diff options
Diffstat (limited to 'sound')
32 files changed, 217 insertions, 120 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 3fd1a7e..552b97a 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1073,10 +1073,10 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) sdev->pcmid = -1; list_del(&ldev->list); layouts_list_items--; + kfree(ldev); outnodev: of_node_put(sound); layout_device = NULL; - kfree(ldev); return -ENODEV; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 86d0caf..62e90b8 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t avail = 0; long wait_time, tout; + init_waitqueue_entry(&wait, current); + set_current_state(TASK_INTERRUPTIBLE); + add_wait_queue(&runtime->tsleep, &wait); + if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { @@ -1771,16 +1775,32 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } wait_time = msecs_to_jiffies(wait_time * 1000); } - init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->tsleep, &wait); + for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; break; } + + /* + * We need to check if space became available already + * (and thus the wakeup happened already) first to close + * the race of space already having become available. + * This check must happen after been added to the waitqueue + * and having current state be INTERRUPTIBLE. + */ + if (is_playback) + avail = snd_pcm_playback_avail(runtime); + else + avail = snd_pcm_capture_avail(runtime); + if (avail >= runtime->twake) + break; snd_pcm_stream_unlock_irq(substream); - tout = schedule_timeout_interruptible(wait_time); + + tout = schedule_timeout(wait_time); + snd_pcm_stream_lock_irq(substream); + set_current_state(TASK_INTERRUPTIBLE); switch (runtime->status->state) { case SNDRV_PCM_STATE_SUSPENDED: err = -ESTRPIPE; @@ -1806,14 +1826,9 @@ static int wait_for_avail(struct snd_pcm_substream *substream, err = -EIO; break; } - if (is_playback) - avail = snd_pcm_playback_avail(runtime); - else - avail = snd_pcm_capture_avail(runtime); - if (avail >= runtime->twake) - break; } _endloop: + set_current_state(TASK_RUNNING); remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3e7850c..f3aefef 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -579,9 +579,13 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, return -1; } recursive++; - for (i = 0; i < nums; i++) + for (i = 0; i < nums; i++) { + unsigned int type = get_wcaps_type(get_wcaps(codec, conn[i])); + if (type == AC_WID_PIN || type == AC_WID_AUD_OUT) + continue; if (snd_hda_get_conn_index(codec, conn[i], nid, recursive) >= 0) return i; + } return -1; } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d..c34f730 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -144,25 +144,17 @@ static int cea_sampling_frequencies[8] = { SNDRV_PCM_RATE_192000, /* 7: 192000Hz */ }; -static unsigned char hdmi_get_eld_byte(struct hda_codec *codec, hda_nid_t nid, +static unsigned int hdmi_get_eld_data(struct hda_codec *codec, hda_nid_t nid, int byte_index) { unsigned int val; val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_ELDD, byte_index); - #ifdef BE_PARANOID printk(KERN_INFO "HDMI: ELD data byte %d: 0x%x\n", byte_index, val); #endif - - if ((val & AC_ELDD_ELD_VALID) == 0) { - snd_printd(KERN_INFO "HDMI: invalid ELD data byte %d\n", - byte_index); - val = 0; - } - - return val & AC_ELDD_ELD_DATA; + return val; } #define GRAB_BITS(buf, byte, lowbit, bits) \ @@ -344,11 +336,26 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, if (!buf) return -ENOMEM; - for (i = 0; i < size; i++) - buf[i] = hdmi_get_eld_byte(codec, nid, i); + for (i = 0; i < size; i++) { + unsigned int val = hdmi_get_eld_data(codec, nid, i); + if (!(val & AC_ELDD_ELD_VALID)) { + if (!i) { + snd_printd(KERN_INFO + "HDMI: invalid ELD data\n"); + ret = -EINVAL; + goto error; + } + snd_printd(KERN_INFO + "HDMI: invalid ELD data byte %d\n", i); + val = 0; + } else + val &= AC_ELDD_ELD_DATA; + buf[i] = val; + } ret = hdmi_update_eld(eld, buf, size); +error: kfree(buf); return ret; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 47d6ffc..c45f3e6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -375,7 +375,7 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, unsigned int *idxp) { - int i; + int i, idx; hda_nid_t nid; nid = codec->start_nid; @@ -384,9 +384,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, type = get_wcaps_type(get_wcaps(codec, nid)); if (type != AC_WID_AUD_IN) continue; - *idxp = snd_hda_get_conn_index(codec, nid, pin, false); - if (*idxp >= 0) + idx = snd_hda_get_conn_index(codec, nid, pin, false); + if (idx >= 0) { + *idxp = idx; return nid; + } } return 0; } @@ -533,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) { - char tmp[32]; + char tmp[44]; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT); knew.private_value = pval; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 502fc94..7696d05 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3348,6 +3348,8 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t pin, #define MAX_AUTO_DACS 5 +#define DAC_SLAVE_FLAG 0x8000 /* filled dac is a slave */ + /* fill analog DAC list from the widget tree */ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) { @@ -3370,16 +3372,26 @@ static int fill_cx_auto_dacs(struct hda_codec *codec, hda_nid_t *dacs) /* fill pin_dac_pair list from the pin and dac list */ static int fill_dacs_for_pins(struct hda_codec *codec, hda_nid_t *pins, int num_pins, hda_nid_t *dacs, int *rest, - struct pin_dac_pair *filled, int type) + struct pin_dac_pair *filled, int nums, + int type) { - int i, nums; + int i, start = nums; - nums = 0; - for (i = 0; i < num_pins; i++) { + for (i = 0; i < num_pins; i++, nums++) { filled[nums].pin = pins[i]; filled[nums].type = type; filled[nums].dac = get_unassigned_dac(codec, pins[i], dacs, rest); - nums++; + if (filled[nums].dac) + continue; + if (filled[start].dac && get_connection_index(codec, pins[i], filled[start].dac) >= 0) { + filled[nums].dac = filled[start].dac | DAC_SLAVE_FLAG; + continue; + } + if (filled[0].dac && get_connection_index(codec, pins[i], filled[0].dac) >= 0) { + filled[nums].dac = filled[0].dac | DAC_SLAVE_FLAG; + continue; + } + snd_printdd("Failed to find a DAC for pin 0x%x", pins[i]); } return nums; } @@ -3395,19 +3407,19 @@ static void cx_auto_parse_output(struct hda_codec *codec) rest = fill_cx_auto_dacs(codec, dacs); /* parse all analog output pins */ nums = fill_dacs_for_pins(codec, cfg->line_out_pins, cfg->line_outs, - dacs, &rest, spec->dac_info, - AUTO_PIN_LINE_OUT); - nums += fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_HP_OUT); - nums += fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, - dacs, &rest, spec->dac_info + nums, - AUTO_PIN_SPEAKER_OUT); + dacs, &rest, spec->dac_info, 0, + AUTO_PIN_LINE_OUT); + nums = fill_dacs_for_pins(codec, cfg->hp_pins, cfg->hp_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_HP_OUT); + nums = fill_dacs_for_pins(codec, cfg->speaker_pins, cfg->speaker_outs, + dacs, &rest, spec->dac_info, nums, + AUTO_PIN_SPEAKER_OUT); spec->dac_info_filled = nums; /* fill multiout struct */ for (i = 0; i < nums; i++) { hda_nid_t dac = spec->dac_info[i].dac; - if (!dac) + if (!dac || (dac & DAC_SLAVE_FLAG)) continue; switch (spec->dac_info[i].type) { case AUTO_PIN_LINE_OUT: @@ -3862,7 +3874,7 @@ static void cx_auto_parse_input(struct hda_codec *codec) } if (imux->num_items >= 2 && cfg->num_inputs == imux->num_items) cx_auto_check_auto_mic(codec); - if (imux->num_items > 1 && !spec->auto_mic) { + if (imux->num_items > 1) { for (i = 1; i < imux->num_items; i++) { if (spec->imux_info[i].adc != spec->imux_info[0].adc) { spec->adc_switching = 1; @@ -4035,6 +4047,8 @@ static void cx_auto_init_output(struct hda_codec *codec) nid = spec->dac_info[i].dac; if (!nid) nid = spec->multiout.dac_nids[0]; + else if (nid & DAC_SLAVE_FLAG) + nid &= ~DAC_SLAVE_FLAG; select_connection(codec, spec->dac_info[i].pin, nid); } if (spec->auto_mute) { @@ -4167,9 +4181,11 @@ static int try_add_pb_volume(struct hda_codec *codec, hda_nid_t dac, hda_nid_t pin, const char *name, int idx) { unsigned int caps; - caps = query_amp_caps(codec, dac, HDA_OUTPUT); - if (caps & AC_AMPCAP_NUM_STEPS) - return cx_auto_add_pb_volume(codec, dac, name, idx); + if (dac && !(dac & DAC_SLAVE_FLAG)) { + caps = query_amp_caps(codec, dac, HDA_OUTPUT); + if (caps & AC_AMPCAP_NUM_STEPS) + return cx_auto_add_pb_volume(codec, dac, name, idx); + } caps = query_amp_caps(codec, pin, HDA_OUTPUT); if (caps & AC_AMPCAP_NUM_STEPS) return cx_auto_add_pb_volume(codec, pin, name, idx); @@ -4191,8 +4207,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) for (i = 0; i < spec->dac_info_filled; i++) { const char *label; int idx, type; - if (!spec->dac_info[i].dac) - continue; + hda_nid_t dac = spec->dac_info[i].dac; type = spec->dac_info[i].type; if (type == AUTO_PIN_LINE_OUT) type = spec->autocfg.line_out_type; @@ -4211,7 +4226,7 @@ static int cx_auto_build_output_controls(struct hda_codec *codec) idx = num_spk++; break; } - err = try_add_pb_volume(codec, spec->dac_info[i].dac, + err = try_add_pb_volume(codec, dac, spec->dac_info[i].pin, label, idx); if (err < 0) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fcb11af..7cabd73 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -565,11 +565,11 @@ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute) - return; spec->jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); + if (!spec->automute) + return; update_speakers(codec); } @@ -578,11 +578,11 @@ static void alc_line_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (!spec->automute || !spec->detect_line) - return; spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); + if (!spec->automute || !spec->detect_line) + return; update_speakers(codec); } @@ -3083,16 +3083,22 @@ static void alc_auto_init_multi_out(struct hda_codec *codec) static void alc_auto_init_extra_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; + hda_nid_t pin, dac; pin = spec->autocfg.hp_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_HP, - spec->multiout.hp_nid); + if (pin) { + dac = spec->multiout.hp_nid; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_HP, dac); + } pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, - spec->multiout.extra_out_nid[0]); + if (pin) { + dac = spec->multiout.extra_out_nid[0]; + if (!dac) + dac = spec->multiout.dac_nids[0]; + alc_auto_set_output_and_unmute(codec, pin, PIN_OUT, dac); + } } /* diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index d6651c0..5956584 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -56,7 +56,7 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 48000: - clk = 12288000; + clk = 24576000; break; } @@ -103,7 +103,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, { @@ -112,7 +112,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name ="ad193x-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad193x.5", + .codec_name = "spi0.5", .ops = &bf5xx_ad193x_ops, }, }; diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 2374ca5..eedb6f5 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -27,11 +27,6 @@ struct ad193x_priv { int sysclk; }; -/* ad193x register cache & default register settings */ -static const u8 ad193x_reg[AD193X_NUM_REGS] = { - 0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0, -}; - /* * AD193X volume/mute/de-emphasis etc. controls */ @@ -307,7 +302,8 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg); reg = snd_soc_read(codec, AD193X_DAC_CTRL2); - reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) | word_len; + reg = (reg & (~AD193X_DAC_WORD_LEN_MASK)) + | (word_len << AD193X_DAC_WORD_LEN_SHFT); snd_soc_write(codec, AD193X_DAC_CTRL2, reg); reg = snd_soc_read(codec, AD193X_ADC_CTRL1); @@ -389,9 +385,6 @@ static int ad193x_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ad193x = { .probe = ad193x_probe, - .reg_cache_default = ad193x_reg, - .reg_cache_size = AD193X_NUM_REGS, - .reg_word_size = sizeof(u16), }; #if defined(CONFIG_SPI_MASTER) diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 9747b54..cccc2e8 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -34,7 +34,8 @@ #define AD193X_DAC_LEFT_HIGH (1 << 3) #define AD193X_DAC_BCLK_INV (1 << 7) #define AD193X_DAC_CTRL2 0x804 -#define AD193X_DAC_WORD_LEN_MASK 0xC +#define AD193X_DAC_WORD_LEN_SHFT 3 +#define AD193X_DAC_WORD_LEN_MASK 0x18 #define AD193X_DAC_MASTER_MUTE 1 #define AD193X_DAC_CHNL_MUTE 0x805 #define AD193X_DACL1_MUTE 0 @@ -63,7 +64,7 @@ #define AD193X_ADC_CTRL1 0x80f #define AD193X_ADC_SERFMT_MASK 0x60 #define AD193X_ADC_SERFMT_STEREO (0 << 5) -#define AD193X_ADC_SERFMT_TDM (1 << 2) +#define AD193X_ADC_SERFMT_TDM (1 << 5) #define AD193X_ADC_SERFMT_AUX (2 << 5) #define AD193X_ADC_WORD_LEN_MASK 0x3 #define AD193X_ADC_CTRL2 0x810 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 409d89d..fbd7eb9 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -857,6 +857,7 @@ static __devinit int sta32x_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &sta32x_codec, &sta32x_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); + kfree(sta32x); return ret; } diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 60d740e..1725550 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2221,6 +2221,8 @@ static int sysclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: if (fll) { + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_ENA, WM8962_FLL_ENA); if (wm8962->irq) { @@ -2927,10 +2929,6 @@ static int wm8962_set_bias_level(struct snd_soc_codec *codec, WM8962_BIAS_ENA | 0x180); msleep(5); - - snd_soc_update_bits(codec, WM8962_CLOCKING2, - WM8962_CLKREG_OVD, - WM8962_CLKREG_OVD); } /* VMID 2*250k */ @@ -3288,6 +3286,8 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda); snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n); + try_wait_for_completion(&wm8962->fll_lock); + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | WM8962_FLL_ENA, fll1); @@ -3868,6 +3868,10 @@ static int wm8962_probe(struct snd_soc_codec *codec) */ snd_soc_update_bits(codec, WM8962_CLOCKING2, WM8962_SYSCLK_ENA, 0); + /* Ensure we have soft control over all registers */ + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_CLKREG_OVD, WM8962_CLKREG_OVD); + regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies); if (pdata) { diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index ab8e9d1..0cdb9d1 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -420,7 +420,7 @@ static const char *sidetone_hpf_text[] = { }; static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 6, sidetone_hpf_text); + SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text); static const char *hpf_mode_text[] = { "HiFi", "Custom", "Voice" @@ -988,15 +988,10 @@ SND_SOC_DAPM_MICBIAS("MICB1", WM8996_POWER_MANAGEMENT_1, 8, 0), SND_SOC_DAPM_PGA("IN1L PGA", WM8996_POWER_MANAGEMENT_2, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("IN1R PGA", WM8996_POWER_MANAGEMENT_2, 4, 0, NULL, 0), -SND_SOC_DAPM_MUX("IN1L Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN1R Mux", SND_SOC_NOPM, 0, 0, &in1_mux), -SND_SOC_DAPM_MUX("IN2L Mux", SND_SOC_NOPM, 0, 0, &in2_mux), -SND_SOC_DAPM_MUX("IN2R Mux", SND_SOC_NOPM, 0, 0, &in2_mux), - -SND_SOC_DAPM_PGA("IN1L", WM8996_POWER_MANAGEMENT_7, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN1R", WM8996_POWER_MANAGEMENT_7, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2L", WM8996_POWER_MANAGEMENT_7, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA("IN2R", WM8996_POWER_MANAGEMENT_7, 7, 0, NULL, 0), +SND_SOC_DAPM_MUX("IN1L Mux", WM8996_POWER_MANAGEMENT_7, 2, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN1R Mux", WM8996_POWER_MANAGEMENT_7, 3, 0, &in1_mux), +SND_SOC_DAPM_MUX("IN2L Mux", WM8996_POWER_MANAGEMENT_7, 6, 0, &in2_mux), +SND_SOC_DAPM_MUX("IN2R Mux", WM8996_POWER_MANAGEMENT_7, 7, 0, &in2_mux), SND_SOC_DAPM_SUPPLY("DMIC2", WM8996_POWER_MANAGEMENT_7, 9, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC1", WM8996_POWER_MANAGEMENT_7, 8, 0, NULL, 0), @@ -1213,6 +1208,16 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "AIF2RX0", NULL, "AIFCLK" }, { "AIF2RX1", NULL, "AIFCLK" }, + { "AIF1TX0", NULL, "AIFCLK" }, + { "AIF1TX1", NULL, "AIFCLK" }, + { "AIF1TX2", NULL, "AIFCLK" }, + { "AIF1TX3", NULL, "AIFCLK" }, + { "AIF1TX4", NULL, "AIFCLK" }, + { "AIF1TX5", NULL, "AIFCLK" }, + + { "AIF2TX0", NULL, "AIFCLK" }, + { "AIF2TX1", NULL, "AIFCLK" }, + { "DSP1RXL", NULL, "SYSDSPCLK" }, { "DSP1RXR", NULL, "SYSDSPCLK" }, { "DSP2RXL", NULL, "SYSDSPCLK" }, @@ -2106,6 +2111,9 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Clear any pending completions (eg, from failed startups) */ + try_wait_for_completion(&wm8996->fll_lock); + snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, WM8996_FLL_ENA); diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 56efa0c..099614e 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -385,14 +385,14 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { err = -ENODEV; - goto fail; + goto fail_free_info; } info->mem = request_mem_region(res->start, resource_size(res), pdev->name); if (!info->mem) { err = -EBUSY; - goto fail; + goto fail_free_info; } info->regs = ioremap(info->mem->start, resource_size(info->mem)); @@ -435,6 +435,7 @@ fail_unmap_mem: iounmap(info->regs); fail_release_mem: release_mem_region(info->mem->start, resource_size(info->mem)); +fail_free_info: kfree(info); fail: return err; diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 732208c..cb50598 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -879,10 +879,12 @@ static struct device_node *find_ssi_node(struct device_node *dma_channel_np) * assume that device_node pointers are a valid comparison. */ np = of_parse_phandle(ssi_np, "fsl,playback-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; np = of_parse_phandle(ssi_np, "fsl,capture-dma", 0); + of_node_put(np); if (np == dma_channel_np) return ssi_np; } diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fd0dc46..5c6c245 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -369,7 +369,7 @@ static struct snd_soc_platform_driver mpc5200_audio_dma_platform = { .pcm_free = &psc_dma_free, }; -static int mpc5200_hpcd_probe(struct of_device *op) +static int mpc5200_hpcd_probe(struct platform_device *op) { phys_addr_t fifo; struct psc_dma *psc_dma; @@ -487,7 +487,7 @@ out_unmap: return ret; } -static int mpc5200_hpcd_remove(struct of_device *op) +static int mpc5200_hpcd_remove(struct platform_device *op) { struct psc_dma *psc_dma = dev_get_drvdata(&op->dev); @@ -519,7 +519,7 @@ MODULE_DEVICE_TABLE(of, mpc5200_hpcd_match); static struct platform_driver mpc5200_hpcd_of_driver = { .probe = mpc5200_hpcd_probe, .remove = mpc5200_hpcd_remove, - .dev = { + .driver = { .owner = THIS_MODULE, .name = "mpc5200-pcm-audio", .of_match_table = mpc5200_hpcd_match, diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index a192979..358f0ba 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -345,8 +345,10 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) } machine_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); - if (!machine_data) - return -ENOMEM; + if (!machine_data) { + ret = -ENOMEM; + goto error_alloc; + } machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; @@ -494,7 +496,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) ret = platform_device_add(sound_device); if (ret) { dev_err(&pdev->dev, "platform device add failed\n"); - goto error; + goto error_sound; } dev_set_drvdata(&pdev->dev, sound_device); @@ -502,14 +504,12 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) return 0; +error_sound: + platform_device_unregister(sound_device); error: - of_node_put(codec_np); - - if (sound_device) - platform_device_unregister(sound_device); - kfree(machine_data); - +error_alloc: + of_node_put(codec_np); return ret; } diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 8fa4d5f..fcb862e 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -297,8 +297,10 @@ static int get_dma_channel(struct device_node *ssi_np, * dai->platform name should already point to an allocated buffer. */ ret = of_address_to_resource(dma_channel_np, 0, &res); - if (ret) + if (ret) { + of_node_put(dma_channel_np); return ret; + } snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", (unsigned long long) res.start, dma_channel_np->name); diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 309c59e..7945625 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -240,7 +240,6 @@ static int ssi_irq = 0; static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int ret; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index a33fc51..d0bcf3f 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!priv->mem) { dev_err(&pdev->dev, "request_mem_region failed\n"); err = -EBUSY; - goto error; + goto err_alloc; } priv->io = ioremap(priv->mem->start, SZ_16K); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 30fe0d0..0aa475f 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -514,7 +514,7 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) } /* Set codec bias level */ - ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ @@ -649,7 +649,9 @@ static void __exit ams_delta_module_exit(void) ams_delta_hook_switch_gpios); /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + ams_delta_set_bias_level(&ams_delta_audio_card, + &ams_delta_audio_card.rtd[0].codec->dapm, + SND_SOC_BIAS_STANDBY); platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index b99091f..65f980e 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -185,6 +185,7 @@ config SND_SOC_SPEYSIDE select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 + select SND_SOC_WM1250_EV1 config SND_SOC_SPEYSIDE_WM8962 tristate "Audio support for Wolfson Speyside with WM8962" diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 241f55d..c6c6589 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -13,6 +13,7 @@ * */ +#include <linux/types.h> #include <linux/gpio.h> #include <sound/soc.h> diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 1e574a5..bc8c167 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -17,6 +17,7 @@ * */ +#include <linux/types.h> #include <linux/gpio.h> #include <sound/soc.h> diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 0b9eb5f..72535f2 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -23,6 +23,9 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { @@ -57,6 +60,9 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; int ret; + if (dapm->dev != codec_dai->dev) + return 0; + switch (level) { case SND_SOC_BIAS_STANDBY: ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index d9f8ade..20b7f3b 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -203,14 +203,14 @@ static int snd_soc_rbtree_cache_sync(struct snd_soc_codec *codec) rbnode = rb_entry(node, struct snd_soc_rbtree_node, node); for (i = 0; i < rbnode->blklen; ++i) { regtmp = rbnode->base_reg + i; - WARN_ON(codec->writable_register && - codec->writable_register(codec, regtmp)); val = snd_soc_rbtree_get_register(rbnode, i); def = snd_soc_get_cache_val(codec->reg_def_copy, i, rbnode->word_size); if (val == def) continue; + WARN_ON(!snd_soc_codec_writable_register(codec, regtmp)); + codec->cache_bypass = 1; ret = snd_soc_write(codec, regtmp, val); codec->cache_bypass = 0; @@ -563,8 +563,7 @@ static int snd_soc_lzo_cache_sync(struct snd_soc_codec *codec) lzo_blocks = codec->reg_cache; for_each_set_bit(i, lzo_blocks[0]->sync_bmp, lzo_blocks[0]->sync_bmp_nbits) { - WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); + WARN_ON(!snd_soc_codec_writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; @@ -823,8 +822,6 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) codec_drv = codec->driver; for (i = 0; i < codec_drv->reg_cache_size; ++i) { - WARN_ON(codec->writable_register && - codec->writable_register(codec, i)); ret = snd_soc_cache_read(codec, i, &val); if (ret) return ret; @@ -832,6 +829,9 @@ static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) if (snd_soc_get_cache_val(codec->reg_def_copy, i, codec_drv->reg_word_size) == val) continue; + + WARN_ON(!snd_soc_codec_writable_register(codec, i)); + ret = snd_soc_write(codec, i, val); if (ret) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 83ad8ca..d2ef014 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1633,7 +1633,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec, if (codec->readable_register) return codec->readable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_readable_register); @@ -1651,7 +1651,7 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, if (codec->writable_register) return codec->writable_register(codec, reg); else - return 0; + return 1; } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); @@ -1913,7 +1913,7 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, if (prefix) { name_len = strlen(long_name) + strlen(prefix) + 2; - name = kmalloc(name_len, GFP_ATOMIC); + name = kmalloc(name_len, GFP_KERNEL); if (!name) return NULL; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e15914..d67c637 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2763,7 +2763,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_ignore_suspend); /** * snd_soc_dapm_free - free dapm resources - * @card: SoC device + * @dapm: DAPM context * * Free all dapm widgets and resources. */ diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c..a62f7dd 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -205,6 +205,25 @@ static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, #define snd_soc_16_8_read_i2c NULL #endif +#if defined(CONFIG_SPI_MASTER) +static unsigned int snd_soc_16_8_read_spi(struct snd_soc_codec *codec, + unsigned int r) +{ + struct spi_device *spi = codec->control_data; + + const u16 reg = cpu_to_be16(r | 0x100); + u8 data; + int ret; + + ret = spi_write_then_read(spi, ®, 2, &data, 1); + if (ret < 0) + return 0; + return data; +} +#else +#define snd_soc_16_8_read_spi NULL +#endif + static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -295,6 +314,7 @@ static struct { int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); + unsigned int (*spi_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { .addr_bits = 4, .data_bits = 12, @@ -318,6 +338,7 @@ static struct { .addr_bits = 16, .data_bits = 8, .write = snd_soc_16_8_write, .i2c_read = snd_soc_16_8_read_i2c, + .spi_read = snd_soc_16_8_read_spi, }, { .addr_bits = 16, .data_bits = 16, @@ -383,6 +404,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, #ifdef CONFIG_SPI_MASTER codec->hw_write = do_spi_write; #endif + if (io_types[i].spi_read) + codec->hw_read = io_types[i].spi_read; codec->control_data = container_of(codec->dev, struct spi_device, diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7c17b98..fa31d9c 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -105,7 +105,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_sync(dapm); - snd_jack_report(jack->jack, status); + snd_jack_report(jack->jack, jack->status); out: mutex_unlock(&codec->mutex); @@ -327,7 +327,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, IRQF_TRIGGER_FALLING, gpios[i].name, &gpios[i]); - if (ret) + if (ret < 0) goto err; if (gpios[i].wake) { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b575939..2879c88 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -290,6 +290,9 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; + if (!cpu_dai->active && !codec_dai->active) + rtd->rate = 0; + /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 661373c..be27f1d 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -319,7 +319,7 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); /* FIXME: Calculate automatically based on DAPM routes? */ - if (!machine_is_harmony() && !machine_is_ventana()) + if (!machine_is_harmony()) snd_soc_dapm_nc_pin(dapm, "IN1L"); if (!machine_is_seaboard() && !machine_is_aebl()) snd_soc_dapm_nc_pin(dapm, "IN1R"); @@ -395,7 +395,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (machine_is_harmony() || machine_is_ventana()) { + if (machine_is_harmony()) { card->dapm_routes = harmony_audio_map; card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); } else if (machine_is_seaboard()) { |