <linux/math64.h> #include <linux/vmalloc.h> #include <linux/io.h> +#include <linux/nospec.h> #include <sound/core.h> #include <sound/control.h> @@ -4065,15 +4066,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream, struct snd_pcm_channel_info *info) { struct hdsp *hdsp = snd_pcm_substream_chip(substream); - int mapped_channel; + unsigned int channel = info->channel; - if (snd_BUG_ON(info->channel >= hdsp->max_channels)) + if (snd_BUG_ON(channel >= hdsp->max_channels)) return -EINVAL; + channel = array_index_nospec(channel, hdsp->max_channels); - if ((mapped_channel = hdsp->channel_map[info->channel]) < 0) + if (hdsp->channel_map[channel] < 0) return -EINVAL; - info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES; + info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES; info->first = 0; info->step = 32; return 0; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7d60d5b03f63..fbb5b979f910 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -228,7 +228,7 @@ config SND_SOC_FSL_SAI_WM9712 config SND_SOC_EUKREA_TLV320 tristate "Eukrea TLV320" - depends on ARCH_MXC && I2C + depends on ARCH_MXC && !ARM64 && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index e8adead8be00..a87836d4de15 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -394,7 +394,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_RIGHT_J: /* Data on rising edge of bclk, frame high, right aligned */ - xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA; + xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP; + xcr |= ESAI_xCR_xWA; break; case SND_SOC_DAIFMT_DSP_A: /* Data on rising edge of bclk, frame high, 1clk before data */ @@ -451,12 +452,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR; + mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr); mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP | - ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA; + ESAI_xCCR_xFSD | ESAI_xCCR_xCKD; regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index fc57da341d61..136df38c4536 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; - ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", + ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS output from %s, ", audmux_port_string((ptcr >> 27) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk output from %s", audmux_port_string((ptcr >> 22) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "TxClk input"); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) { - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "Port is symmetric"); } else { if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS output from %s, ", audmux_port_string((ptcr >> 17) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxFS input, "); if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk output from %s", audmux_port_string((ptcr >> 12) & 0x7)); else - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "RxClk input"); } - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\nData received from %s\n", audmux_port_string((pdcr >> 13) & 0x7)); diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 2b96b11fbe71..1d9dfb92b3b4 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -398,7 +398,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + int ret; + + ret = + snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); + if (ret) + return ret; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); return 0; } diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 33917146d9c4..054b1d514e8a 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst) const struct firmware *fw; retval = request_firmware(&fw, sst->firmware_name, sst->dev); - if (fw == NULL) { - dev_err(sst->dev, "fw is returning as null\n"); - return -EINVAL; - } if (retval) { dev_err(sst->dev, "request fw failed %d\n", retval); return retval; } + if (fw == NULL) { + dev_err(sst->dev, "fw is returning as null\n"); + return -EINVAL; + } mutex_lock(&sst->sst_lock); retval = sst_cache_and_parse_fw(sst, fw); mutex_unlock(&sst->sst_lock); diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 3f8a1e10bed0..e5ca41ffa890 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -191,7 +191,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 22558572cb9c..de955c2e8c4e 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { .stream_name = "Loopback", .cpu_dai_name = "Loopback Pin", .platform_name = "haswell-pcm-audio", - .dynamic = 0, + .dynamic = 1, .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 09db2aec12a3..776e809a8aab 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -48,6 +48,8 @@ struct omap_dmic { struct device *dev; void __iomem *io_base; struct clk *fclk; + struct pm_qos_request pm_qos_req; + int latency; int fclk_freq; int out_freq; int clk_div; @@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream, mutex_lock(&dmic->mutex); + pm_qos_remove_request(&dmic->pm_qos_req); + if (!dai->active) dmic->active = 0; @@ -226,6 +230,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, /* packet size is threshold * channels */ dma_data = snd_soc_dai_get_dma_data(dai, substream); dma_data->maxburst = dmic->threshold * channels; + dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC / + params_rate(params); return 0; } @@ -236,6 +242,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream, struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai); u32 ctrl; + if (pm_qos_request_active(&dmic->pm_qos_req)) + pm_qos_update_request(&dmic->pm_qos_req, dmic->latency); + /* Configure uplink threshold */ omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold); diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 8d0d45d330e7..8eb2d12b6a34 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -54,6 +54,8 @@ struct omap_mcpdm { unsigned long phys_base; void __iomem *io_base; int irq; + struct pm_qos_request pm_qos_req; + int latency[2]; struct mutex mutex; @@ -273,6 +275,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; mutex_lock(&mcpdm->mutex); @@ -285,6 +290,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream, } } + if (mcpdm->latency[stream2]) + pm_qos_update_request(&mcpdm->pm_qos_req, + mcpdm->latency[stream2]); + else if (mcpdm->latency[stream1]) + pm_qos_remove_request(&mcpdm->pm_qos_req); + + mcpdm->latency[stream1] = 0; + mutex_unlock(&mcpdm->mutex); } @@ -296,7 +309,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; struct snd_dmaengine_dai_dma_data *dma_data; u32 threshold; - int channels; + int channels, latency; int link_mask = 0; channels = params_channels(params); @@ -336,14 +349,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, dma_data->maxburst = (MCPDM_DN_THRES_MAX - threshold) * channels; + latency = threshold; } else { /* If playback is not running assume a stereo stream to come */ if (!mcpdm->config[!stream].link_mask) mcpdm->config[!stream].link_mask = (0x3 << 3); dma_data->maxburst = threshold * channels; + latency = (MCPDM_DN_THRES_MAX - threshold); } + /* + * The DMA must act to a DMA request within latency time (usec) to avoid + * under/overflow + */ + mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params); + + if (!mcpdm->latency[stream]) + mcpdm->latency[stream] = 10; + /* Check if we need to restart McPDM with this stream */ if (mcpdm->config[stream].link_mask && mcpdm->config[stream].link_mask != link_mask) @@ -358,6 +382,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req; + int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; + int latency = mcpdm->latency[stream2]; + + /* Prevent omap hardware from hitting off between FIFO fills */ + if (!latency || mcpdm->latency[stream1] < latency) + latency = mcpdm->latency[stream1]; + + if (pm_qos_request_active(pm_qos_req)) + pm_qos_update_request(pm_qos_req, latency); + else if (latency) + pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency); if (!omap_mcpdm_active(mcpdm)) { omap_mcpdm_start(mcpdm); @@ -419,6 +457,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); + if (pm_qos_request_active(&mcpdm->pm_qos_req)) + pm_qos_remove_request(&mcpdm->pm_qos_req); + return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fa6b74a304a7..b927f9c81d92 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1711,6 +1711,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } card->instantiated = 1; + dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0aefed8ab0cf..7e26d173da41 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1943,19 +1943,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file, out = is_connected_output_ep(w, NULL); } - ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", + ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", w->name, w->power ? "On" : "Off", w->force ? " (forced)" : "", in, out); if (w->reg >= 0) - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " - R%d(0x%x) mask 0x%x", w->reg, w->reg, w->mask << w->shift); - ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n"); + ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n"); if (w->sname) - ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n", w->sname, w->active ? "active" : "inactive"); @@ -1968,7 +1968,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!p->connect) continue; - ret += snprintf(buf + ret, PAGE_SIZE - ret, + ret += scnprintf(buf + ret, PAGE_SIZE - ret, " %s \"%s\" \"%s\"\n", (rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out", p->name ? p->name : "static", diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index c1e76feb3529..824f4d7fc41f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1770,6 +1770,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id) { struct soc_tplg tplg; + int ret; /* setup parsing context */ memset(&tplg, 0, sizeof(tplg)); @@ -1783,7 +1784,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.bytes_ext_ops = ops->bytes_ext_ops; tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; - return soc_tplg_load(&tplg); + ret = soc_tplg_load(&tplg); + /* free the created components if fail to load topology */ + if (ret) + snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load); diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index e557946718a9..d9fcae071b47 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -22,9 +22,9 @@ #include <sound/core.h> #include <sound/hwdep.h> #include <linux/uaccess.h> +#include <linux/nospec.h> #include "emux_voice.h" - #define TMP_CLIENT_ID 0x1001 /* @@ -66,13 +66,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg) return -EFAULT; if (info.mode < 0 || info.mode >= EMUX_MD_END) return -EINVAL; + info.mode = array_index_nospec(info.mode, EMUX_MD_END); if (info.port < 0) { for (i = 0; i < emu->num_ports; i++) emu->portptrs[i]->ctrls[info.mode] = info.value; } else { - if (info.port < emu->num_ports) + if (info.port < emu->num_ports) { + info.port = array_index_nospec(info.port, emu->num_ports); emu->portptrs[info.port]->ctrls[info.mode] = info.value; + } } return 0; } diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 97d6a18e6956..f7eb0d2f797b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1816,7 +1816,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, char *name) { struct uac_processing_unit_descriptor *desc = raw_desc; - int num_ins = desc->bNrInPins; + int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; int i, err, nameid, type, len; @@ -1831,7 +1831,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, 0, NULL, default_value_info }; - if (desc->bLength < 13 || desc->bLength < 13 + num_ins || + if (desc->bLength < 13) { + usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); + return -EINVAL; + } + + num_ins = desc->bNrInPins; + if (desc->bLength < 13 + num_ins || desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) { usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid); return -EINVAL; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a9079654107c..1ea1384bc236 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -313,6 +313,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } +/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk + * applies. Returns 1 if a quirk was found. + */ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, struct usb_device *dev, struct usb_interface_descriptor *altsd, @@ -381,7 +384,7 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; - return 0; + return 1; } static int set_sync_endpoint(struct snd_usb_substream *subs, @@ -420,6 +423,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, if (err < 0) return err; + /* endpoint set by quirk */ + if (err > 0) + return 0; + if (altsd->bNumEndpoints < 2) return 0; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 15cbe2565703..d32727c74a16 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3321,6 +3321,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, + { + .ifnum = -1 + }, } } }, |