Merge tag 'sound-5.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been some significant changes in the core side, both for
  ALSA and ASoC, while lots of development have been seen in SOF, as
  well as many small fixes/improvements for ASoC codecs and platforms.
  Below is a highlight in this cycle:

  Core:
   - The unification of PCM vmalloc buffer allocation helpers into the
     standard API
   - Clean up of the default PCM mmap handling for vmalloc & SG-buffer
   - Fix potential races at ALSA timer open
   - A few new PCM API extensions; just preliminary core changes, the
     actual changes in drivers will be merged in 5.6
   - Continued ASoC componentization works; now almost everything is a
     common ASoC component object. A lot of refactoring and
     simplification have been done along with it.

  ASoC:
   - Many fixes to the Sound Open Firmware (SOF) code
   - Wake on voice support for Chromebooks
   - SPI support and trigger word detection for RT5677
   - New drivers for Analog Devices ADAU7118, Intel Cannonlake systems
     with RT1011 and RT5682, Texas Instruments TAS2562 and TAS2770

  HD-audio:
   - Improved Intel DSP configuration / probe code for SOF
   - Plumbing the legacy HD-audio driver with Intel SOF HDMI
   - DP-MST support for Nvidia HDMI codecs
   - Realtek quirks cleanups and new additions as usual

  Others:
   - Lots of refactoring and cleanups for FireWire; period-size sharing,
     h/w IRQ interval configuration, clock recovery improvements, etc
   - USB-audio: Scarlett mixer quirks
   - Cleanups of PCM calls in various drivers (including media and USB)
     to adapt the core API changes"

* tag 'sound-5.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (497 commits)
  ALSA: usb-audio: Fix Focusrite Scarlett 6i6 gen1 - input handling
  ALSA: hda/realtek - Enable internal speaker of ASUS UX431FLC
  ALSA: aloop: Fix dependency on timer API
  ASoC: DMI long name - avoid to add board name if matches with product name
  ASoC: improve the DMI long card code in asoc-core
  ASoC: rsnd: fix DALIGN register for SSIU
  ALSA: aloop: Avoid unexpected timer event callback tasklets
  ALSA: aloop: Remove redundant locking in timer open function
  ASoC: component: Add sync_stop PCM ops
  ASoC: pcm: Make ioctl ops optional
  ALSA: hda/hdmi - Clear codec->relaxed_resume flag at unbinding
  ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs
  ALSA: cs4236: fix error return comparison of an unsigned integer
  ALSA: usb-audio: Fix NULL dereference at parsing BADD
  ALSA: usb-audio: Fix Scarlett 6i6 Gen 2 port data
  ALSA: hda/realtek - Enable the headset-mic on a Xiaomi's laptop
  ALSA: hda/realtek - Move some alc236 pintbls to fallback table
  ALSA: hda/realtek - Move some alc256 pintbls to fallback table
  ALSA: docs: Update about the new PCM sync_stop ops
  ALSA: pcm: Add card sync_irq field
  ...

1 files changed, 55 insertions, 0 deletions

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index e8141a33a55e..36e0eab13a98 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -38,6 +38,7 @@
 #define ASM_SESSION_CMD_RUN_V2			0x00010DAA
 #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2	0x00010DA5
 #define ASM_MEDIA_FMT_MP3			0x00010BE9
+#define ASM_MEDIA_FMT_FLAC			0x00010C16
 #define ASM_DATA_CMD_WRITE_V2			0x00010DAB
 #define ASM_DATA_CMD_READ_V2			0x00010DAC
 #define ASM_SESSION_CMD_SUSPEND			0x00010DEC
@@ -89,6 +90,20 @@ struct asm_multi_channel_pcm_fmt_blk_v2 {
 	u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
 } __packed;
 
+struct asm_flac_fmt_blk_v2 {
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	u16 is_stream_info_present;
+	u16 num_channels;
+	u16 min_blk_size;
+	u16 max_blk_size;
+	u16 md5_sum[8];
+	u32 sample_rate;
+	u32 min_frame_size;
+	u32 max_frame_size;
+	u16 sample_size;
+	u16 reserved;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	u32                  param_size;
@@ -876,6 +891,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
 	case FORMAT_LINEAR_PCM:
 		open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
 		break;
+	case SND_AUDIOCODEC_FLAC:
+		open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
+		break;
 	default:
 		dev_err(ac->dev, "Invalid format 0x%x\n", format);
 		rc = -EINVAL;
@@ -1021,6 +1039,42 @@ err:
 }
 EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
 
+
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+					 struct q6asm_flac_cfg *cfg)
+{
+	struct asm_flac_fmt_blk_v2 *fmt;
+	struct apr_pkt *pkt;
+	void *p;
+	int rc, pkt_size;
+
+	pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+	p = kzalloc(pkt_size, GFP_KERNEL);
+	if (!p)
+		return -ENOMEM;
+
+	pkt = p;
+	fmt = p + APR_HDR_SIZE;
+
+	q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+	pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+	fmt->is_stream_info_present = cfg->stream_info_present;
+	fmt->num_channels = cfg->ch_cfg;
+	fmt->min_blk_size = cfg->min_blk_size;
+	fmt->max_blk_size = cfg->max_blk_size;
+	fmt->sample_rate = cfg->sample_rate;
+	fmt->min_frame_size = cfg->min_frame_size;
+	fmt->max_frame_size = cfg->max_frame_size;
+	fmt->sample_size = cfg->sample_size;
+
+	rc = q6asm_ac_send_cmd_sync(ac, pkt);
+	kfree(pkt);
+
+	return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
 /**
  * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
  *
@@ -1075,6 +1129,7 @@ err:
 }
 EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
 
+
 /**
  * q6asm_read() - read data of period size from audio client
  *