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Commit 01d5500f authored by Takashi Iwai's avatar Takashi Iwai
Browse files

Merge tag 'asoc-v3.17-rc1' of... 

Merge tag 'asoc-v3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.17

Nothing too exciting here, a bunch of driver fixes that came along since
the initial pull request but none that really stand our and a warning
fix in the core.
parents 61074c1a 7c063ede

MAINTAINERS

+3 −0
Original line number Diff line number Diff line
@@ -3794,10 +3794,13 @@ F: drivers/tty/serial/ucc_uart.c 


FREESCALE SOC SOUND DRIVERS

M:	Timur Tabi <timur@tabi.org>

M:	Nicolin Chen <nicoleotsuka@gmail.com>

M:	Xiubo Li <Li.Xiubo@freescale.com>

L:	alsa-devel@alsa-project.org (moderated for non-subscribers)

L:	linuxppc-dev@lists.ozlabs.org

S:	Maintained

F:	sound/soc/fsl/fsl*

F:	sound/soc/fsl/imx*

F:	sound/soc/fsl/mpc8610_hpcd.c



FREEVXFS FILESYSTEM

sound/soc/codecs/arizona.c

+4 −2
Original line number Diff line number Diff line
@@ -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);

sound/soc/codecs/pcm512x.c

+2 −2
Original line number Diff line number Diff line
@@ -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),

sound/soc/davinci/davinci-mcasp.c

+11 −3
Original line number Diff line number Diff line
@@ -403,7 +403,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, 
	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);
@@ -420,6 +421,7 @@ 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);

		if (explicit)

			mcasp->bclk_div = div;

		break;
@@ -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)

{
@@ -738,7 +746,7 @@ 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,

sound/soc/fsl/Kconfig

+0 −1
Original line number Diff line number Diff line
@@ -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.