From c5489f9fc053c744c609f34b32efca395cc2fdad Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk Date: Fri, 2 Feb 2018 13:10:29 +0100 Subject: sgtl5000: change digital_mute policy Current implementation mute codec in global way (DAC block). That means when user routes sound not from I2S but from AUX source (LINE_IN) it also will be muted by alsa core. This should not happen. Signed-off-by: Michal Oleszczyk Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index e1ab5537d27a..c445a0794a27 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -529,10 +529,15 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; - u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT; + u16 i2s_pwr = SGTL5000_I2S_IN_POWERUP; - snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, - adcdac_ctrl, mute ? adcdac_ctrl : 0); + /* + * During 'digital mute' do not mute DAC + * because LINE_IN would be muted aswell. We want to mute + * only I2S block - this can be done by powering it off + */ + snd_soc_update_bits(codec, SGTL5000_CHIP_DIG_POWER, + i2s_pwr, mute ? 0 : i2s_pwr); return 0; } @@ -1237,6 +1242,10 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) */ snd_soc_write(codec, SGTL5000_DAP_CTRL, 0); + /* Unmute DAC after start */ + snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL, + SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT, 0); + return 0; err: -- cgit v1.2.1 From a8992973edbb2555e956b90f6fe97c4bc14d761d Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 16 Feb 2018 11:58:54 -0200 Subject: ASoC: sgtl5000: Fix suspend/resume Commit 8419caa72702 ("ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF") causes the sgtl5000 to fail after a suspend/resume sequence: Playing WAVE '/media/a2002011001-e02.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:2051: write error: Input/output error The problem is caused by the fact that the aforementioned commit dropped the cache handling, so re-introduce the register map resync to fix the problem. Suggested-by: Mark Brown Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown Cc: --- sound/soc/codecs/sgtl5000.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c445a0794a27..c5c76ab8ccf1 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -876,15 +876,26 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream, static int sgtl5000_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct sgtl5000_priv *sgtl = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: + regcache_cache_only(sgtl->regmap, false); + ret = regcache_sync(sgtl->regmap); + if (ret) { + regcache_cache_only(sgtl->regmap, true); + return ret; + } + snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, SGTL5000_REFTOP_POWERUP); break; case SND_SOC_BIAS_OFF: + regcache_cache_only(sgtl->regmap, true); snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_REFTOP_POWERUP, 0); break; -- cgit v1.2.1 From 2d30e9494f1ea320aaaad0cff9ddd92c87eac355 Mon Sep 17 00:00:00 2001 From: Hans de Goede Date: Sun, 18 Feb 2018 23:01:44 +0100 Subject: ASoC: rt5651: Fix regcache sync errors on resume The ALC5651 does not like multi-write accesses, avoid them. This fixes: rt5651 i2c-10EC5651:00: Unable to sync registers 0x27-0x28. -121 Errors on resume (and all registers after the registers in the error not being synced). Signed-off-by: Hans de Goede Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/rt5651.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 831b297978a4..45a73049cf64 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1722,6 +1722,7 @@ static const struct regmap_config rt5651_regmap = { .num_reg_defaults = ARRAY_SIZE(rt5651_reg), .ranges = rt5651_ranges, .num_ranges = ARRAY_SIZE(rt5651_ranges), + .use_single_rw = true, }; #if defined(CONFIG_OF) -- cgit v1.2.1 From 5e558f8afaec8957932b1dbe5aeff800f9fc6957 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 20 Feb 2018 16:19:05 +0200 Subject: ASoC: hdmi-codec: Fix module unloading caused kernel crash The hcp->chmap_info must not be freed up in the hdmi_codec_remove() function as it leads to kernel crash due ALSA core's pcm_chmap_ctl_private_free() is trying to free it up again when the card destroyed via snd_card_free. Commit cd6111b26280a ("ASoC: hdmi-codec: add channel mapping control") should not have added the kfree(hcp->chmap_info); to the hdmi_codec_remove function. Signed-off-by: Peter Ujfalusi Reviewed-by: Jyri Sarha Tested-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi-codec.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 5672e516bec3..c1830ccd3bb8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -798,12 +798,7 @@ static int hdmi_codec_probe(struct platform_device *pdev) static int hdmi_codec_remove(struct platform_device *pdev) { - struct device *dev = &pdev->dev; - struct hdmi_codec_priv *hcp; - - hcp = dev_get_drvdata(dev); - kfree(hcp->chmap_info); - snd_soc_unregister_codec(dev); + snd_soc_unregister_codec(&pdev->dev); return 0; } -- cgit v1.2.1 From 5a3386790a172cf738194e1574f631cd43c6140a Mon Sep 17 00:00:00 2001 From: Yong Deng Date: Mon, 26 Feb 2018 10:43:52 +0800 Subject: ASoC: sun4i-i2s: Fix RX slot number of SUN8I I2S's RX slot number of SUN8I should be shifted 4 bit to left. Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3") Signed-off-by: Yong Deng Reviewed-by: Chen-Yu Tsai Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index dca1143c1150..a4aa931ebfae 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -104,7 +104,7 @@ #define SUN8I_I2S_CHAN_CFG_REG 0x30 #define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK GENMASK(6, 4) -#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) (chan - 1) +#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) ((chan - 1) << 4) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK GENMASK(2, 0) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan) (chan - 1) -- cgit v1.2.1 From d7789f5bcdb298c4a302db471b1b20f74a20de95 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Wed, 28 Feb 2018 10:31:10 +0000 Subject: ASoC: wm_adsp: For TLV controls only register TLV get/set Normal 512-byte get/set of a TLV isn't supported but we were registering the normal get/set anyway and relying on omitting the SNDRV_CTL_ELEM_ACCESS_[READ|WRITE] flags to prevent them being called. Trouble is if this gets broken in the core ALSA code - as it has been since at least 4.14 - the standard get/set can be called unexpectedly and corrupt memory. There's no point providing functions that won't be called and it's a trivial change. The benefit is that if the ALSA core gets broken again we get a big fat immediate NULL dereference instead of a memory corruption timebomb. Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 66e32f5d2917..989d093abda7 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1204,12 +1204,14 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->put = wm_coeff_put_acked; break; default: - kcontrol->get = wm_coeff_get; - kcontrol->put = wm_coeff_put; - - ctl->bytes_ext.max = ctl->len; - ctl->bytes_ext.get = wm_coeff_tlv_get; - ctl->bytes_ext.put = wm_coeff_tlv_put; + if (kcontrol->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + } else { + kcontrol->get = wm_coeff_get; + kcontrol->put = wm_coeff_put; + } break; } -- cgit v1.2.1 From a37d48e32303d535bdfd554c57952ce31f428b3a Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 9 Mar 2018 21:13:02 +0530 Subject: ASoC: amd: 16bit resolution support for i2s sp instance Moved 16bit resolution condition check for stoney platform to acp_hw_params.Depending upon substream required register value need to be programmed rather than enabling 16bit resolution support all time in acp init. Signed-off-by: Vijendar Mukunda Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 16 +++++++++------- sound/soc/amd/acp.h | 2 ++ 2 files changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index c33a512283a4..9fb356db3ab2 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -579,13 +579,6 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) for (bank = 1; bank < 48; bank++) acp_set_sram_bank_state(acp_mmio, bank, false); } - - /* Stoney supports 16bit resolution */ - if (asic_type == CHIP_STONEY) { - val = acp_reg_read(acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); - val |= 0x03; - acp_reg_write(val, acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); - } return 0; } @@ -774,6 +767,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, { int status; uint64_t size; + u32 val = 0; struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; @@ -786,6 +780,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; + if (adata->asic_type == CHIP_STONEY) { + val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + else + val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + } size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index ecb458935d1e..9293f179f272 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -70,6 +70,8 @@ #define CAPTURE_END_DMA_DESCR_CH15 7 #define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 +#define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 +#define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0, -- cgit v1.2.1