diff options
Diffstat (limited to 'sound')
61 files changed, 1095 insertions, 649 deletions
diff --git a/sound/core/info.c b/sound/core/info.c index 051d55b05521..9f404e965ea2 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -684,7 +684,7 @@ int snd_info_card_free(struct snd_card *card) * snd_info_get_line - read one line from the procfs buffer * @buffer: the procfs buffer * @line: the buffer to store - * @len: the max. buffer size - 1 + * @len: the max. buffer size * * Reads one line from the buffer and stores the string. * @@ -704,7 +704,7 @@ int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) buffer->stop = 1; if (c == '\n') break; - if (len) { + if (len > 1) { len--; *line++ = c; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9acc77eae487..0032278567ad 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1782,14 +1782,16 @@ static int snd_pcm_lib_ioctl_fifo_size(struct snd_pcm_substream *substream, { struct snd_pcm_hw_params *params = arg; snd_pcm_format_t format; - int channels, width; + int channels; + ssize_t frame_size; params->fifo_size = substream->runtime->hw.fifo_size; if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_FIFO_IN_FRAMES)) { format = params_format(params); channels = params_channels(params); - width = snd_pcm_format_physical_width(format); - params->fifo_size /= width * channels; + frame_size = snd_pcm_format_size(format, channels); + if (frame_size > 0) + params->fifo_size /= (unsigned)frame_size; } return 0; } diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 4560ca0e5651..2c6fd80e0bd1 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -142,11 +142,11 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { }, [SNDRV_PCM_FORMAT_DSD_U8] = { .width = 8, .phys = 8, .le = 1, .signd = 0, - .silence = {}, + .silence = { 0x69 }, }, [SNDRV_PCM_FORMAT_DSD_U16_LE] = { .width = 16, .phys = 16, .le = 1, .signd = 0, - .silence = {}, + .silence = { 0x69, 0x69 }, }, /* FIXME: the following three formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index f96bf4c7c232..95fc2eaf11dc 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -507,7 +507,16 @@ static void amdtp_pull_midi(struct amdtp_stream *s, static void update_pcm_pointers(struct amdtp_stream *s, struct snd_pcm_substream *pcm, unsigned int frames) -{ unsigned int ptr; +{ + unsigned int ptr; + + /* + * In IEC 61883-6, one data block represents one event. In ALSA, one + * event equals to one PCM frame. But Dice has a quirk to transfer + * two PCM frames in one data block. + */ + if (s->double_pcm_frames) + frames *= 2; ptr = s->pcm_buffer_pointer + frames; if (ptr >= pcm->runtime->buffer_size) diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index d8ee7b0e9386..4823c08196ac 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -125,6 +125,7 @@ struct amdtp_stream { unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; bool pointer_flush; + bool double_pcm_frames; struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8]; diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index a9a30c0161f1..e3a04d69c853 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -567,10 +567,14 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; /* - * At rates above 96 kHz, pretend that the stream runs at half the - * actual sample rate with twice the number of channels; two samples - * of a channel are stored consecutively in the packet. Requires - * blocking mode and PCM buffer size should be aligned to SYT_INTERVAL. + * At 176.4/192.0 kHz, Dice has a quirk to transfer two PCM frames in + * one data block of AMDTP packet. Thus sampling transfer frequency is + * a half of PCM sampling frequency, i.e. PCM frames at 192.0 kHz are + * transferred on AMDTP packets at 96 kHz. Two successive samples of a + * channel are stored consecutively in the packet. This quirk is called + * as 'Dual Wire'. + * For this quirk, blocking mode is required and PCM buffer size should + * be aligned to SYT_INTERVAL. */ channels = params_channels(hw_params); if (rate_index > 4) { @@ -579,18 +583,25 @@ static int dice_hw_params(struct snd_pcm_substream *substream, return err; } - for (i = 0; i < channels; i++) { - dice->stream.pcm_positions[i * 2] = i; - dice->stream.pcm_positions[i * 2 + 1] = i + channels; - } - rate /= 2; channels *= 2; + dice->stream.double_pcm_frames = true; + } else { + dice->stream.double_pcm_frames = false; } mode = rate_index_to_mode(rate_index); amdtp_stream_set_parameters(&dice->stream, rate, channels, dice->rx_midi_ports[mode]); + if (rate_index > 4) { + channels /= 2; + + for (i = 0; i < channels; i++) { + dice->stream.pcm_positions[i] = i * 2; + dice->stream.pcm_positions[i + channels] = i * 2 + 1; + } + } + amdtp_stream_set_pcm_format(&dice->stream, params_format(hw_params)); diff --git a/sound/pci/ctxfi/ct20k1reg.h b/sound/pci/ctxfi/ct20k1reg.h index f2e34e3f27ee..5851249f11d9 100644 --- a/sound/pci/ctxfi/ct20k1reg.h +++ b/sound/pci/ctxfi/ct20k1reg.h @@ -7,7 +7,7 @@ */ #ifndef CT20K1REG_H -#define CT20k1REG_H +#define CT20K1REG_H /* 20k1 registers */ #define DSPXRAM_START 0x000000 @@ -632,5 +632,3 @@ #define I2SD_R 0x19L #endif /* CT20K1REG_H */ - - diff --git a/sound/pci/hda/ca0132_regs.h b/sound/pci/hda/ca0132_regs.h index 07e760937d3c..8371274aa811 100644 --- a/sound/pci/hda/ca0132_regs.h +++ b/sound/pci/hda/ca0132_regs.h @@ -20,7 +20,7 @@ */ #ifndef __CA0132_REGS_H -#define __CA0312_REGS_H +#define __CA0132_REGS_H #define DSP_CHIP_OFFSET 0x100000 #define DSP_DBGCNTL_MODULE_OFFSET 0xE30 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6f2fa838b635..47ccb8f44adb 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -217,6 +217,7 @@ enum { CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, CXT_FIXUP_GPIO1, + CXT_FIXUP_ASPIRE_DMIC, CXT_FIXUP_THINKPAD_ACPI, CXT_FIXUP_OLPC_XO, CXT_FIXUP_CAP_MIX_AMP, @@ -664,6 +665,12 @@ static const struct hda_fixup cxt_fixups[] = { { } }, }, + [CXT_FIXUP_ASPIRE_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_stereo_dmic, + .chained = true, + .chain_id = CXT_FIXUP_GPIO1, + }, [CXT_FIXUP_THINKPAD_ACPI] = { .type = HDA_FIXUP_FUNC, .v.func = hda_fixup_thinkpad_acpi, @@ -744,7 +751,7 @@ static const struct hda_model_fixup cxt5051_fixup_models[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), @@ -770,6 +777,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_PINCFG_LENOVO_TP410, .name = "tp410" }, { .id = CXT_FIXUP_THINKPAD_ACPI, .name = "thinkpad" }, { .id = CXT_PINCFG_LEMOTE_A1004, .name = "lemote-a1004" }, + { .id = CXT_PINCFG_LEMOTE_A1205, .name = "lemote-a1205" }, { .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" }, {} }; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 36badba2dcec..99d7d7fecaad 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -50,6 +50,8 @@ MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); #define is_haswell_plus(codec) (is_haswell(codec) || is_broadwell(codec)) #define is_valleyview(codec) ((codec)->vendor_id == 0x80862882) +#define is_cherryview(codec) ((codec)->vendor_id == 0x80862883) +#define is_valleyview_plus(codec) (is_valleyview(codec) || is_cherryview(codec)) struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; @@ -1459,7 +1461,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview(codec)) + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1598,7 +1600,8 @@ static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || + is_valleyview_plus(codec)) { intel_verify_pin_cvt_connect(codec, per_pin); intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); @@ -1779,7 +1782,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, bool non_pcm; int pinctl; - if (is_haswell_plus(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { /* Verify pin:cvt selections to avoid silent audio after S3. * After S3, the audio driver restores pin:cvt selections * but this can happen before gfx is ready and such selection @@ -2330,9 +2333,8 @@ static int patch_generic_hdmi(struct hda_codec *codec) intel_haswell_fixup_enable_dp12(codec); } - if (is_haswell(codec) || is_valleyview(codec)) { + if (is_haswell_plus(codec) || is_valleyview_plus(codec)) codec->depop_delay = 0; - } if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6b38ec3c6e57..1ba22fb527c2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -181,6 +181,8 @@ static void alc_fix_pll(struct hda_codec *codec) spec->pll_coef_idx); val = snd_hda_codec_read(codec, spec->pll_nid, 0, AC_VERB_GET_PROC_COEF, 0); + if (val == -1) + return; snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_COEF_INDEX, spec->pll_coef_idx); snd_hda_codec_write(codec, spec->pll_nid, 0, AC_VERB_SET_PROC_COEF, @@ -326,6 +328,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0885: case 0x10ec0887: /*case 0x10ec0889:*/ /* this causes an SPDIF problem */ + case 0x10ec0900: alc889_coef_init(codec); break; case 0x10ec0888: @@ -2348,6 +2351,7 @@ static int patch_alc882(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0882: case 0x10ec0885: + case 0x10ec0900: break; default: /* ALC883 and variants */ @@ -2806,6 +2810,8 @@ static void alc286_shutup(struct hda_codec *codec) static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { int val = alc_read_coef_idx(codec, 0x04); + if (val == -1) + return; if (power_up) val |= 1 << 11; else @@ -3264,6 +3270,15 @@ static int alc269_resume(struct hda_codec *codec) snd_hda_codec_resume_cache(codec); alc_inv_dmic_sync(codec, true); hda_call_check_power_status(codec, 0x01); + + /* on some machine, the BIOS will clear the codec gpio data when enter + * suspend, and won't restore the data after resume, so we restore it + * in the driver. + */ + if (spec->gpio_led) + snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_led); + if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); @@ -4395,6 +4410,7 @@ enum { ALC292_FIXUP_TPT440_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, + ALC282_FIXUP_ASPIRE_V5_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -4842,6 +4858,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained_before = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC282_FIXUP_ASPIRE_V5_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x90a60130 }, + { 0x14, 0x90170110 }, + { 0x17, 0x40000008 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40f89b2d }, + { 0x1e, 0x411111f0 }, + { 0x21, 0x0321101f }, + { }, + }, + }, }; @@ -4853,6 +4885,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), + SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -5311,27 +5344,30 @@ static void alc269_fill_coef(struct hda_codec *codec) if ((alc_get_coef0(codec) & 0x00ff) == 0x017) { val = alc_read_coef_idx(codec, 0x04); /* Power up output pin */ - alc_write_coef_idx(codec, 0x04, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x04, val | (1<<11)); } if ((alc_get_coef0(codec) & 0x00ff) == 0x018) { val = alc_read_coef_idx(codec, 0xd); - if ((val & 0x0c00) >> 10 != 0x1) { + if (val != -1 && (val & 0x0c00) >> 10 != 0x1) { /* Capless ramp up clock control */ alc_write_coef_idx(codec, 0xd, val | (1<<10)); } val = alc_read_coef_idx(codec, 0x17); - if ((val & 0x01c0) >> 6 != 0x4) { + if (val != -1 && (val & 0x01c0) >> 6 != 0x4) { /* Class D power on reset */ alc_write_coef_idx(codec, 0x17, val | (1<<7)); } } val = alc_read_coef_idx(codec, 0xd); /* Class D */ - alc_write_coef_idx(codec, 0xd, val | (1<<14)); + if (val != -1) + alc_write_coef_idx(codec, 0xd, val | (1<<14)); val = alc_read_coef_idx(codec, 0x4); /* HP */ - alc_write_coef_idx(codec, 0x4, val | (1<<11)); + if (val != -1) + alc_write_coef_idx(codec, 0x4, val | (1<<11)); } /* diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea823e1100da..98cd1908c039 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -566,8 +566,8 @@ static void stac_init_power_map(struct hda_codec *codec) if (snd_hda_jack_tbl_get(codec, nid)) continue; if (def_conf == AC_JACK_PORT_COMPLEX && - !(spec->vref_mute_led_nid == nid || - is_jack_detectable(codec, nid))) { + spec->vref_mute_led_nid != nid && + is_jack_detectable(codec, nid)) { snd_hda_jack_detect_enable_callback(codec, nid, STAC_PWR_EVENT, jack_update_power); @@ -4276,11 +4276,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) return err; } - stac_init_power_map(codec); - return 0; } +static int stac_build_controls(struct hda_codec *codec) +{ + int err = snd_hda_gen_build_controls(codec); + + if (err < 0) + return err; + stac_init_power_map(codec); + return 0; +} static int stac_init(struct hda_codec *codec) { @@ -4392,7 +4399,7 @@ static int stac_suspend(struct hda_codec *codec) #endif /* CONFIG_PM */ static const struct hda_codec_ops stac_patch_ops = { - .build_controls = snd_hda_gen_build_controls, + .build_controls = stac_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = stac_init, .free = stac_free, diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index bd41ee4da078..2c71f16bd661 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, else rates = &arizona_48k_bclk_rates[0]; + wl = snd_pcm_format_width(params_format(params)); + if (tdm_slots) { arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", tdm_slots, tdm_width); @@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, channels = tdm_slots; } else { bclk_target = snd_soc_params_to_bclk(params); + tdm_width = wl; } if (chan_limit && chan_limit < channels) { @@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", rates[bclk], rates[bclk] / lrclk); - wl = snd_pcm_format_width(params_format(params)); - frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width; reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index a20b30ca52c0..69a85164357c 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -282,10 +282,10 @@ static const struct cs4265_clk_para clk_map_table[] = { /*64k*/ {8192000, 64000, 1, 0}, - {1228800, 64000, 1, 1}, - {1693440, 64000, 1, 2}, - {2457600, 64000, 1, 3}, - {3276800, 64000, 1, 4}, + {12288000, 64000, 1, 1}, + {16934400, 64000, 1, 2}, + {24576000, 64000, 1, 3}, + {32768000, 64000, 1, 4}, /* 88.2k */ {11289600, 88200, 1, 0}, @@ -435,10 +435,10 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, index = cs4265_get_clk_index(cs4265->sysclk, params_rate(params)); if (index >= 0) { snd_soc_update_bits(codec, CS4265_ADC_CTL, - CS4265_ADC_FM, clk_map_table[index].fm_mode); + CS4265_ADC_FM, clk_map_table[index].fm_mode << 6); snd_soc_update_bits(codec, CS4265_MCLK_FREQ, CS4265_MCLK_FREQ_MASK, - clk_map_table[index].mclkdiv); + clk_map_table[index].mclkdiv << 4); } else { dev_err(codec->dev, "can't get correct mclk\n"); @@ -458,12 +458,12 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, if (params_width(params) == 16) { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (1 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } else { snd_soc_update_bits(codec, CS4265_DAC_CTL, CS4265_DAC_CTL_DIF, (3 << 5)); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 7)); } break; @@ -472,7 +472,7 @@ static int cs4265_pcm_hw_params(struct snd_pcm_substream *substream, CS4265_DAC_CTL_DIF, 0); snd_soc_update_bits(codec, CS4265_ADC_CTL, CS4265_ADC_DIF, 0); - snd_soc_update_bits(codec, CS4265_ADC_CTL, + snd_soc_update_bits(codec, CS4265_SPDIF_CTL2, CS4265_SPDIF_CTL2_DIF, (1 << 6)); break; diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h index 1dceafeec415..f586cbd30b77 100644 --- a/sound/soc/codecs/da732x.h +++ b/sound/soc/codecs/da732x.h @@ -11,7 +11,7 @@ */ #ifndef __DA732X_H_ -#define __DA732X_H +#define __DA732X_H_ #include <sound/soc.h> diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 4a063fa88526..7e111865946a 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1311,8 +1311,6 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, - {"DMICL", NULL, "DMICL_ENA"}, - {"DMICR", NULL, "DMICR_ENA"}, {"DMICL", NULL, "AHPF"}, {"DMICR", NULL, "AHPF"}, @@ -1370,6 +1368,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"DMIC Mux", "ADC", "ADCR"}, {"DMIC Mux", "DMIC", "DMICL"}, {"DMIC Mux", "DMIC", "DMICR"}, + {"DMIC Mux", "DMIC", "DMICL_ENA"}, + {"DMIC Mux", "DMIC", "DMICR_ENA"}, {"LBENL Mux", "Normal", "DMIC Mux"}, {"LBENL Mux", "Loopback", "LTENL Mux"}, @@ -1972,6 +1972,102 @@ static int max98090_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } +static int max98090_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!max98090->master && dai->active == 1) + queue_delayed_work(system_power_efficient_wq, + &max98090->pll_det_enable_work, + msecs_to_jiffies(10)); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!max98090->master && dai->active == 1) + schedule_work(&max98090->pll_det_disable_work); + break; + default: + break; + } + + return 0; +} + +static void max98090_pll_det_enable_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, + pll_det_enable_work.work); + struct snd_soc_codec *codec = max98090->codec; + unsigned int status, mask; + + /* + * Clear status register in order to clear possibly already occurred + * PLL unlock. If PLL hasn't still locked, the status will be set + * again and PLL unlock interrupt will occur. + * Note this will clear all status bits + */ + regmap_read(max98090->regmap, M98090_REG_DEVICE_STATUS, &status); + + /* + * Queue jack work in case jack state has just changed but handler + * hasn't run yet + */ + regmap_read(max98090->regmap, M98090_REG_INTERRUPT_S, &mask); + status &= mask; + if (status & M98090_JDET_MASK) + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); + + /* Enable PLL unlock interrupt */ + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IULK_MASK, + 1 << M98090_IULK_SHIFT); +} + +static void max98090_pll_det_disable_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, pll_det_disable_work); + struct snd_soc_codec *codec = max98090->codec; + + cancel_delayed_work_sync(&max98090->pll_det_enable_work); + + /* Disable PLL unlock interrupt */ + snd_soc_update_bits(codec, M98090_REG_INTERRUPT_S, + M98090_IULK_MASK, 0); +} + +static void max98090_pll_work(struct work_struct *work) +{ + struct max98090_priv *max98090 = + container_of(work, struct max98090_priv, pll_work); + struct snd_soc_codec *codec = max98090->codec; + + if (!snd_soc_codec_is_active(codec)) + return; + + dev_info(codec->dev, "PLL unlocked\n"); + + /* Toggle shutdown OFF then ON */ + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, 0); + msleep(10); + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, M98090_SHDNN_MASK); + + /* Give PLL time to lock */ + msleep(10); +} + static void max98090_jack_work(struct work_struct *work) { struct max98090_priv *max98090 = container_of(work, @@ -2103,8 +2199,10 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (active & M98090_SLD_MASK) dev_dbg(codec->dev, "M98090_SLD_MASK\n"); - if (active & M98090_ULK_MASK) - dev_err(codec->dev, "M98090_ULK_MASK\n"); + if (active & M98090_ULK_MASK) { + dev_dbg(codec->dev, "M98090_ULK_MASK\n"); + schedule_work(&max98090->pll_work); + } if (active & M98090_JDET_MASK) { dev_dbg(codec->dev, "M98090_JDET_MASK\n"); @@ -2177,6 +2275,7 @@ static struct snd_soc_dai_ops max98090_dai_ops = { .set_tdm_slot = max98090_set_tdm_slot, .hw_params = max98090_dai_hw_params, .digital_mute = max98090_dai_digital_mute, + .trigger = max98090_dai_trigger, }; static struct snd_soc_dai_driver max98090_dai[] = { @@ -2258,6 +2357,11 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); + INIT_DELAYED_WORK(&max98090->pll_det_enable_work, + 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_write(codec, M98090_REG_JACK_DETECT, @@ -2310,6 +2414,9 @@ static int max98090_remove(struct snd_soc_codec *codec) struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); 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); return 0; } diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index cf1b6062ba8c..14427a566f41 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1532,6 +1532,9 @@ struct max98090_priv { int irq; int jack_state; 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/pcm512x.c b/sound/soc/codecs/pcm512x.c index 163ec3855fd4..0c8aefab404c 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds = pcm512x_ramp_step_text); static const struct snd_kcontrol_new pcm512x_controls[] = { -SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, +SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, +SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, PCM512x_RQMR_SHIFT, 1, 1), SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e4f6102efc1a..b86b426f159d 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -51,7 +51,7 @@ static struct reg_default rt286_index_def[] = { { 0x04, 0xaf01 }, { 0x08, 0x000d }, { 0x09, 0xd810 }, - { 0x0a, 0x0060 }, + { 0x0a, 0x0120 }, { 0x0b, 0x0000 }, { 0x0d, 0x2800 }, { 0x0f, 0x0000 }, @@ -60,7 +60,7 @@ static struct reg_default rt286_index_def[] = { { 0x33, 0x0208 }, { 0x49, 0x0004 }, { 0x4f, 0x50e9 }, - { 0x50, 0x2c00 }, + { 0x50, 0x2000 }, { 0x63, 0x2902 }, { 0x67, 0x1111 }, { 0x68, 0x1016 }, @@ -104,7 +104,6 @@ static const struct reg_default rt286_reg[] = { { 0x02170700, 0x00000000 }, { 0x02270100, 0x00000000 }, { 0x02370100, 0x00000000 }, - { 0x02040000, 0x00004002 }, { 0x01870700, 0x00000020 }, { 0x00830000, 0x000000c3 }, { 0x00930000, 0x000000c3 }, @@ -192,7 +191,6 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) /*handle index registers*/ if (reg <= 0xff) { rt286_hw_write(client, RT286_COEF_INDEX, reg); - reg = RT286_PROC_COEF; for (i = 0; i < INDEX_CACHE_SIZE; i++) { if (reg == rt286->index_cache[i].reg) { rt286->index_cache[i].def = value; @@ -200,6 +198,7 @@ static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) } } + reg = RT286_PROC_COEF; } data[0] = (reg >> 24) & 0xff; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6bc6efdec550..f1ec6e6bd08a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2059,6 +2059,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { static const struct regmap_config rt5640_regmap = { .reg_bits = 8, .val_bits = 16, + .use_single_rw = true, .max_register = RT5640_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5640_ranges) * RT5640_PR_SPACING), diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 67f14556462f..5337c448b5e3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -2135,10 +2135,10 @@ static const struct snd_soc_dapm_route rt5677_dapm_routes[] = { { "BST2", NULL, "IN2P" }, { "BST2", NULL, "IN2N" }, - { "IN1P", NULL, "micbias1" }, - { "IN1N", NULL, "micbias1" }, - { "IN2P", NULL, "micbias1" }, - { "IN2N", NULL, "micbias1" }, + { "IN1P", NULL, "MICBIAS1" }, + { "IN1N", NULL, "MICBIAS1" }, + { "IN2P", NULL, "MICBIAS1" }, + { "IN2N", NULL, "MICBIAS1" }, { "ADC 1", NULL, "BST1" }, { "ADC 1", NULL, "ADC 1 power" }, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 484b3bbe8624..4021cd435740 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -647,7 +647,7 @@ int ssm2602_probe(struct device *dev, enum ssm2602_type type, return -ENOMEM; dev_set_drvdata(dev, ssm2602); - ssm2602->type = SSM2602; + ssm2602->type = type; ssm2602->regmap = regmap; return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 9aa1323fb2ab..89c748dd3d6e 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -4,7 +4,7 @@ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * Rajeev Kumar <rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -426,5 +426,5 @@ static struct i2c_driver sta529_i2c_driver = { module_i2c_driver(sta529_i2c_driver); MODULE_DESCRIPTION("ASoC STA529 codec driver"); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 0f64c7890eed..aea9e1ff9126 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -189,46 +189,57 @@ static const struct aic31xx_rate_divs aic31xx_divs[] = { /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ /* 8k rate */ {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {12000000, 8000, 1, 8, 1920, 128, 32, 3, 128, 32, 3}, {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, /* 11.025k rate */ {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {12000000, 11025, 1, 8, 4672, 128, 24, 3, 128, 24, 3}, {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, /* 16k rate */ {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {12000000, 16000, 1, 8, 1920, 128, 16, 3, 128, 16, 3}, {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, /* 22.05k rate */ {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {12000000, 22050, 1, 8, 4672, 128, 12, 3, 128, 12, 3}, {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, /* 32k rate */ {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {12000000, 32000, 1, 8, 1920, 128, 8, 3, 128, 8, 3}, {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, /* 44.1k rate */ {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {12000000, 44100, 1, 8, 4672, 128, 6, 3, 128, 6, 3}, {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, /* 48k rate */ {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {12000000, 48000, 1, 7, 6800, 96, 5, 4, 96, 5, 4}, {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, /* 88.2k rate */ {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {12000000, 88200, 1, 8, 4672, 64, 6, 3, 64, 6, 3}, {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, /* 96k rate */ {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {12000000, 96000, 1, 7, 6800, 48, 5, 4, 48, 5, 4}, {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, /* 176.4k rate */ {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {12000000, 176400, 1, 8, 4672, 32, 6, 3, 32, 6, 3}, {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, /* 192k rate */ {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {12000000, 192000, 1, 7, 6800, 24, 5, 4, 24, 5, 4}, {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, }; @@ -680,7 +691,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, struct snd_pcm_hw_params *params) { struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_score = snd_soc_params_to_frame_size(params); int bclk_n = 0; + int match = -1; int i; /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ @@ -691,15 +704,37 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { if (aic31xx_divs[i].rate == params_rate(params) && - aic31xx_divs[i].mclk == aic31xx->sysclk) - break; + aic31xx_divs[i].mclk == aic31xx->sysclk) { + int s = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) % + snd_soc_params_to_frame_size(params); + int bn = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) / + snd_soc_params_to_frame_size(params); + if (s < bclk_score && bn > 0) { + match = i; + bclk_n = bn; + bclk_score = s; + } + } } - if (i == ARRAY_SIZE(aic31xx_divs)) { - dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + if (match == -1) { + dev_err(codec->dev, + "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); + /* See bellow for details how fix this. */ return -EINVAL; } + if (bclk_score != 0) { + dev_warn(codec->dev, "Can not produce exact bitclock"); + /* This is fine if using dsp format, but if using i2s + there may be trouble. To fix the issue edit the + aic31xx_divs table for your mclk and sample + rate. Details can be found from: + http://www.ti.com/lit/ds/symlink/tlv320aic3100.pdf + Section: 5.6 CLOCK Generation and PLL + */ + } + i = match; /* PLL configuration */ snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, @@ -729,14 +764,6 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); /* Bit clock divider configuration. */ - bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) - / snd_soc_params_to_frame_size(params); - if (bclk_n == 0) { - dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", - __func__); - return -EINVAL; - } - snd_soc_update_bits(codec, AIC31XX_BCLKN, AIC31XX_PLL_MASK, bclk_n); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 3dfdcc4197fa..628ec774cf22 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -212,7 +212,7 @@ static void wm8350_pga_work(struct work_struct *work) { struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8350_data *wm8350_data = snd_soc_codec_get_drvdata(codec); struct wm8350_output *out1 = &wm8350_data->out1, *out2 = &wm8350_data->out2; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e54e097f4fcb..21ca3a94fc96 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1433,7 +1433,7 @@ static void wm8753_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8753_set_bias_level(codec, dapm->bias_level); } diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 0499cd4cfb71..39ddb9b8834c 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -615,7 +615,7 @@ static void wm8971_work(struct work_struct *work) struct snd_soc_dapm_context *dapm = container_of(work, struct snd_soc_dapm_context, delayed_work.work); - struct snd_soc_codec *codec = dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); wm8971_set_bias_level(codec, codec->dapm.bias_level); } diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index d69510c53239..8e948c63f3d9 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -63,7 +63,8 @@ config SND_DM365_AIC3X_CODEC Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC - bool "Voice Codec - CQ93VC" + tristate "Voice Codec - CQ93VC" + depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c28508da34cf..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -403,7 +403,8 @@ out: return ret; } -static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div, bool explicit) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -420,7 +421,8 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRDIV(div - 1), ACLKRDIV_MASK); - mcasp->bclk_div = div; + if (explicit) + mcasp->bclk_div = div; break; case 2: /* BCLK/LRCLK ratio */ @@ -434,6 +436,12 @@ static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div return 0; } +static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, + int div) +{ + return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1); +} + static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { @@ -459,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() @@ -738,7 +755,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, "Inaccurate BCLK: %u Hz / %u != %u Hz\n", mcasp->sysclk_freq, div, bclk_freq); } - davinci_mcasp_set_clkdiv(cpu_dai, 1, div); + __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); } ret = mcasp_common_hw_param(mcasp, substream->stream, diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 25c31f1655f6..e961388e6e9c 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -4,7 +4,7 @@ * sound/soc/dwc/designware_i2s.c * * Copyright (C) 2010 ST Microelectronics - * Rajeev Kumar <rajeev-dlh.kumar@st.com> + * Rajeev Kumar <rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -455,7 +455,7 @@ static struct platform_driver dw_i2s_driver = { module_platform_driver(dw_i2s_driver); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:designware_i2s"); diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index f54a8fc99291..f3012b645b51 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -49,7 +49,6 @@ config SND_SOC_FSL_ESAI tristate "Enhanced Serial Audio Interface (ESAI) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n - select SND_SOC_FSL_UTILS help Say Y if you want to add Enhanced Synchronous Audio Interface (ESAI) support for the Freescale CPUs. diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 72d154e7dd03..a3b29ed84963 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -18,7 +18,6 @@ #include "fsl_esai.h" #include "imx-pcm.h" -#include "fsl_utils.h" #define FSL_ESAI_RATES SNDRV_PCM_RATE_8000_192000 #define FSL_ESAI_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ @@ -607,7 +606,6 @@ static struct snd_soc_dai_ops fsl_esai_dai_ops = { .hw_params = fsl_esai_hw_params, .set_sysclk = fsl_esai_set_dai_sysclk, .set_fmt = fsl_esai_set_dai_fmt, - .xlate_tdm_slot_mask = fsl_asoc_xlate_tdm_slot_mask, .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 87eb5776a39b..de6ab06f58a5 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -748,8 +748,9 @@ static int fsl_ssi_hw_free(struct snd_pcm_substream *substream, return 0; } -static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, - unsigned int fmt) +static int _fsl_ssi_set_dai_fmt(struct device *dev, + struct fsl_ssi_private *ssi_private, + unsigned int fmt) { struct regmap *regs = ssi_private->regs; u32 strcr = 0, stcr, srcr, scr, mask; @@ -758,7 +759,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi_private *ssi_private, ssi_private->dai_fmt = fmt; if (fsl_ssi_is_i2s_master(ssi_private) && IS_ERR(ssi_private->baudclk)) { - dev_err(&ssi_private->pdev->dev, "baudclk is missing which is necessary for master mode\n"); + dev_err(dev, "baudclk is missing which is necessary for master mode\n"); return -EINVAL; } @@ -913,7 +914,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); - return _fsl_ssi_set_dai_fmt(ssi_private, fmt); + return _fsl_ssi_set_dai_fmt(cpu_dai->dev, ssi_private, fmt); } /** @@ -1387,7 +1388,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) done: if (ssi_private->dai_fmt) - _fsl_ssi_set_dai_fmt(ssi_private, ssi_private->dai_fmt); + _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, + ssi_private->dai_fmt); return 0; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 159e517fa09a..cef7776b712c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -481,12 +481,19 @@ static int asoc_simple_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(&priv->snd_card, priv); ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); + if (ret >= 0) + return ret; err: asoc_simple_card_unref(pdev); return ret; } +static int asoc_simple_card_remove(struct platform_device *pdev) +{ + return asoc_simple_card_unref(pdev); +} + static const struct of_device_id asoc_simple_of_match[] = { { .compatible = "simple-audio-card", }, {}, @@ -500,6 +507,7 @@ static struct platform_driver asoc_simple_card = { .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, + .remove = asoc_simple_card_remove, }; module_platform_driver(asoc_simple_card); diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 7acbfc43a0c6..f841786dad15 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -2,7 +2,8 @@ snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o snd-soc-sst-acpi-objs := sst-acpi.o -snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o sst-mfld-platform-compress.o +snd-soc-sst-mfld-platform-objs := sst-mfld-platform-pcm.o \ + sst-mfld-platform-compress.o sst-atom-controls.o snd-soc-mfld-machine-objs := mfld_machine.o obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c index 42edc6f4fc4a..03d0a166b635 100644 --- a/sound/soc/intel/sst-acpi.c +++ b/sound/soc/intel/sst-acpi.c @@ -246,8 +246,8 @@ static struct sst_acpi_desc sst_acpi_broadwell_desc = { }; static struct sst_acpi_mach baytrail_machines[] = { - { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, - { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-i2s_master" }, + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, + { "193C9890", "byt-max98090", "intel/fw_sst_0f28.bin-48kHz_i2s_master" }, {} }; diff --git a/sound/soc/intel/sst-atom-controls.c b/sound/soc/intel/sst-atom-controls.c new file mode 100644 index 000000000000..ace3c4a59b14 --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.c @@ -0,0 +1,39 @@ +/* + * sst-atom-controls.c - Intel MID Platform driver DPCM ALSA controls for Mrfld + * + * Copyright (C) 2013-14 Intel Corp + * Author: Omair Mohammed Abdullah <omair.m.abdullah@intel.com> + * Vinod Koul <vinod.koul@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + */ +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "sst-mfld-platform.h" +#include "sst-atom-controls.h" + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform) +{ + int ret = 0; + struct sst_data *drv = snd_soc_platform_get_drvdata(platform); + + drv->byte_stream = devm_kzalloc(platform->dev, + SST_MAX_BIN_BYTES, GFP_KERNEL); + if (!drv->byte_stream) + return -ENOMEM; + + return ret; +} diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h index 14063ab8c7c5..8554889c0694 100644 --- a/sound/soc/intel/sst-atom-controls.h +++ b/sound/soc/intel/sst-atom-controls.h @@ -1,4 +1,6 @@ /* + * sst-atom-controls.h - Intel MID Platform driver header file + * * Copyright (C) 2013-14 Intel Corp * Author: Ramesh Babu <ramesh.babu.koul@intel.com> * Omair M Abdullah <omair.m.abdullah@intel.com> @@ -18,13 +20,293 @@ * */ -#ifndef __SST_CONTROLS_V2_H__ -#define __SST_CONTROLS_V2_H__ +#ifndef __SST_ATOM_CONTROLS_H__ +#define __SST_ATOM_CONTROLS_H__ enum { MERR_DPCM_AUDIO = 0, MERR_DPCM_COMPR, }; +/* define a bit for each mixer input */ +#define SST_MIX_IP(x) (x) + +#define SST_IP_CODEC0 SST_MIX_IP(2) +#define SST_IP_CODEC1 SST_MIX_IP(3) +#define SST_IP_LOOP0 SST_MIX_IP(4) +#define SST_IP_LOOP1 SST_MIX_IP(5) +#define SST_IP_LOOP2 SST_MIX_IP(6) +#define SST_IP_PROBE SST_MIX_IP(7) +#define SST_IP_VOIP SST_MIX_IP(12) +#define SST_IP_PCM0 SST_MIX_IP(13) +#define SST_IP_PCM1 SST_MIX_IP(14) +#define SST_IP_MEDIA0 SST_MIX_IP(17) +#define SST_IP_MEDIA1 SST_MIX_IP(18) +#define SST_IP_MEDIA2 SST_MIX_IP(19) +#define SST_IP_MEDIA3 SST_MIX_IP(20) + +#define SST_IP_LAST SST_IP_MEDIA3 + +#define SST_SWM_INPUT_COUNT (SST_IP_LAST + 1) +#define SST_CMD_SWM_MAX_INPUTS 6 + +#define SST_PATH_ID_SHIFT 8 +#define SST_DEFAULT_LOCATION_ID 0xFFFF +#define SST_DEFAULT_CELL_NBR 0xFF +#define SST_DEFAULT_MODULE_ID 0xFFFF + +/* + * Audio DSP Path Ids. Specified by the audio DSP FW + */ +enum sst_path_index { + SST_PATH_INDEX_CODEC_OUT0 = (0x02 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_OUT1 = (0x03 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_OUT = (0x04 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_OUT = (0x05 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_OUT = (0x06 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_OUT = (0x0C << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM0_OUT = (0x0D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_OUT = (0x0E << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM2_OUT = (0x0F << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_OUT = (0x12 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_OUT = (0x13 << SST_PATH_ID_SHIFT), + + + /* Start of input paths */ + SST_PATH_INDEX_CODEC_IN0 = (0x82 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_CODEC_IN1 = (0x83 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_SPROT_LOOP_IN = (0x84 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP1_IN = (0x85 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA_LOOP2_IN = (0x86 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_VOIP_IN = (0x8C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_PCM0_IN = (0x8D << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_PCM1_IN = (0x8E << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA0_IN = (0x8F << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA1_IN = (0x90 << SST_PATH_ID_SHIFT), + SST_PATH_INDEX_MEDIA2_IN = (0x91 << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_MEDIA3_IN = (0x9C << SST_PATH_ID_SHIFT), + + SST_PATH_INDEX_RESERVED = (0xFF << SST_PATH_ID_SHIFT), +}; + +/* + * path IDs + */ +enum sst_swm_inputs { + SST_SWM_IN_CODEC0 = (SST_PATH_INDEX_CODEC_IN0 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_CODEC1 = (SST_PATH_INDEX_CODEC_IN1 | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_VOIP = (SST_PATH_INDEX_VOIP_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM0 = (SST_PATH_INDEX_PCM0_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_PCM1 = (SST_PATH_INDEX_PCM1_IN | SST_DEFAULT_CELL_NBR), + SST_SWM_IN_MEDIA0 = (SST_PATH_INDEX_MEDIA0_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA1 = (SST_PATH_INDEX_MEDIA1_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA2 = (SST_PATH_INDEX_MEDIA2_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_MEDIA3 = (SST_PATH_INDEX_MEDIA3_IN | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_IN_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR) +}; + +/* + * path IDs + */ +enum sst_swm_outputs { + SST_SWM_OUT_CODEC0 = (SST_PATH_INDEX_CODEC_OUT0 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_CODEC1 = (SST_PATH_INDEX_CODEC_OUT1 | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_SPROT_LOOP = (SST_PATH_INDEX_SPROT_LOOP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP1 = (SST_PATH_INDEX_MEDIA_LOOP1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA_LOOP2 = (SST_PATH_INDEX_MEDIA_LOOP2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_VOIP = (SST_PATH_INDEX_VOIP_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM0 = (SST_PATH_INDEX_PCM0_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM1 = (SST_PATH_INDEX_PCM1_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_PCM2 = (SST_PATH_INDEX_PCM2_OUT | SST_DEFAULT_CELL_NBR), + SST_SWM_OUT_MEDIA0 = (SST_PATH_INDEX_MEDIA0_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_MEDIA1 = (SST_PATH_INDEX_MEDIA1_OUT | SST_DEFAULT_CELL_NBR), /* Part of Media Mixer */ + SST_SWM_OUT_END = (SST_PATH_INDEX_RESERVED | SST_DEFAULT_CELL_NBR), +}; + +enum sst_ipc_msg { + SST_IPC_IA_CMD = 1, + SST_IPC_IA_SET_PARAMS, + SST_IPC_IA_GET_PARAMS, +}; + +enum sst_cmd_type { + SST_CMD_BYTES_SET = 1, + SST_CMD_BYTES_GET = 2, +}; + +enum sst_task { + SST_TASK_SBA = 1, + SST_TASK_MMX, +}; + +enum sst_type { + SST_TYPE_CMD = 1, + SST_TYPE_PARAMS, +}; + +enum sst_flag { + SST_FLAG_BLOCKED = 1, + SST_FLAG_NONBLOCK, +}; + +/* + * Enumeration for indexing the gain cells in VB_SET_GAIN DSP command + */ +enum sst_gain_index { + /* GAIN IDs for SB task start here */ + SST_GAIN_INDEX_CODEC_OUT0, + SST_GAIN_INDEX_CODEC_OUT1, + SST_GAIN_INDEX_CODEC_IN0, + SST_GAIN_INDEX_CODEC_IN1, + + SST_GAIN_INDEX_SPROT_LOOP_OUT, + SST_GAIN_INDEX_MEDIA_LOOP1_OUT, + SST_GAIN_INDEX_MEDIA_LOOP2_OUT, + + SST_GAIN_INDEX_PCM0_IN_LEFT, + SST_GAIN_INDEX_PCM0_IN_RIGHT, + + SST_GAIN_INDEX_PCM1_OUT_LEFT, + SST_GAIN_INDEX_PCM1_OUT_RIGHT, + SST_GAIN_INDEX_PCM1_IN_LEFT, + SST_GAIN_INDEX_PCM1_IN_RIGHT, + SST_GAIN_INDEX_PCM2_OUT_LEFT, + + SST_GAIN_INDEX_PCM2_OUT_RIGHT, + SST_GAIN_INDEX_VOIP_OUT, + SST_GAIN_INDEX_VOIP_IN, + + /* Gain IDs for MMX task start here */ + SST_GAIN_INDEX_MEDIA0_IN_LEFT, + SST_GAIN_INDEX_MEDIA0_IN_RIGHT, + SST_GAIN_INDEX_MEDIA1_IN_LEFT, + SST_GAIN_INDEX_MEDIA1_IN_RIGHT, + + SST_GAIN_INDEX_MEDIA2_IN_LEFT, + SST_GAIN_INDEX_MEDIA2_IN_RIGHT, + + SST_GAIN_INDEX_GAIN_END +}; + +/* + * Audio DSP module IDs specified by FW spec + * TODO: Update with all modules + */ +enum sst_module_id { + SST_MODULE_ID_PCM = 0x0001, + SST_MODULE_ID_MP3 = 0x0002, + SST_MODULE_ID_MP24 = 0x0003, + SST_MODULE_ID_AAC = 0x0004, + SST_MODULE_ID_AACP = 0x0005, + SST_MODULE_ID_EAACP = 0x0006, + SST_MODULE_ID_WMA9 = 0x0007, + SST_MODULE_ID_WMA10 = 0x0008, + SST_MODULE_ID_WMA10P = 0x0009, + SST_MODULE_ID_RA = 0x000A, + SST_MODULE_ID_DDAC3 = 0x000B, + SST_MODULE_ID_TRUE_HD = 0x000C, + SST_MODULE_ID_HD_PLUS = 0x000D, + + SST_MODULE_ID_SRC = 0x0064, + SST_MODULE_ID_DOWNMIX = 0x0066, + SST_MODULE_ID_GAIN_CELL = 0x0067, + SST_MODULE_ID_SPROT = 0x006D, + SST_MODULE_ID_BASS_BOOST = 0x006E, + SST_MODULE_ID_STEREO_WDNG = 0x006F, + SST_MODULE_ID_AV_REMOVAL = 0x0070, + SST_MODULE_ID_MIC_EQ = 0x0071, + SST_MODULE_ID_SPL = 0x0072, + SST_MODULE_ID_ALGO_VTSV = 0x0073, + SST_MODULE_ID_NR = 0x0076, + SST_MODULE_ID_BWX = 0x0077, + SST_MODULE_ID_DRP = 0x0078, + SST_MODULE_ID_MDRP = 0x0079, + + SST_MODULE_ID_ANA = 0x007A, + SST_MODULE_ID_AEC = 0x007B, + SST_MODULE_ID_NR_SNS = 0x007C, + SST_MODULE_ID_SER = 0x007D, + SST_MODULE_ID_AGC = 0x007E, + + SST_MODULE_ID_CNI = 0x007F, + SST_MODULE_ID_CONTEXT_ALGO_AWARE = 0x0080, + SST_MODULE_ID_FIR_24 = 0x0081, + SST_MODULE_ID_IIR_24 = 0x0082, + + SST_MODULE_ID_ASRC = 0x0083, + SST_MODULE_ID_TONE_GEN = 0x0084, + SST_MODULE_ID_BMF = 0x0086, + SST_MODULE_ID_EDL = 0x0087, + SST_MODULE_ID_GLC = 0x0088, + + SST_MODULE_ID_FIR_16 = 0x0089, + SST_MODULE_ID_IIR_16 = 0x008A, + SST_MODULE_ID_DNR = 0x008B, + + SST_MODULE_ID_VIRTUALIZER = 0x008C, + SST_MODULE_ID_VISUALIZATION = 0x008D, + SST_MODULE_ID_LOUDNESS_OPTIMIZER = 0x008E, + SST_MODULE_ID_REVERBERATION = 0x008F, + + SST_MODULE_ID_CNI_TX = 0x0090, + SST_MODULE_ID_REF_LINE = 0x0091, + SST_MODULE_ID_VOLUME = 0x0092, + SST_MODULE_ID_FILT_DCR = 0x0094, + SST_MODULE_ID_SLV = 0x009A, + SST_MODULE_ID_NLF = 0x009B, + SST_MODULE_ID_TNR = 0x009C, + SST_MODULE_ID_WNR = 0x009D, + + SST_MODULE_ID_LOG = 0xFF00, + + SST_MODULE_ID_TASK = 0xFFFF, +}; + +enum sst_cmd { + SBA_IDLE = 14, + SBA_VB_SET_SPEECH_PATH = 26, + MMX_SET_GAIN = 33, + SBA_VB_SET_GAIN = 33, + FBA_VB_RX_CNI = 35, + MMX_SET_GAIN_TIMECONST = 36, + SBA_VB_SET_TIMECONST = 36, + SBA_VB_START = 85, + SBA_SET_SWM = 114, + SBA_SET_MDRP = 116, + SBA_HW_SET_SSP = 117, + SBA_SET_MEDIA_LOOP_MAP = 118, + SBA_SET_MEDIA_PATH = 119, + MMX_SET_MEDIA_PATH = 119, + SBA_VB_LPRO = 126, + SBA_VB_SET_FIR = 128, + SBA_VB_SET_IIR = 129, + SBA_SET_SSP_SLOT_MAP = 130, +}; + +enum sst_dsp_switch { + SST_SWITCH_OFF = 0, + SST_SWITCH_ON = 3, +}; + +enum sst_path_switch { + SST_PATH_OFF = 0, + SST_PATH_ON = 1, +}; + +enum sst_swm_state { + SST_SWM_OFF = 0, + SST_SWM_ON = 3, +}; #endif diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index 67673a2c0f41..b4ad98c43e5c 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -817,7 +817,7 @@ static struct sst_dsp_device byt_dev = { .ops = &sst_byt_ops, }; -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) +int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt = pdata->dsp; @@ -826,14 +826,6 @@ int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata) sst_byt_drop_all(byt); dev_dbg(byt->dev, "dsp in reset\n"); - return 0; -} -EXPORT_SYMBOL_GPL(sst_byt_dsp_suspend_noirq); - -int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata) -{ - struct sst_byt *byt = pdata->dsp; - dev_dbg(byt->dev, "free all blocks and unload fw\n"); sst_fw_unload(byt->fw); diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h index 06a4d202689b..8faff6dcf25d 100644 --- a/sound/soc/intel/sst-baytrail-ipc.h +++ b/sound/soc/intel/sst-baytrail-ipc.h @@ -66,7 +66,6 @@ int sst_byt_get_dsp_position(struct sst_byt *byt, int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata); struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt); -int sst_byt_dsp_suspend_noirq(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_suspend_late(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_boot(struct device *dev, struct sst_pdata *pdata); int sst_byt_dsp_wait_for_ready(struct device *dev, struct sst_pdata *pdata); diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 599401c0c655..eab1c7d85187 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -59,6 +59,9 @@ struct sst_byt_priv_data { /* DAI data */ struct sst_byt_pcm_data pcm[BYT_PCM_COUNT]; + + /* flag indicating is stream context restore needed after suspend */ + bool restore_stream; }; /* this may get called several times by oss emulation */ @@ -184,7 +187,10 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_start(byt, pcm_data->stream, 0); break; case SNDRV_PCM_TRIGGER_RESUME: - schedule_work(&pcm_data->work); + if (pdata->restore_stream == true) + schedule_work(&pcm_data->work); + else + sst_byt_stream_resume(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: sst_byt_stream_resume(byt, pcm_data->stream); @@ -193,6 +199,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) sst_byt_stream_stop(byt, pcm_data->stream); break; case SNDRV_PCM_TRIGGER_SUSPEND: + pdata->restore_stream = false; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; @@ -404,26 +411,10 @@ static const struct snd_soc_component_driver byt_dai_component = { }; #ifdef CONFIG_PM -static int sst_byt_pcm_dev_suspend_noirq(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - int ret; - - dev_dbg(dev, "suspending noirq\n"); - - /* at this point all streams will be stopped and context saved */ - ret = sst_byt_dsp_suspend_noirq(dev, sst_pdata); - if (ret < 0) { - dev_err(dev, "failed to suspend %d\n", ret); - return ret; - } - - return ret; -} - static int sst_byt_pcm_dev_suspend_late(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + struct sst_byt_priv_data *priv_data = dev_get_drvdata(dev); int ret; dev_dbg(dev, "suspending late\n"); @@ -434,34 +425,30 @@ static int sst_byt_pcm_dev_suspend_late(struct device *dev) return ret; } + priv_data->restore_stream = true; + return ret; } static int sst_byt_pcm_dev_resume_early(struct device *dev) { struct sst_pdata *sst_pdata = dev_get_platdata(dev); + int ret; dev_dbg(dev, "resume early\n"); /* load fw and boot DSP */ - return sst_byt_dsp_boot(dev, sst_pdata); -} - -static int sst_byt_pcm_dev_resume(struct device *dev) -{ - struct sst_pdata *sst_pdata = dev_get_platdata(dev); - - dev_dbg(dev, "resume\n"); + ret = sst_byt_dsp_boot(dev, sst_pdata); + if (ret) + return ret; /* wait for FW to finish booting */ return sst_byt_dsp_wait_for_ready(dev, sst_pdata); } static const struct dev_pm_ops sst_byt_pm_ops = { - .suspend_noirq = sst_byt_pcm_dev_suspend_noirq, .suspend_late = sst_byt_pcm_dev_suspend_late, .resume_early = sst_byt_pcm_dev_resume_early, - .resume = sst_byt_pcm_dev_resume, }; #define SST_BYT_PM_OPS (&sst_byt_pm_ops) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 61bf6da4bb02..33fc5c3abf55 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -138,11 +138,10 @@ static inline unsigned int hsw_ipc_to_mixer(u32 value) static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - struct hsw_priv_data *pdata = - snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -176,11 +175,10 @@ static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - struct hsw_priv_data *pdata = - snd_soc_platform_get_drvdata(platform); struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -208,8 +206,8 @@ static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, static int hsw_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; u32 volume; @@ -233,8 +231,8 @@ static int hsw_volume_put(struct snd_kcontrol *kcontrol, static int hsw_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_platform *platform = snd_soc_kcontrol_platform(kcontrol); - struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct hsw_priv_data *pdata = snd_soc_component_get_drvdata(cmpnt); struct sst_hsw *hsw = pdata->hsw; unsigned int volume = 0; @@ -778,20 +776,11 @@ static const struct snd_soc_dapm_route graph[] = { static int hsw_pcm_probe(struct snd_soc_platform *platform) { + struct hsw_priv_data *priv_data = snd_soc_platform_get_drvdata(platform); struct sst_pdata *pdata = dev_get_platdata(platform->dev); - struct hsw_priv_data *priv_data; - struct device *dma_dev; + struct device *dma_dev = pdata->dma_dev; int i, ret = 0; - if (!pdata) - return -ENODEV; - - dma_dev = pdata->dma_dev; - - priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); - priv_data->hsw = pdata->dsp; - snd_soc_platform_set_drvdata(platform, priv_data); - /* allocate DSP buffer page tables */ for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { @@ -848,27 +837,38 @@ static struct snd_soc_platform_driver hsw_soc_platform = { .ops = &hsw_pcm_ops, .pcm_new = hsw_pcm_new, .pcm_free = hsw_pcm_free, - .controls = hsw_volume_controls, - .num_controls = ARRAY_SIZE(hsw_volume_controls), - .dapm_widgets = widgets, - .num_dapm_widgets = ARRAY_SIZE(widgets), - .dapm_routes = graph, - .num_dapm_routes = ARRAY_SIZE(graph), }; static const struct snd_soc_component_driver hsw_dai_component = { - .name = "haswell-dai", + .name = "haswell-dai", + .controls = hsw_volume_controls, + .num_controls = ARRAY_SIZE(hsw_volume_controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = graph, + .num_dapm_routes = ARRAY_SIZE(graph), }; static int hsw_pcm_dev_probe(struct platform_device *pdev) { struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + struct hsw_priv_data *priv_data; int ret; + if (!sst_pdata) + return -EINVAL; + + priv_data = devm_kzalloc(&pdev->dev, sizeof(*priv_data), GFP_KERNEL); + if (!priv_data) + return -ENOMEM; + ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata); if (ret < 0) return -ENODEV; + priv_data->hsw = sst_pdata->dsp; + platform_set_drvdata(pdev, priv_data); + ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform); if (ret < 0) goto err_plat; diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 29c059ca19e8..59467775c9b8 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -86,7 +86,7 @@ static int sst_platform_compr_free(struct snd_compr_stream *cstream) /*need to check*/ str_id = stream->id; if (str_id) - ret_val = stream->compr_ops->close(str_id); + ret_val = stream->compr_ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); pr_debug("%s: %d\n", __func__, ret_val); @@ -158,7 +158,7 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, cb.drain_cb_param = cstream; cb.drain_notify = sst_drain_notify; - retval = stream->compr_ops->open(&str_params, &cb); + retval = stream->compr_ops->open(sst->dev, &str_params, &cb); if (retval < 0) { pr_err("stream allocation failed %d\n", retval); return retval; @@ -170,10 +170,30 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, static int sst_platform_compr_trigger(struct snd_compr_stream *cstream, int cmd) { - struct sst_runtime_stream *stream = - cstream->runtime->private_data; - - return stream->compr_ops->control(cmd, stream->id); + struct sst_runtime_stream *stream = cstream->runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream->compr_ops->stream_start) + return stream->compr_ops->stream_start(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_STOP: + if (stream->compr_ops->stream_drop) + return stream->compr_ops->stream_drop(sst->dev, stream->id); + case SND_COMPR_TRIGGER_DRAIN: + if (stream->compr_ops->stream_drain) + return stream->compr_ops->stream_drain(sst->dev, stream->id); + case SND_COMPR_TRIGGER_PARTIAL_DRAIN: + if (stream->compr_ops->stream_partial_drain) + return stream->compr_ops->stream_partial_drain(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (stream->compr_ops->stream_pause) + return stream->compr_ops->stream_pause(sst->dev, stream->id); + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (stream->compr_ops->stream_pause_release) + return stream->compr_ops->stream_pause_release(sst->dev, stream->id); + default: + return -EINVAL; + } } static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, @@ -182,7 +202,7 @@ static int sst_platform_compr_pointer(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->tstamp(stream->id, tstamp); + stream->compr_ops->tstamp(sst->dev, stream->id, tstamp); tstamp->byte_offset = tstamp->copied_total % (u32)cstream->runtime->buffer_size; pr_debug("calc bytes offset/copied bytes as %d\n", tstamp->byte_offset); @@ -195,7 +215,7 @@ static int sst_platform_compr_ack(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream; stream = cstream->runtime->private_data; - stream->compr_ops->ack(stream->id, (unsigned long)bytes); + stream->compr_ops->ack(sst->dev, stream->id, (unsigned long)bytes); stream->bytes_written += bytes; return 0; @@ -225,7 +245,7 @@ static int sst_platform_compr_set_metadata(struct snd_compr_stream *cstream, struct sst_runtime_stream *stream = cstream->runtime->private_data; - return stream->compr_ops->set_metadata(stream->id, metadata); + return stream->compr_ops->set_metadata(sst->dev, stream->id, metadata); } struct snd_compr_ops sst_platform_compr_ops = { diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 706212a6a68c..8e1e9bc27642 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -277,7 +277,7 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, stream->stream_info.str_id = str_params.stream_id; - ret_val = stream->ops->open(&str_params); + ret_val = stream->ops->open(sst->dev, &str_params); if (ret_val <= 0) return ret_val; @@ -314,14 +314,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; - ret_val = stream->ops->device_control( - SST_SND_STREAM_INIT, &stream->stream_info); + ret_val = stream->ops->stream_init(sst->dev, &stream->stream_info); if (ret_val) pr_err("control_set ret error %d\n", ret_val); return ret_val; } -/* end -- helper functions */ static int sst_media_open(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -373,7 +371,7 @@ static void sst_media_close(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) - ret_val = stream->ops->close(str_id); + ret_val = stream->ops->close(sst->dev, str_id); module_put(sst->dev->driver->owner); kfree(stream); } @@ -403,8 +401,7 @@ static int sst_media_prepare(struct snd_pcm_substream *substream, stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { - ret_val = stream->ops->device_control( - SST_SND_DROP, &str_id); + ret_val = stream->ops->stream_drop(sst->dev, str_id); return ret_val; } @@ -461,7 +458,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, { int ret_val = 0, str_id; struct sst_runtime_stream *stream; - int str_cmd, status; + int status; pr_debug("sst_platform_pcm_trigger called\n"); stream = substream->runtime->private_data; @@ -469,29 +466,29 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: pr_debug("sst: Trigger Start\n"); - str_cmd = SST_SND_START; status = SST_PLATFORM_RUNNING; stream->stream_info.arg = substream; + ret_val = stream->ops->stream_start(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); - str_cmd = SST_SND_DROP; status = SST_PLATFORM_DROPPED; + ret_val = stream->ops->stream_drop(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: pr_debug("sst: in pause\n"); - str_cmd = SST_SND_PAUSE; status = SST_PLATFORM_PAUSED; + ret_val = stream->ops->stream_pause(sst->dev, str_id); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: pr_debug("sst: in pause release\n"); - str_cmd = SST_SND_RESUME; status = SST_PLATFORM_RUNNING; + ret_val = stream->ops->stream_pause_release(sst->dev, str_id); break; default: return -EINVAL; } - ret_val = stream->ops->device_control(str_cmd, &str_id); + if (!ret_val) sst_set_stream_status(stream, status); @@ -511,8 +508,7 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->device_control( - SST_SND_BUFFER_POINTER, str_info); + ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); if (ret_val) { pr_err("sst: error code = %d\n", ret_val); return ret_val; @@ -554,7 +550,13 @@ static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) return retval; } -static struct snd_soc_platform_driver sst_soc_platform_drv = { +static int sst_soc_probe(struct snd_soc_platform *platform) +{ + return sst_dsp_init_v2_dpcm(platform); +} + +static struct snd_soc_platform_driver sst_soc_platform_drv = { + .probe = sst_soc_probe, .ops = &sst_platform_ops, .compr_ops = &sst_platform_compr_ops, .pcm_new = sst_pcm_new, diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c6a42c08e24..0c5b943daff3 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -54,20 +54,6 @@ enum sst_drv_status { SST_PLATFORM_DROPPED, }; -enum sst_controls { - SST_SND_ALLOC = 0x00, - SST_SND_PAUSE = 0x01, - SST_SND_RESUME = 0x02, - SST_SND_DROP = 0x03, - SST_SND_FREE = 0x04, - SST_SND_BUFFER_POINTER = 0x05, - SST_SND_STREAM_INIT = 0x06, - SST_SND_START = 0x07, - SST_SET_BYTE_STREAM = 0x100A, - SST_GET_BYTE_STREAM = 0x100B, - SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, -}; - enum sst_stream_ops { STREAM_OPS_PLAYBACK = 0, STREAM_OPS_CAPTURE, @@ -113,24 +99,36 @@ struct sst_compress_cb { struct compress_sst_ops { const char *name; - int (*open) (struct snd_sst_params *str_params, - struct sst_compress_cb *cb); - int (*control) (unsigned int cmd, unsigned int str_id); - int (*tstamp) (unsigned int str_id, struct snd_compr_tstamp *tstamp); - int (*ack) (unsigned int str_id, unsigned long bytes); - int (*close) (unsigned int str_id); - int (*get_caps) (struct snd_compr_caps *caps); - int (*get_codec_caps) (struct snd_compr_codec_caps *codec); - int (*set_metadata) (unsigned int str_id, + int (*open)(struct device *dev, + struct snd_sst_params *str_params, struct sst_compress_cb *cb); + int (*stream_start)(struct device *dev, unsigned int str_id); + int (*stream_drop)(struct device *dev, unsigned int str_id); + int (*stream_drain)(struct device *dev, unsigned int str_id); + int (*stream_partial_drain)(struct device *dev, unsigned int str_id); + int (*stream_pause)(struct device *dev, unsigned int str_id); + int (*stream_pause_release)(struct device *dev, unsigned int str_id); + + int (*tstamp)(struct device *dev, unsigned int str_id, + struct snd_compr_tstamp *tstamp); + int (*ack)(struct device *dev, unsigned int str_id, + unsigned long bytes); + int (*close)(struct device *dev, unsigned int str_id); + int (*get_caps)(struct snd_compr_caps *caps); + int (*get_codec_caps)(struct snd_compr_codec_caps *codec); + int (*set_metadata)(struct device *dev, unsigned int str_id, struct snd_compr_metadata *mdata); - }; struct sst_ops { - int (*open) (struct snd_sst_params *str_param); - int (*device_control) (int cmd, void *arg); - int (*set_generic_params)(enum sst_controls cmd, void *arg); - int (*close) (unsigned int str_id); + int (*open) (struct device *dev, struct snd_sst_params *str_param); + int (*stream_init) (struct device *dev, struct pcm_stream_info *str_info); + int (*stream_start) (struct device *dev, int str_id); + int (*stream_drop) (struct device *dev, int str_id); + int (*stream_pause) (struct device *dev, int str_id); + int (*stream_pause_release) (struct device *dev, int str_id); + int (*stream_read_tstamp) (struct device *dev, struct pcm_stream_info *str_info); + int (*send_byte_stream)(struct device *dev, struct snd_sst_bytes_v2 *bytes); + int (*close) (struct device *dev, unsigned int str_id); }; struct sst_runtime_stream { @@ -152,6 +150,8 @@ struct sst_device { }; struct sst_data; + +int sst_dsp_init_v2_dpcm(struct snd_soc_platform *platform); void sst_set_stream_status(struct sst_runtime_stream *stream, int state); int sst_fill_stream_params(void *substream, const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress); @@ -166,6 +166,7 @@ struct sst_algo_int_control_v2 { struct sst_data { struct platform_device *pdev; struct sst_platform_data *pdata; + char *byte_stream; struct mutex lock; }; int sst_register_dsp(struct sst_device *sst); diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index f8a6adc2d81c..4336d1831485 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -260,7 +260,7 @@ static struct snd_soc_dai_link omap_twl4030_dai_links[] = { .stream_name = "TWL4030 Voice", .cpu_dai_name = "omap-mcbsp.3", .codec_dai_name = "twl4030-voice", - .platform_name = "omap-mcbsp.2", + .platform_name = "omap-mcbsp.3", .codec_name = "twl4030-codec", .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM, diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 943922c79f78..b10ae8074461 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -168,7 +168,7 @@ static int rx51_spk_event(struct snd_soc_dapm_widget *w, static int rx51_hp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - struct snd_soc_codec *codec = w->dapm->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); if (SND_SOC_DAPM_EVENT_ON(event)) tpa6130a2_stereo_enable(codec, 1); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 0109f6c2334e..a8e097433074 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -765,9 +765,7 @@ static int pxa_ssp_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) +#define PXA_SSP_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops pxa_ssp_dai_ops = { .startup = pxa_ssp_startup, diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 8d8e4b59049f..fb9e05c9f471 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -165,13 +165,14 @@ static int rockchip_i2s_set_fmt(struct snd_soc_dai *cpu_dai, struct rk_i2s_dev *i2s = to_info(cpu_dai); unsigned int mask = 0, val = 0; - mask = I2S_CKR_MSS_SLAVE; + mask = I2S_CKR_MSS_MASK; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = I2S_CKR_MSS_SLAVE; + /* Set source clock in Master mode */ + val = I2S_CKR_MSS_MASTER; break; case SND_SOC_DAIFMT_CBM_CFM: - val = I2S_CKR_MSS_MASTER; + val = I2S_CKR_MSS_SLAVE; break; default: return -EINVAL; @@ -361,6 +362,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) case I2S_XFER: case I2S_CLR: case I2S_RXDR: + case I2S_FIFOLR: + case I2S_INTSR: return true; default: return false; @@ -370,8 +373,8 @@ static bool rockchip_i2s_rd_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: case I2S_INTSR: + case I2S_CLR: return true; default: return false; @@ -381,8 +384,6 @@ static bool rockchip_i2s_volatile_reg(struct device *dev, unsigned int reg) static bool rockchip_i2s_precious_reg(struct device *dev, unsigned int reg) { switch (reg) { - case I2S_FIFOLR: - return true; default: return false; } diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 03eec22f0f46..9d513473b300 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -462,7 +462,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, if (dir == SND_SOC_CLOCK_IN) rfs = 0; - if ((rfs && other->rfs && (other->rfs != rfs)) || + if ((rfs && other && other->rfs && (other->rfs != rfs)) || (any_active(i2s) && (((dir == SND_SOC_CLOCK_IN) && !(mod & MOD_CDCLKCON)) || @@ -762,7 +762,8 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, } else { u32 mod = readl(i2s->addr + I2SMOD); i2s->cdclk_out = !(mod & MOD_CDCLKCON); - other->cdclk_out = i2s->cdclk_out; + if (other) + other->cdclk_out = i2s->cdclk_out; } /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index 9902efcb8ea1..a05482651aae 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -228,10 +228,12 @@ static struct snd_soc_dai_link speyside_dai[] = { }, }; -static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) +static int speyside_wm9081_init(struct snd_soc_component *component) { + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, + return snd_soc_codec_set_sysclk(codec, WM9081_SYSCLK_MCLK, 0, MCLK_AUDIO_RATE, 0); } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 3fdf3be7b99a..f95e7ab135e8 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -247,7 +247,7 @@ rsnd_gen2_dma_addr(struct rsnd_priv *priv, }; /* it shouldn't happen */ - if (use_dvc & !use_src) + if (use_dvc && !use_src) dev_err(dev, "DVC is selected without SRC\n"); /* use SSIU or SSI ? */ diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 27c06acce205..cecfab3cc948 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -101,10 +101,12 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto fe_err; + else if (ret == 0) dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - } /* calculate valid and active FE <-> BE dpcms */ dpcm_process_paths(fe, stream, &list, 1); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dde4b82ad41d..57de6a7d7ffa 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -270,79 +270,54 @@ static const struct file_operations codec_reg_fops = { .llseek = default_llseek, }; -static struct dentry *soc_debugfs_create_dir(struct dentry *parent, - const char *fmt, ...) +static void soc_init_component_debugfs(struct snd_soc_component *component) { - struct dentry *de; - va_list ap; - char *s; + if (component->debugfs_prefix) { + char *name; - va_start(ap, fmt); - s = kvasprintf(GFP_KERNEL, fmt, ap); - va_end(ap); + name = kasprintf(GFP_KERNEL, "%s:%s", + component->debugfs_prefix, component->name); + if (name) { + component->debugfs_root = debugfs_create_dir(name, + component->card->debugfs_card_root); + kfree(name); + } + } else { + component->debugfs_root = debugfs_create_dir(component->name, + component->card->debugfs_card_root); + } - if (!s) - return NULL; + if (!component->debugfs_root) { + dev_warn(component->dev, + "ASoC: Failed to create component debugfs directory\n"); + return; + } - de = debugfs_create_dir(s, parent); - kfree(s); + snd_soc_dapm_debugfs_init(snd_soc_component_get_dapm(component), + component->debugfs_root); - return de; + if (component->init_debugfs) + component->init_debugfs(component); } -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +static void soc_cleanup_component_debugfs(struct snd_soc_component *component) { - struct dentry *debugfs_card_root = codec->component.card->debugfs_card_root; + debugfs_remove_recursive(component->debugfs_root); +} - codec->debugfs_codec_root = soc_debugfs_create_dir(debugfs_card_root, - "codec:%s", - codec->component.name); - if (!codec->debugfs_codec_root) { - dev_warn(codec->dev, - "ASoC: Failed to create codec debugfs directory\n"); - return; - } +static void soc_init_codec_debugfs(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); - debugfs_create_bool("cache_sync", 0444, codec->debugfs_codec_root, + debugfs_create_bool("cache_sync", 0444, codec->component.debugfs_root, &codec->cache_sync); - debugfs_create_bool("cache_only", 0444, codec->debugfs_codec_root, - &codec->cache_only); codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - codec->debugfs_codec_root, + codec->component.debugfs_root, codec, &codec_reg_fops); if (!codec->debugfs_reg) dev_warn(codec->dev, "ASoC: Failed to create codec register debugfs file\n"); - - snd_soc_dapm_debugfs_init(&codec->dapm, codec->debugfs_codec_root); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_codec_root); -} - -static void soc_init_platform_debugfs(struct snd_soc_platform *platform) -{ - struct dentry *debugfs_card_root = platform->component.card->debugfs_card_root; - - platform->debugfs_platform_root = soc_debugfs_create_dir(debugfs_card_root, - "platform:%s", - platform->component.name); - if (!platform->debugfs_platform_root) { - dev_warn(platform->dev, - "ASoC: Failed to create platform debugfs directory\n"); - return; - } - - snd_soc_dapm_debugfs_init(&platform->component.dapm, - platform->debugfs_platform_root); -} - -static void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) -{ - debugfs_remove_recursive(platform->debugfs_platform_root); } static ssize_t codec_list_read_file(struct file *file, char __user *user_buf, @@ -474,19 +449,15 @@ static void soc_cleanup_card_debugfs(struct snd_soc_card *card) #else -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} - -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ -} +#define soc_init_codec_debugfs NULL -static inline void soc_init_platform_debugfs(struct snd_soc_platform *platform) +static inline void soc_init_component_debugfs( + struct snd_soc_component *component) { } -static inline void soc_cleanup_platform_debugfs(struct snd_soc_platform *platform) +static inline void soc_cleanup_component_debugfs( + struct snd_soc_component *component) { } @@ -579,10 +550,8 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_codec *codec; int i, j; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* Due to the resume being scheduled into a workqueue we could @@ -835,10 +804,8 @@ int snd_soc_resume(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); int i, ac97_control = 0; - /* If the initialization of this soc device failed, there is no codec - * associated with it. Just bail out in this case. - */ - if (list_empty(&card->codec_dev_list)) + /* If the card is not initialized yet there is nothing to do */ + if (!card->instantiated) return 0; /* activate pins from sleep state */ @@ -887,35 +854,40 @@ EXPORT_SYMBOL_GPL(snd_soc_resume); static const struct snd_soc_dai_ops null_dai_ops = { }; -static struct snd_soc_codec *soc_find_codec( - const struct device_node *codec_of_node, - const char *codec_name) +static struct snd_soc_component *soc_find_component( + const struct device_node *of_node, const char *name) { - struct snd_soc_codec *codec; + struct snd_soc_component *component; - list_for_each_entry(codec, &codec_list, list) { - if (codec_of_node) { - if (codec->dev->of_node != codec_of_node) - continue; - } else { - if (strcmp(codec->component.name, codec_name)) - continue; + list_for_each_entry(component, &component_list, list) { + if (of_node) { + if (component->dev->of_node == of_node) + return component; + } else if (strcmp(component->name, name) == 0) { + return component; } - - return codec; } return NULL; } -static struct snd_soc_dai *soc_find_codec_dai(struct snd_soc_codec *codec, - const char *codec_dai_name) +static struct snd_soc_dai *snd_soc_find_dai( + const struct snd_soc_dai_link_component *dlc) { - struct snd_soc_dai *codec_dai; + struct snd_soc_component *component; + struct snd_soc_dai *dai; + + /* Find CPU DAI from registered DAIs*/ + list_for_each_entry(component, &component_list, list) { + if (dlc->of_node && component->dev->of_node != dlc->of_node) + continue; + if (dlc->name && strcmp(dev_name(component->dev), dlc->name)) + continue; + list_for_each_entry(dai, &component->dai_list, list) { + if (dlc->dai_name && strcmp(dai->name, dlc->dai_name)) + continue; - list_for_each_entry(codec_dai, &codec->component.dai_list, list) { - if (!strcmp(codec_dai->name, codec_dai_name)) { - return codec_dai; + return dai; } } @@ -926,33 +898,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_component *component; struct snd_soc_dai_link_component *codecs = dai_link->codecs; + struct snd_soc_dai_link_component cpu_dai_component; struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_platform *platform; - struct snd_soc_dai *cpu_dai; const char *platform_name; int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); - /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(component, &component_list, list) { - if (dai_link->cpu_of_node && - component->dev->of_node != dai_link->cpu_of_node) - continue; - if (dai_link->cpu_name && - strcmp(dev_name(component->dev), dai_link->cpu_name)) - continue; - list_for_each_entry(cpu_dai, &component->dai_list, list) { - if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - - rtd->cpu_dai = cpu_dai; - } - } - + cpu_dai_component.name = dai_link->cpu_name; + cpu_dai_component.of_node = dai_link->cpu_of_node; + cpu_dai_component.dai_name = dai_link->cpu_dai_name; + rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); @@ -963,15 +921,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Find CODEC from registered CODECs */ for (i = 0; i < rtd->num_codecs; i++) { - struct snd_soc_codec *codec; - codec = soc_find_codec(codecs[i].of_node, codecs[i].name); - if (!codec) { - dev_err(card->dev, "ASoC: CODEC %s not registered\n", - codecs[i].name); - return -EPROBE_DEFER; - } - - codec_dais[i] = soc_find_codec_dai(codec, codecs[i].dai_name); + codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); @@ -1012,68 +962,46 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return 0; } -static int soc_remove_platform(struct snd_soc_platform *platform) +static void soc_remove_component(struct snd_soc_component *component) { - int ret; - - if (platform->driver->remove) { - ret = platform->driver->remove(platform); - if (ret < 0) - dev_err(platform->dev, "ASoC: failed to remove %d\n", - ret); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->component.dapm); - - soc_cleanup_platform_debugfs(platform); - platform->probed = 0; - module_put(platform->dev->driver->owner); - - return 0; -} + if (!component->probed) + return; -static void soc_remove_codec(struct snd_soc_codec *codec) -{ - int err; + /* This is a HACK and will be removed soon */ + if (component->codec) + list_del(&component->codec->card_list); - if (codec->driver->remove) { - err = codec->driver->remove(codec); - if (err < 0) - dev_err(codec->dev, "ASoC: failed to remove %d\n", err); - } + if (component->remove) + component->remove(component); - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&codec->dapm); + snd_soc_dapm_free(snd_soc_component_get_dapm(component)); - soc_cleanup_codec_debugfs(codec); - codec->probed = 0; - list_del(&codec->card_list); - module_put(codec->dev->driver->owner); + soc_cleanup_component_debugfs(component); + component->probed = 0; + module_put(component->dev->driver->owner); } -static void soc_remove_codec_dai(struct snd_soc_dai *codec_dai, int order) +static void soc_remove_dai(struct snd_soc_dai *dai, int order) { int err; - if (codec_dai && codec_dai->probed && - codec_dai->driver->remove_order == order) { - if (codec_dai->driver->remove) { - err = codec_dai->driver->remove(codec_dai); + if (dai && dai->probed && + dai->driver->remove_order == order) { + if (dai->driver->remove) { + err = dai->driver->remove(dai); if (err < 0) - dev_err(codec_dai->dev, + dev_err(dai->dev, "ASoC: failed to remove %s: %d\n", - codec_dai->name, err); + dai->name, err); } - codec_dai->probed = 0; + dai->probed = 0; } } static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, err; + int i; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1085,22 +1013,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* remove the CODEC DAI */ for (i = 0; i < rtd->num_codecs; i++) - soc_remove_codec_dai(rtd->codec_dais[i], order); + soc_remove_dai(rtd->codec_dais[i], order); - /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed && - cpu_dai->driver->remove_order == order) { - if (cpu_dai->driver->remove) { - err = cpu_dai->driver->remove(cpu_dai); - if (err < 0) - dev_err(cpu_dai->dev, - "ASoC: failed to remove %s: %d\n", - cpu_dai->name, err); - } - cpu_dai->probed = 0; - if (!cpu_dai->codec) - module_put(cpu_dai->dev->driver->owner); - } + soc_remove_dai(rtd->cpu_dai, order); } static void soc_remove_link_components(struct snd_soc_card *card, int num, @@ -1109,29 +1024,24 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_codec *codec; + struct snd_soc_component *component; int i; /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) { - soc_remove_platform(platform); - } + if (platform && platform->component.driver->remove_order == order) + soc_remove_component(&platform->component); /* remove the CODEC-side CODEC */ for (i = 0; i < rtd->num_codecs; i++) { - codec = rtd->codec_dais[i]->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + component = rtd->codec_dais[i]->component; + if (component->driver->remove_order == order) + soc_remove_component(component); } /* remove any CPU-side CODEC */ if (cpu_dai) { - codec = cpu_dai->codec; - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); + if (cpu_dai->component->driver->remove_order == order) + soc_remove_component(cpu_dai->component); } } @@ -1173,137 +1083,78 @@ static void soc_set_name_prefix(struct snd_soc_card *card, } } -static int soc_probe_codec(struct snd_soc_card *card, - struct snd_soc_codec *codec) +static int soc_probe_component(struct snd_soc_card *card, + struct snd_soc_component *component) { - int ret = 0; - const struct snd_soc_codec_driver *driver = codec->driver; + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct snd_soc_dai *dai; + int ret; - codec->component.card = card; - codec->dapm.card = card; - soc_set_name_prefix(card, &codec->component); + if (component->probed) + return 0; + + component->card = card; + dapm->card = card; + soc_set_name_prefix(card, component); - if (!try_module_get(codec->dev->driver->owner)) + if (!try_module_get(component->dev->driver->owner)) return -ENODEV; - soc_init_codec_debugfs(codec); + soc_init_component_debugfs(component); - if (driver->dapm_widgets) { - ret = snd_soc_dapm_new_controls(&codec->dapm, - driver->dapm_widgets, - driver->num_dapm_widgets); + if (component->dapm_widgets) { + ret = snd_soc_dapm_new_controls(dapm, component->dapm_widgets, + component->num_dapm_widgets); if (ret != 0) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to create new controls %d\n", ret); goto err_probe; } } - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) { - ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); - + list_for_each_entry(dai, &component->dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); if (ret != 0) { - dev_err(codec->dev, + dev_err(component->dev, "Failed to create DAI widgets %d\n", ret); goto err_probe; } } - codec->dapm.idle_bias_off = driver->idle_bias_off; - - if (driver->probe) { - ret = driver->probe(codec); + if (component->probe) { + ret = component->probe(component); if (ret < 0) { - dev_err(codec->dev, - "ASoC: failed to probe CODEC %d\n", ret); + dev_err(component->dev, + "ASoC: failed to probe component %d\n", ret); goto err_probe; } - WARN(codec->dapm.idle_bias_off && - codec->dapm.bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias with idle_bias_off==1\n", - codec->component.name); - } - - if (driver->controls) - snd_soc_add_codec_controls(codec, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes, - driver->num_dapm_routes); - - /* mark codec as probed and add to card codec list */ - codec->probed = 1; - list_add(&codec->card_list, &card->codec_dev_list); - list_add(&codec->dapm.list, &card->dapm_list); - - return 0; - -err_probe: - soc_cleanup_codec_debugfs(codec); - module_put(codec->dev->driver->owner); - - return ret; -} - -static int soc_probe_platform(struct snd_soc_card *card, - struct snd_soc_platform *platform) -{ - int ret = 0; - const struct snd_soc_platform_driver *driver = platform->driver; - struct snd_soc_component *component; - struct snd_soc_dai *dai; - platform->component.card = card; - platform->component.dapm.card = card; - - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - soc_init_platform_debugfs(platform); - - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&platform->component.dapm, - driver->dapm_widgets, driver->num_dapm_widgets); - - /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(component, &component_list, list) { - if (component->dev != platform->dev) - continue; - list_for_each_entry(dai, &component->dai_list, list) - snd_soc_dapm_new_dai_widgets(&platform->component.dapm, - dai); + WARN(dapm->idle_bias_off && + dapm->bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias with idle_bias_off==1\n", + component->name); } - platform->component.dapm.idle_bias_off = 1; + if (component->controls) + snd_soc_add_component_controls(component, component->controls, + component->num_controls); + if (component->dapm_routes) + snd_soc_dapm_add_routes(dapm, component->dapm_routes, + component->num_dapm_routes); - if (driver->probe) { - ret = driver->probe(platform); - if (ret < 0) { - dev_err(platform->dev, - "ASoC: failed to probe platform %d\n", ret); - goto err_probe; - } - } + component->probed = 1; + list_add(&dapm->list, &card->dapm_list); - if (driver->controls) - snd_soc_add_platform_controls(platform, driver->controls, - driver->num_controls); - if (driver->dapm_routes) - snd_soc_dapm_add_routes(&platform->component.dapm, - driver->dapm_routes, driver->num_dapm_routes); - - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->component.dapm.list, &card->dapm_list); + /* This is a HACK and will be removed soon */ + if (component->codec) + list_add(&component->codec->card_list, &card->codec_dev_list); return 0; err_probe: - soc_cleanup_platform_debugfs(platform); - module_put(platform->dev->driver->owner); + soc_cleanup_component_debugfs(component); + module_put(component->dev->driver->owner); return ret; } @@ -1325,7 +1176,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; - rtd->dev->init_name = name; + dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); @@ -1342,17 +1193,21 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, } rtd->dev_registered = 1; - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", ret); + if (rtd->codec) { + /* add DAPM sysfs entries for this codec */ + ret = snd_soc_dapm_sys_add(rtd->dev); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec dapm sysfs entries: %d\n", + ret); - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", ret); + /* add codec sysfs entries */ + ret = device_create_file(rtd->dev, &dev_attr_codec_reg); + if (ret < 0) + dev_err(rtd->dev, + "ASoC: failed to add codec sysfs files: %d\n", + ret); + } return 0; } @@ -1361,33 +1216,31 @@ static int soc_probe_link_components(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_component *component; int i, ret; /* probe the CPU-side component, if it is a CODEC */ - if (cpu_dai->codec && - !cpu_dai->codec->probed && - cpu_dai->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, cpu_dai->codec); + component = rtd->cpu_dai->component; + if (component->driver->probe_order == order) { + ret = soc_probe_component(card, component); if (ret < 0) return ret; } /* probe the CODEC-side components */ for (i = 0; i < rtd->num_codecs; i++) { - if (!rtd->codec_dais[i]->codec->probed && - rtd->codec_dais[i]->codec->driver->probe_order == order) { - ret = soc_probe_codec(card, rtd->codec_dais[i]->codec); + component = rtd->codec_dais[i]->component; + if (component->driver->probe_order == order) { + ret = soc_probe_component(card, component); if (ret < 0) return ret; } } /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); + if (platform->component.driver->probe_order == order) { + ret = soc_probe_component(card, &platform->component); if (ret < 0) return ret; } @@ -1482,18 +1335,12 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - if (!cpu_dai->codec) { - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; - } - if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: failed to probe CPU DAI %s: %d\n", cpu_dai->name, ret); - module_put(cpu_dai->dev->driver->owner); return ret; } } @@ -1654,17 +1501,24 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; - const char *codecname = aux_dev->codec_name; + const char *name = aux_dev->codec_name; - rtd->codec = soc_find_codec(aux_dev->codec_of_node, codecname); - if (!rtd->codec) { + rtd->component = soc_find_component(aux_dev->codec_of_node, name); + if (!rtd->component) { if (aux_dev->codec_of_node) - codecname = of_node_full_name(aux_dev->codec_of_node); + name = of_node_full_name(aux_dev->codec_of_node); - dev_err(card->dev, "ASoC: %s not registered\n", codecname); + dev_err(card->dev, "ASoC: %s not registered\n", name); return -EPROBE_DEFER; } + /* + * Some places still reference rtd->codec, so we have to keep that + * initialized if the component is a CODEC. Once all those references + * have been removed, this code can be removed as well. + */ + rtd->codec = rtd->component->codec; + return 0; } @@ -1674,18 +1528,13 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; int ret; - if (rtd->codec->probed) { - dev_err(rtd->codec->dev, "ASoC: codec already probed\n"); - return -EBUSY; - } - - ret = soc_probe_codec(card, rtd->codec); + ret = soc_probe_component(card, rtd->component); if (ret < 0) return ret; /* do machine specific initialization */ if (aux_dev->init) { - ret = aux_dev->init(&rtd->codec->dapm); + ret = aux_dev->init(rtd->component); if (ret < 0) { dev_err(card->dev, "ASoC: failed to init %s: %d\n", aux_dev->name, ret); @@ -1699,7 +1548,7 @@ static int soc_probe_aux_dev(struct snd_soc_card *card, int num) static void soc_remove_aux_dev(struct snd_soc_card *card, int num) { struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_component *component = rtd->component; /* unregister the rtd device */ if (rtd->dev_registered) { @@ -1708,8 +1557,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) rtd->dev_registered = 0; } - if (codec && codec->probed) - soc_remove_codec(codec); + if (component && component->probed) + soc_remove_component(component); } static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) @@ -3199,7 +3048,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, unsigned int val, mask; void *data; - if (!component->regmap) + if (!component->regmap || !params->num_regs) return -EINVAL; len = params->num_regs * component->val_bytes; @@ -4112,6 +3961,8 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; + component->probe = component->driver->probe; + component->remove = component->driver->remove; if (!component->dapm_ptr) component->dapm_ptr = &component->dapm; @@ -4120,19 +3971,42 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; + dapm->idle_bias_off = true; if (driver->seq_notifier) dapm->seq_notifier = snd_soc_component_seq_notifier; if (driver->stream_event) dapm->stream_event = snd_soc_component_stream_event; + component->controls = driver->controls; + component->num_controls = driver->num_controls; + component->dapm_widgets = driver->dapm_widgets; + component->num_dapm_widgets = driver->num_dapm_widgets; + component->dapm_routes = driver->dapm_routes; + component->num_dapm_routes = driver->num_dapm_routes; + INIT_LIST_HEAD(&component->dai_list); mutex_init(&component->io_mutex); return 0; } +static void snd_soc_component_init_regmap(struct snd_soc_component *component) +{ + if (!component->regmap) + component->regmap = dev_get_regmap(component->dev, NULL); + if (component->regmap) { + int val_bytes = regmap_get_val_bytes(component->regmap); + /* Errors are legitimate for non-integer byte multiples */ + if (val_bytes > 0) + component->val_bytes = val_bytes; + } +} + static void snd_soc_component_add_unlocked(struct snd_soc_component *component) { + if (!component->write && !component->read) + snd_soc_component_init_regmap(component); + list_add(&component->list, &component_list); } @@ -4221,22 +4095,18 @@ found: } EXPORT_SYMBOL_GPL(snd_soc_unregister_component); -static int snd_soc_platform_drv_write(struct snd_soc_component *component, - unsigned int reg, unsigned int val) +static int snd_soc_platform_drv_probe(struct snd_soc_component *component) { struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - return platform->driver->write(platform, reg, val); + return platform->driver->probe(platform); } -static int snd_soc_platform_drv_read(struct snd_soc_component *component, - unsigned int reg, unsigned int *val) +static void snd_soc_platform_drv_remove(struct snd_soc_component *component) { struct snd_soc_platform *platform = snd_soc_component_to_platform(component); - *val = platform->driver->read(platform, reg); - - return 0; + platform->driver->remove(platform); } /** @@ -4257,10 +4127,15 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, platform->dev = dev; platform->driver = platform_drv; - if (platform_drv->write) - platform->component.write = snd_soc_platform_drv_write; - if (platform_drv->read) - platform->component.read = snd_soc_platform_drv_read; + + if (platform_drv->probe) + platform->component.probe = snd_soc_platform_drv_probe; + if (platform_drv->remove) + platform->component.remove = snd_soc_platform_drv_remove; + +#ifdef CONFIG_DEBUG_FS + platform->component.debugfs_prefix = "platform"; +#endif mutex_lock(&client_mutex); snd_soc_component_add_unlocked(&platform->component); @@ -4382,6 +4257,20 @@ static void fixup_codec_formats(struct snd_soc_pcm_stream *stream) stream->formats |= codec_format_map[i]; } +static int snd_soc_codec_drv_probe(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + return codec->driver->probe(codec); +} + +static void snd_soc_codec_drv_remove(struct snd_soc_component *component) +{ + struct snd_soc_codec *codec = snd_soc_component_to_codec(component); + + codec->driver->remove(codec); +} + static int snd_soc_codec_drv_write(struct snd_soc_component *component, unsigned int reg, unsigned int val) { @@ -4420,7 +4309,6 @@ int snd_soc_register_codec(struct device *dev, { struct snd_soc_codec *codec; struct snd_soc_dai *dai; - struct regmap *regmap; int ret, i; dev_dbg(dev, "codec register %s\n", dev_name(dev)); @@ -4430,18 +4318,36 @@ int snd_soc_register_codec(struct device *dev, return -ENOMEM; codec->component.dapm_ptr = &codec->dapm; + codec->component.codec = codec; ret = snd_soc_component_initialize(&codec->component, &codec_drv->component_driver, dev); if (ret) goto err_free; + if (codec_drv->controls) { + codec->component.controls = codec_drv->controls; + codec->component.num_controls = codec_drv->num_controls; + } + if (codec_drv->dapm_widgets) { + codec->component.dapm_widgets = codec_drv->dapm_widgets; + codec->component.num_dapm_widgets = codec_drv->num_dapm_widgets; + } + if (codec_drv->dapm_routes) { + codec->component.dapm_routes = codec_drv->dapm_routes; + codec->component.num_dapm_routes = codec_drv->num_dapm_routes; + } + + if (codec_drv->probe) + codec->component.probe = snd_soc_codec_drv_probe; + if (codec_drv->remove) + codec->component.remove = snd_soc_codec_drv_remove; if (codec_drv->write) codec->component.write = snd_soc_codec_drv_write; if (codec_drv->read) codec->component.read = snd_soc_codec_drv_read; codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; - codec->dapm.codec = codec; + codec->dapm.idle_bias_off = codec_drv->idle_bias_off; if (codec_drv->seq_notifier) codec->dapm.seq_notifier = codec_drv->seq_notifier; if (codec_drv->set_bias_level) @@ -4451,23 +4357,13 @@ int snd_soc_register_codec(struct device *dev, codec->component.val_bytes = codec_drv->reg_word_size; mutex_init(&codec->mutex); - if (!codec->component.write) { - if (codec_drv->get_regmap) - regmap = codec_drv->get_regmap(dev); - else - regmap = dev_get_regmap(dev, NULL); - - if (regmap) { - ret = snd_soc_component_init_io(&codec->component, - regmap); - if (ret) { - dev_err(codec->dev, - "Failed to set cache I/O:%d\n", - ret); - goto err_cleanup; - } - } - } +#ifdef CONFIG_DEBUG_FS + codec->component.init_debugfs = soc_init_codec_debugfs; + codec->component.debugfs_prefix = "codec"; +#endif + + if (codec_drv->get_regmap) + codec->component.regmap = codec_drv->get_regmap(dev); for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8348352dc2c6..79f3b1eaf3b1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2860,12 +2860,14 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - int ret = 0; - if (e->reg != SND_SOC_NOPM) - ret = soc_dapm_read(dapm, e->reg, ®_val); - else + if (e->reg != SND_SOC_NOPM) { + int ret = soc_dapm_read(dapm, e->reg, ®_val); + if (ret) + return ret; + } else { reg_val = dapm_kcontrol_get_value(kcontrol); + } val = (reg_val >> e->shift_l) & e->mask; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); @@ -2875,7 +2877,7 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[1] = val; } - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); @@ -3107,7 +3109,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->codec; + if (dapm->component) + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6307f85e871b..b329b84bc5af 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -336,10 +336,12 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { }; static const struct snd_soc_platform_driver dmaengine_pcm_platform = { + .component_driver = { + .probe_order = SND_SOC_COMP_ORDER_LATE, + }, .ops = &dmaengine_pcm_ops, .pcm_new = dmaengine_pcm_new, .pcm_free = dmaengine_pcm_free, - .probe_order = SND_SOC_COMP_ORDER_LATE, }; static const char * const dmaengine_pcm_dma_channel_names[] = { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 7767fbd73eb7..9b3939049cef 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -271,31 +271,3 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, return snd_soc_component_write(&platform->component, reg, val); } EXPORT_SYMBOL_GPL(snd_soc_platform_write); - -/** - * snd_soc_component_init_io() - Initialize regmap IO - * - * @component: component to initialize - * @regmap: regmap instance to use for IO operations - * - * Return: 0 on success, a negative error code otherwise - */ -int snd_soc_component_init_io(struct snd_soc_component *component, - struct regmap *regmap) -{ - int ret; - - if (!regmap) - return -EINVAL; - - ret = regmap_get_val_bytes(regmap); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - component->val_bytes = ret; - - component->regmap = regmap; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_component_init_io); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 731fdb5b5f9b..642c86240752 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2352,7 +2352,11 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); fe->dpcm[stream].runtime = fe_substream->runtime; - if (dpcm_path_get(fe, stream, &list) <= 0) { + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) { + mutex_unlock(&fe->card->mutex); + return ret; + } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 0e5a8f35d0ad..a7dc3c56f44d 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -4,7 +4,7 @@ * sound/soc/spear/spear_pcm.c * * Copyright (C) 2012 ST Microelectronics - * Rajeev Kumar<rajeev-dlh.kumar@st.com> + * Rajeev Kumar<rajeevkumar.linux@gmail.com> * * This file is licensed under the terms of the GNU General Public * License version 2. This program is licensed "as is" without any @@ -50,6 +50,6 @@ int devm_spear_pcm_platform_register(struct device *dev, } EXPORT_SYMBOL_GPL(devm_spear_pcm_platform_register); -MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>"); +MODULE_AUTHOR("Rajeev Kumar <rajeevkumar.linux@gmail.com>"); MODULE_DESCRIPTION("SPEAr PCM DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 9577121ce971..ca8037634100 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -21,7 +21,7 @@ */ #ifndef __TEGRA_ASOC_UTILS_H__ -#define __TEGRA_ASOC_UTILS_H_ +#define __TEGRA_ASOC_UTILS_H__ struct clk; struct device; diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c index f65fc0987cfb..b7a7c805d63f 100644 --- a/sound/usb/caiaq/control.c +++ b/sound/usb/caiaq/control.c @@ -100,15 +100,19 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *cdev = caiaqdev(chip->card); int pos = kcontrol->private_value; int v = ucontrol->value.integer.value[0]; - unsigned char cmd = EP1_CMD_WRITE_IO; + unsigned char cmd; - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1)) - cmd = EP1_CMD_DIMM_LEDS; - - if (cdev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER)) + switch (cdev->chip.usb_id) { + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_MASCHINECONTROLLER): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER2): + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_KORECONTROLLER): cmd = EP1_CMD_DIMM_LEDS; + break; + default: + cmd = EP1_CMD_WRITE_IO; + break; + } if (pos & CNT_INTVAL) { int i = pos & ~CNT_INTVAL; |