Merge tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It's been busy summer weeks and hence lots of changes, partly for a
  few new drivers and partly for a wide range of fixes.

  Here are highlights:

  ALSA Core:
   - Fix rawmidi buffer management, code cleanup / refactoring
   - Fix the SG-buffer page handling with incorrect fallback size
   - Fix the stall at virmidi trigger callback with a large buffer; also
     offloading and code-refactoring along with it
   - Various ALSA sequencer code cleanups

  ASoC:
   - Deploy the standard snd_pcm_stop_xrun() helper in several drivers
   - Support for providing name prefixes to generic component nodes
   - Quite a few fixes for DPCM as it gains a bit wider use and more
     robust testing
   - Generalization of the DIO2125 support to a simple amplifier driver
   - Accessory detection support for the audio graph card
   - DT support for PXA AC'97 devices
   - Quirks for a number of new x86 systems
   - Support for AM Logic Meson, Everest ES7154, Intel systems with
     RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707

  HD-audio:
   - Code refactoring in HD-audio ext codec codes to drop own classes;
     preliminary works for the upcoming legacy codec support
   - Generalized DRM audio component for the upcoming radeon / amdgpu
     support
   - Unification of mic mute-LED and GPIO support for various codecs
   - Further improvement of CA0132 codec support including Recon3D
   - Proper vga_switcheroo handling for AMD i-GPU
   - Update of model list in documentation
   - Fixups for another HP Spectre x360, Conexant codecs, power-save
     blacklist update

  USB-audio:
   - Fix the invalid sample rate setup with external clock
   - Support of UAC3 selector units and processing units
   - Basic UAC3 power-domain support
   - Support for Encore mDSD and Thesycon-based DSD devices
   - Preparation for future complete callback changes

  Firewire:
   - Add support for MOTU Traveler

  Misc:
   - The endianess notation fixes in various drivers
   - Add fall-through comment in lots of drivers
   - Various sparse warning fixes, e.g. about PCM format types"

* tag 'sound-4.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (529 commits)
  ASoC: adav80x: mark expected switch fall-through
  ASoC: da7219: Add delays to capture path to remove DC offset noise
  ALSA: usb-audio: Mark expected switch fall-through
  ALSA: mixart: Mark expected switch fall-through
  ALSA: opl3: Mark expected switch fall-through
  ALSA: hda/ca0132 - Add exit commands for Recon3D
  ALSA: hda/ca0132 - Change mixer controls for Recon3D
  ALSA: hda/ca0132 - Add Recon3D input and output select commands
  ALSA: hda/ca0132 - Add DSP setup defaults for Recon3D
  ALSA: hda/ca0132 - Add Recon3D startup functions and setup
  ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio
  ALSA: hda/ca0132 - Add Recon3D pincfg
  ALSA: hda/ca0132 - Add quirk ID and enum for Recon3D
  ALSA: hda/ca0132 - Add alt_functions unsolicited response
  ALSA: hda/ca0132 - Clean up ca0132_init function.
  ALSA: hda/ca0132 - Create mmio gpio function to make code clearer
  ASoC: wm_adsp: Make DSP name configurable by codec driver
  ASoC: wm_adsp: Declare firmware controls from codec driver
  ASoC: max98373: Added software reset register to readable registers
  ASoC: wm_adsp: Correct DSP pointer for preloader control
  ...

1 files changed, 2 insertions, 104 deletions

diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index 2fc012b06c43..935a248e5bf6 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	unsigned int acps, acds, width;
-	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+	unsigned int width;
 	int ret = 0;
 
-	width = snd_pcm_format_physical_width(params_format(params));
-
-	/*
-	 * rate = SSPSCLK / (2 * width(16 or 32))
-	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
-	 */
-	switch (params_rate(params)) {
-	case 8000:
-		/* off by a factor of 2: bug in the PXA27x audio clock? */
-		acps = 32842000;
-		switch (width) {
-		case 16:
-			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_16;
-			break;
-		default: /* 32 */
-			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_8;
-		}
-		break;
-	case 11025:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_4;
-			break;
-		default: /* 32 */
-			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-		}
-		break;
-	case 22050:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 44100:
-		acps = 5622000;
-		switch (width) {
-		case 16:
-			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 48000:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			break;
-		default: /* 32 */
-			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-		}
-		break;
-	case 96000:
-	default:
-		acps = 12235000;
-		switch (width) {
-		case 16:
-			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_1;
-			break;
-		default: /* 32 */
-			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
-			acds = PXA_SSP_CLK_AUDIO_DIV_2;
-			div4 = PXA_SSP_CLK_SCDB_1;
-			break;
-		}
-		break;
-	}
-
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
 			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
@@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
+	width = snd_pcm_format_physical_width(params_format(params));
 	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
 	if (ret < 0)
 		return ret;
@@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	/* set the SSP audio system clock ACDS divider */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_ACDS, acds);
-	if (ret < 0)
-		return ret;
-
-	/* set the SSP audio system clock SCDB divider4 */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai,
-			PXA_SSP_AUDIO_DIV_SCDB, div4);
-	if (ret < 0)
-		return ret;
-
-	/* set SSP audio pll clock */
-	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
-	if (ret < 0)
-		return ret;
-
 	return 0;
 }