diff options
-rw-r--r-- | Documentation/sound/alsa/SB-Live-mixer.txt | 6 | ||||
-rw-r--r-- | MAINTAINERS | 2 | ||||
-rw-r--r-- | arch/arm/mach-s3c2440/mach-gta02.c | 5 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 4 | ||||
-rw-r--r-- | sound/pci/hda/patch_realtek.c | 25 | ||||
-rw-r--r-- | sound/soc/codecs/jz4740.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/sn95031.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/wm8903.c | 38 | ||||
-rw-r--r-- | sound/soc/codecs/wm8994.c | 16 | ||||
-rw-r--r-- | sound/soc/codecs/wm_hubs.c | 8 | ||||
-rw-r--r-- | sound/soc/mid-x86/sst_platform.c | 10 | ||||
-rw-r--r-- | sound/soc/samsung/pcm.c | 4 | ||||
-rw-r--r-- | sound/soc/sh/fsi.c | 22 | ||||
-rw-r--r-- | sound/soc/soc-core.c | 5 | ||||
-rw-r--r-- | sound/soc/tegra/harmony.c | 1 |
15 files changed, 110 insertions, 40 deletions
diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt index f5639d4..f4b5988 100644 --- a/Documentation/sound/alsa/SB-Live-mixer.txt +++ b/Documentation/sound/alsa/SB-Live-mixer.txt @@ -87,14 +87,14 @@ accumulator. ALSA uses accumulators 0 and 1 for left and right PCM. The result is forwarded to the ADC capture FIFO (thus to the standard capture PCM device). -name='Music Playback Volume',index=0 +name='Synth Playback Volume',index=0 This control is used to attenuate samples for left and right MIDI FX-bus accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples. The result samples are forwarded to the front DAC PCM slots of the AC97 codec. -name='Music Capture Volume',index=0 -name='Music Capture Switch',index=0 +name='Synth Capture Volume',index=0 +name='Synth Capture Switch',index=0 These controls are used to attenuate samples for left and right MIDI FX-bus accumulator. ALSA uses accumulators 4 and 5 for left and right PCM. diff --git a/MAINTAINERS b/MAINTAINERS index 1e2724e..1380312 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5396,7 +5396,7 @@ F: drivers/media/video/*7146* F: include/media/*7146* SAMSUNG AUDIO (ASoC) DRIVERS -M: Jassi Brar <jassi.brar@samsung.com> +M: Jassi Brar <jassisinghbrar@gmail.com> L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung diff --git a/arch/arm/mach-s3c2440/mach-gta02.c b/arch/arm/mach-s3c2440/mach-gta02.c index 0db2411..7166620 100644 --- a/arch/arm/mach-s3c2440/mach-gta02.c +++ b/arch/arm/mach-s3c2440/mach-gta02.c @@ -409,6 +409,10 @@ struct platform_device s3c24xx_pwm_device = { .num_resources = 0, }; +static struct platform_device gta02_dfbmcs320_device = { + .name = "dfbmcs320", +}; + static struct i2c_board_info gta02_i2c_devs[] __initdata = { { I2C_BOARD_INFO("pcf50633", 0x73), @@ -523,6 +527,7 @@ static struct platform_device *gta02_devices[] __initdata = { &s3c_device_iis, &samsung_asoc_dma, &s3c_device_i2c0, + >a02_dfbmcs320_device, >a02_buttons_device, &s3c_device_adc, &s3c_device_ts, diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 430f41d..759ade1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -937,6 +937,7 @@ void snd_hda_shutup_pins(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); +#ifdef SND_HDA_NEEDS_RESUME /* Restore the pin controls cleared previously via snd_hda_shutup_pins() */ static void restore_shutup_pins(struct hda_codec *codec) { @@ -953,6 +954,7 @@ static void restore_shutup_pins(struct hda_codec *codec) } codec->pins_shutup = 0; } +#endif static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); @@ -1329,6 +1331,7 @@ static void purify_inactive_streams(struct hda_codec *codec) } } +#ifdef SND_HDA_NEEDS_RESUME /* clean up all streams; called from suspend */ static void hda_cleanup_all_streams(struct hda_codec *codec) { @@ -1340,6 +1343,7 @@ static void hda_cleanup_all_streams(struct hda_codec *codec) really_cleanup_stream(codec, p); } } +#endif /* * amp access functions diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52928d9..d3bd2c1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14868,6 +14868,23 @@ static void alc269_fixup_hweq(struct hda_codec *codec, alc_write_coef_idx(codec, 0x1e, coef | 0x80); } +static void alc271_fixup_dmic(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + static struct hda_verb verbs[] = { + {0x20, AC_VERB_SET_COEF_INDEX, 0x0d}, + {0x20, AC_VERB_SET_PROC_COEF, 0x4000}, + {} + }; + unsigned int cfg; + + if (strcmp(codec->chip_name, "ALC271X")) + return; + cfg = snd_hda_codec_get_pincfg(codec, 0x12); + if (get_defcfg_connect(cfg) == AC_JACK_PORT_FIXED) + snd_hda_sequence_write(codec, verbs); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -14876,6 +14893,7 @@ enum { ALC269_FIXUP_ASUS_G73JW, ALC269_FIXUP_LENOVO_EAPD, ALC275_FIXUP_SONY_HWEQ, + ALC271_FIXUP_DMIC, }; static const struct alc_fixup alc269_fixups[] = { @@ -14929,7 +14947,11 @@ static const struct alc_fixup alc269_fixups[] = { .v.func = alc269_fixup_hweq, .chained = true, .chain_id = ALC275_FIXUP_SONY_VAIO_GPIO2 - } + }, + [ALC271_FIXUP_DMIC] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc271_fixup_dmic, + }, }; static struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -14938,6 +14960,7 @@ static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), + SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index f7cd346..f5ccdbf 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -308,8 +308,6 @@ static int jz4740_codec_dev_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, jz4740_codec_dapm_routes, ARRAY_SIZE(jz4740_codec_dapm_routes)); - snd_soc_dapm_new_widgets(codec); - jz4740_codec_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index a54d2a5..4d9fb27 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -927,7 +927,7 @@ static struct platform_driver sn95031_codec_driver = { .owner = THIS_MODULE, }, .probe = sn95031_device_probe, - .remove = sn95031_device_remove, + .remove = __devexit_p(sn95031_device_remove), }; static int __init sn95031_init(void) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ae1cadf..f52b623 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: - case WM8903_POWER_MANAGEMENT_3: - case WM8903_POWER_MANAGEMENT_2: case WM8903_DC_SERVO_READBACK_1: case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: @@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0, SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, - 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, +SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, + 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, 0, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0, +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0, +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), @@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" }, - { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" }, - { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, - { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_ENA", NULL, "Left Headphone Output PGA" }, + { "HPR_ENA", NULL, "Right Headphone Output PGA" }, + { "HPL_ENA_DLY", NULL, "HPL_ENA" }, + { "HPR_ENA_DLY", NULL, "HPR_ENA" }, + { "LINEOUTL_ENA", NULL, "Left Line Output PGA" }, + { "LINEOUTR_ENA", NULL, "Right Line Output PGA" }, + { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" }, + { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" }, { "HPL_DCS", NULL, "DCS Master" }, { "HPR_DCS", NULL, "DCS Master" }, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 3290333..84e1bd1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3261,20 +3261,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch volume updates (right only; we always do left then right). */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, + WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME, + WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME, + WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME, + WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_VU, WM8994_DAC1_VU); snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME, WM8994_DAC1_VU, WM8994_DAC1_VU); + snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_VU, WM8994_DAC2_VU); snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU, WM8994_DAC2_VU); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 7b6b3c1..4005e9a 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -740,12 +740,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKL", "Input Switch", "MIXINL" }, { "SPKL", "IN1LP Switch", "IN1LP" }, - { "SPKL", "Output Switch", "Left Output Mixer" }, + { "SPKL", "Output Switch", "Left Output PGA" }, { "SPKL", NULL, "TOCLK" }, { "SPKR", "Input Switch", "MIXINR" }, { "SPKR", "IN1RP Switch", "IN1RP" }, - { "SPKR", "Output Switch", "Right Output Mixer" }, + { "SPKR", "Output Switch", "Right Output PGA" }, { "SPKR", NULL, "TOCLK" }, { "SPKL Boost", "Direct Voice Switch", "Direct Voice" }, @@ -767,8 +767,8 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKOUTRP", NULL, "SPKR Driver" }, { "SPKOUTRN", NULL, "SPKR Driver" }, - { "Left Headphone Mux", "Mixer", "Left Output Mixer" }, - { "Right Headphone Mux", "Mixer", "Right Output Mixer" }, + { "Left Headphone Mux", "Mixer", "Left Output PGA" }, + { "Right Headphone Mux", "Mixer", "Right Output PGA" }, { "Headphone PGA", NULL, "Left Headphone Mux" }, { "Headphone PGA", NULL, "Right Headphone Mux" }, diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index b2e9198..d567c32 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -116,18 +116,20 @@ struct snd_soc_dai_driver sst_platform_dai[] = { static inline void sst_set_stream_status(struct sst_runtime_stream *stream, int state) { - spin_lock(&stream->status_lock); + unsigned long flags; + spin_lock_irqsave(&stream->status_lock, flags); stream->stream_status = state; - spin_unlock(&stream->status_lock); + spin_unlock_irqrestore(&stream->status_lock, flags); } static inline int sst_get_stream_status(struct sst_runtime_stream *stream) { int state; + unsigned long flags; - spin_lock(&stream->status_lock); + spin_lock_irqsave(&stream->status_lock, flags); state = stream->stream_status; - spin_unlock(&stream->status_lock); + spin_unlock_irqrestore(&stream->status_lock, flags); return state; } diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 38aac7d..9c7e8b4 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -350,8 +350,8 @@ static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, ctl = readl(regs + S3C_PCM_CTL); switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - /* Nothing to do, NB_NF by default */ + case SND_SOC_DAIFMT_IB_NF: + /* Nothing to do, IB_NF by default */ break; default: dev_err(pcm->dev, "Unsupported clock inversion!\n"); diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 0c9997e..23c0e83 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1200,10 +1200,11 @@ static int fsi_probe(struct platform_device *pdev) master->fsib.master = master; pm_runtime_enable(&pdev->dev); - pm_runtime_resume(&pdev->dev); dev_set_drvdata(&pdev->dev, master); + pm_runtime_get_sync(&pdev->dev); fsi_soft_all_reset(master); + pm_runtime_put_sync(&pdev->dev); ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, id_entry->name, master); @@ -1218,8 +1219,17 @@ static int fsi_probe(struct platform_device *pdev) goto exit_free_irq; } - return snd_soc_register_dais(&pdev->dev, fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); + ret = snd_soc_register_dais(&pdev->dev, fsi_soc_dai, + ARRAY_SIZE(fsi_soc_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "cannot snd dai register\n"); + goto exit_snd_soc; + } + + return ret; +exit_snd_soc: + snd_soc_unregister_platform(&pdev->dev); exit_free_irq: free_irq(irq, master); exit_iounmap: @@ -1238,12 +1248,11 @@ static int fsi_remove(struct platform_device *pdev) master = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); - snd_soc_unregister_platform(&pdev->dev); - + free_irq(master->irq, master); pm_runtime_disable(&pdev->dev); - free_irq(master->irq, master); + snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(fsi_soc_dai)); + snd_soc_unregister_platform(&pdev->dev); iounmap(master->base); kfree(master); @@ -1321,3 +1330,4 @@ module_exit(fsi_mobile_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SuperH onchip FSI audio driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>"); +MODULE_ALIAS("platform:fsi-pcm-audio"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b76b74d..d8562ce 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -629,6 +629,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rates |= codec_dai_drv->capture.rates; } + ret = -EINVAL; snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", @@ -640,7 +641,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) codec_dai->name, cpu_dai->name); goto config_err; } - if (!runtime->hw.channels_min || !runtime->hw.channels_max) { + if (!runtime->hw.channels_min || !runtime->hw.channels_max || + runtime->hw.channels_min > runtime->hw.channels_max) { printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", codec_dai->name, cpu_dai->name); goto config_err; @@ -2060,6 +2062,7 @@ const struct dev_pm_ops snd_soc_pm_ops = { .resume = snd_soc_resume, .poweroff = snd_soc_poweroff, }; +EXPORT_SYMBOL_GPL(snd_soc_pm_ops); /* ASoC platform driver */ static struct platform_driver soc_driver = { diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 8585957..556a571 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -370,6 +370,7 @@ static struct platform_driver tegra_snd_harmony_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = tegra_snd_harmony_probe, .remove = __devexit_p(tegra_snd_harmony_remove), |