March 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Remove warnings in ep93xx-i2s.c
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Remove warnings in ep93xx-i2s.c Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 3d4c086..042f4e9 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -313,10 +313,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } static int ep93xx_i2s_resume(struct snd_soc_dai *dai) @@ -324,10 +326,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai) struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); if (!dai->active) - return; + return 0; ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK); ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE); + + return 0; } #else #define ep93xx_i2s_suspend NULL

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[alsa-devel] [PATCH v2] intel8x0m: wait a bit before warm reset check
by Paul Bolle 07 Mar '11

07 Mar '11

At every resume a laptop I use prints this message (at KERN_ERR level): ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2] The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what seems to be happening is that the register involved indicated a warm reset for some time (as the ICH_AC97WARM bit was set) but by the time the warning is printed, and that same register is checked again, that bit is already cleared and only the ICH_AC97COLD bit is still set. It turns out a warm reset needs some time to settle, but it is currently checked right away. The test therefore fails the first time it is done and schedule_timeout_uninterruptible() will be called. Once we return from that jiffies is already (far) past end_time on this laptop, so we exit the loop, print a warning, and exit the function while the warm reset actually succeeded. One way to fix this is to call udelay() after writing to the register involved. A handful of tests suggests 500 usecs is a safe value. (This might punish the "finish cold reset" case, but on this laptop such a cold reset apparently never happens, so I can't say for sure.) While we're at it drop the extra single tick from end_time, as it looks rather silly. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- 0) v1 was called " intel8x0m: schedule timeout before warm reset check". 1) A udelay of 250 usecs always failed while 300 usecs always worked. So 500 usecs might be a bit cautious. sound/pci/intel8x0m.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 13cec1e..62fad24 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -894,7 +894,8 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) /* finish cold or do warm reset */ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; iputdword(chip, ICHREG(GLOB_CNT), cnt); - end_time = (jiffies + (HZ / 4)) + 1; + udelay(500); + end_time = jiffies + HZ / 4; do { if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0) goto __ok; -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Use snd_soc_dapm_context as required
by Dimitris Papastamos 07 Mar '11

07 Mar '11

Fix compilation issues by using the snd_soc_dapm_context pointer whenever it is required. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/tlv320aic32x4.c | 10 ++++++---- 1 files changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index ee82e38..c6c3520 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -340,13 +340,15 @@ static inline int aic32x4_get_divs(int mclk, int rate) static int aic32x4_add_widgets(struct snd_soc_codec *codec) { - snd_soc_dapm_new_controls(codec, aic32x4_dapm_widgets, + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, aic32x4_dapm_widgets, ARRAY_SIZE(aic32x4_dapm_widgets)); - snd_soc_dapm_add_routes(codec, aic32x4_dapm_routes, + snd_soc_dapm_add_routes(dapm, aic32x4_dapm_routes, ARRAY_SIZE(aic32x4_dapm_routes)); - snd_soc_dapm_new_widgets(codec); + snd_soc_dapm_new_widgets(dapm); return 0; } @@ -602,7 +604,7 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: break; } - codec->bias_level = level; + codec->dapm.bias_level = level; return 0; } -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Improve EP93xx I2S clocks management.
by Alexander Sverdlin 07 Mar '11

07 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Improve EP93xx I2S clocks management. Some freqs values are set not exact as they requested for MCLK and original code was not able to find divisors for SCLK and LRCLK. This code just picks up nearest value from 3 possible variants. This patch makes 44100 and 192000 rates working and fixes capture function. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 19 +++++++++---------- 1 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index 0458dd4..3d4c086 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai); unsigned word_len, div, sdiv, lrdiv; - int found = 0, err; + int err; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream, * the codec uses. */ div = clk_get_rate(info->mclk) / params_rate(params); - for (sdiv = 2; sdiv <= 4; sdiv += 2) - for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1) - if (sdiv * lrdiv == div) { - found = 1; - goto out; - } -out: - if (!found) - return -EINVAL; + sdiv = 4; + if (div > (256 + 512) / 2) { + lrdiv = 128; + } else { + lrdiv = 64; + if (div < (128 + 256) / 2) + sdiv = 2; + } err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv); if (err)

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[alsa-devel] hda : computer using EAPD as external amplifier controller
by Anisse Astier 06 Mar '11

06 Mar '11

Hi, I have this computer, with an integrated speaker. It's using an ALC662, see attached alsa-info output. By default, speaker doesn't work. Now, after playing a bit with hda-analyzer, I figured that output PIN 0x14 has EAPD enabled, which is the default way of doing things in auto configure mode. But disabling EAPD on this PIN makes speaker work. See second alsa-info attached. Is it necessary to write a full blown quirk just to modify this parameter? From what I've seen in patch_realtek, it seems we need to write the whole config for the device when adding a quirk: we can't rely on auto configuration and then put a small quirk on top of that. Am I right ? Regards, Anisse

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[alsa-devel] 2.6.37, snd-hda-intel: BUG: unable to handle kernel paging request at ffffc90011c08000
by Paul Menzel 06 Mar '11

06 Mar '11

Dear ALSA folks, is the following oops already known to you [1]? ** Kernel log: [ 41.265636] parport0: PC-style at 0x378, irq 7 [PCSPP,TRISTATE] [ 41.680357] HDA Intel 0000:20:01.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 [ 41.680444] HDA Intel 0000:20:01.0: setting latency timer to 64 [ 41.680462] BUG: unable to handle kernel paging request at ffffc90011c08000 [ 41.680617] IP: [<ffffffffa0578402>] azx_probe+0x3ad/0x86b [snd_hda_intel] [ 41.680728] PGD 13781a067 PUD 13781b067 PMD 1300ba067 PTE 800000fd00000173 [ 41.680956] Oops: 0009 [#1] SMP [ 41.681098] last sysfs file: /sys/module/snd_pcm/initstate [ 41.681159] CPU 0 [ 41.681203] Modules linked in: snd_hda_intel(+) snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_midi snd_rawmidi snd_seq_midi_event tpm_tis tpm snd_seq tpm_bios psmouse parport_pc snd_timer snd_seq_device parport processor evdev snd i2c_viapro thermal_sys amd64_edac_mod k8temp i2c_core soundcore shpchp pcspkr serio_raw asus_atk0110 pci_hotplug edac_core button snd_page_alloc edac_mce_amd ext3 jbd mbcache sha256_generic cryptd aes_x86_64 aes_generic cbc dm_crypt dm_mod raid1 md_mod usbhid hid sg sd_mod crc_t10dif sr_mod cdrom ata_generic uhci_hcd sata_via pata_via libata ehci_hcd usbcore scsi_mod via_rhine mii nls_base [last unloaded: scsi_wait_scan] [ 41.684180] [ 41.684180] Pid: 1153, comm: work_for_cpu Not tainted 2.6.37-1-amd64 #1 M2V-MX SE/System Product Name [ 41.684180] RIP: 0010:[<ffffffffa0578402>] [<ffffffffa0578402>] azx_probe+0x3ad/0x86b [snd_hda_intel] [ 41.684180] RSP: 0018:ffff88013153fe50 EFLAGS: 00010286 [ 41.684180] RAX: ffffc90011c08000 RBX: ffff88013029ec00 RCX: 0000000000000006 [ 41.684180] RDX: 0000000000000000 RSI: 0000000000000246 RDI: 0000000000000246 [ 41.684180] RBP: ffff88013341d000 R08: 0000000000000000 R09: 0000000000000040 [ 41.684180] R10: 0000000000000286 R11: 0000000000003731 R12: ffff88013029c400 [ 41.684180] R13: 0000000000000000 R14: 0000000000000000 R15: ffff88013341d090 [ 41.684180] FS: 0000000000000000(0000) GS:ffff8800bfc00000(0000) knlGS:00000000f7610ab0 [ 41.684180] CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b [ 41.684180] CR2: ffffc90011c08000 CR3: 0000000132f57000 CR4: 00000000000006f0 [ 41.684180] DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 [ 41.684180] DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 [ 41.684180] Process work_for_cpu (pid: 1153, threadinfo ffff88013153e000, task ffff8801303c86c0) [ 41.684180] Stack: [ 41.684180] 0000000000000005 ffffffff8123ad65 00000000000136c0 ffff88013029c400 [ 41.684180] ffff8801303c8998 ffff88013341d000 ffff88013341d090 ffff8801322d9dc8 [ 41.684180] ffff88013341d208 0000000000000000 0000000000000000 ffffffff811ad232 [ 41.684180] Call Trace: [ 41.684180] [<ffffffff8123ad65>] ? __pm_runtime_set_status+0x162/0x186 [ 41.684180] [<ffffffff811ad232>] ? local_pci_probe+0x49/0x92 [ 41.684180] [<ffffffff8105afc5>] ? do_work_for_cpu+0x0/0x1b [ 41.684180] [<ffffffff8105afc5>] ? do_work_for_cpu+0x0/0x1b [ 41.684180] [<ffffffff8105afd0>] ? do_work_for_cpu+0xb/0x1b [ 41.684180] [<ffffffff8105fd3f>] ? kthread+0x7a/0x82 [ 41.684180] [<ffffffff8100a824>] ? kernel_thread_helper+0x4/0x10 [ 41.684180] [<ffffffff8105fcc5>] ? kthread+0x0/0x82 [ 41.684180] [<ffffffff8100a820>] ? kernel_thread_helper+0x0/0x10 [ 41.684180] Code: f4 01 00 00 ef 31 f6 48 89 df e8 29 dd ff ff 85 c0 0f 88 2b 03 00 00 48 89 ef e8 b4 39 c3 e0 8b 7b 40 e8 fc 9d b1 e0 48 8b 43 38 <66> 8b 10 66 89 14 24 8b 43 14 83 e8 03 83 f8 01 77 32 31 d2 be [ 41.684180] RIP [<ffffffffa0578402>] azx_probe+0x3ad/0x86b [snd_hda_intel] [ 41.684180] RSP <ffff88013153fe50> [ 41.684180] CR2: ffffc90011c08000 [ 41.684180] ---[ end trace 8d1f3ebc136437fd ]--- [ 42.127750] input: ImPS/2 Logitech Wheel Mouse as /devices/platform/i8042/serio1/input/input5 [ 63.455908] EXT3-fs (dm-1): using internal journal Searching the Web a comment to a report [2] says that such problems are hard to debug. Can you give me any hints? I will try to provide the output of `alsa-info.sh` in some hours. Thanks, Paul [1] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=613979 [2] https://bugzilla.kernel.org/show_bug.cgi?id=9305

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[alsa-devel] [PATCH] ASoC: Enable 192kHz sample rate for EP92xx.
by Alexander Sverdlin 05 Mar '11

05 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Enable 192kHz sample rate for EP92xx. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/ep93xx-i2s.c | 4 ++-- sound/soc/ep93xx/ep93xx-pcm.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c index e1a58d3..87dc1ee 100644 --- a/sound/soc/ep93xx/ep93xx-i2s.c +++ b/sound/soc/ep93xx/ep93xx-i2s.c @@ -351,13 +351,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = EP93XX_I2S_FORMATS, }, .ops = &ep93xx_i2s_dai_ops, diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 0667077..a456e49 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = { SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .rate_min = SNDRV_PCM_RATE_8000, - .rate_max = SNDRV_PCM_RATE_96000, + .rate_max = SNDRV_PCM_RATE_192000, .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |

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[alsa-devel] [PATCH] add new Fermi 5xx codec IDs to snd-hda
by Richard Samson 05 Mar '11

05 Mar '11

Hello, It's my first patch/post here, i'm not sure if this list is the right place to suggest this patch. To follow up a previous commit on snd-hda : http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=commitdiff;… I added a new entry in patch_hdmi.c to get sound working on my own 560 Ti card : --- patch_hdmi.c 2011-02-15 22:29:51.000000000 +0100 +++ patch_hdmi_GTX-560Ti.c 2011-02-27 23:52:56.984613414 +0100 @@ -1634,6 +1634,7 @@ { .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, +{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi_8ch_89 }, @@ -1676,6 +1677,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0012"); MODULE_ALIAS("snd-hda-codec-id:10de0013"); MODULE_ALIAS("snd-hda-codec-id:10de0014"); +MODULE_ALIAS("snd-hda-codec-id:10de0016"); MODULE_ALIAS("snd-hda-codec-id:10de0018"); MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); I think other 5xx cards need new value also. Regards, --- Richard Samson samson.richard(a)gmail.com

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[alsa-devel] [PATCH] ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.
by Alexander Sverdlin 05 Mar '11

05 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx machine driver. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/ep93xx/edb93xx.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c index b270085..d3aa151 100644 --- a/sound/soc/ep93xx/edb93xx.c +++ b/sound/soc/ep93xx/edb93xx.c @@ -41,17 +41,17 @@ static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; + unsigned int mclk_rate; unsigned int rate = params_rate(params); + /* - * We set LRCLK equal to `rate' and SCLK = LRCLK * 64, - * because our sample size is 32 bit * 2 channels. - * I2S standard permits us to transmit more bits than - * the codec uses. - * MCLK = SCLK * 4 is the best recommended value, - * but we have to fall back to ratio 2 for higher - * sample rates. + * According to CS4271 datasheet we use MCLK/LRCK=256 for + * rates below 50kHz and 128 for higher sample rates */ - unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2); + if (rate < 50000) + mclk_rate = rate * 64 * 4; + else + mclk_rate = rate * 64 * 2; err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |

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[alsa-devel] [PATCH] ASoC: Manage mode and rate bits correctly for CS4271 CODEC.
by Alexander Sverdlin 05 Mar '11

05 Mar '11

From: Alexander Sverdlin <subaparts(a)yandex.ru> Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC. This is done to make capture work properly. Signed-off-by: Alexander Sverdlin <subaparts(a)yandex.ru> --- sound/soc/codecs/cs4271.c | 83 ++++++++++++++++++++++++++++---------------- 1 files changed, 53 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e5cc4da..c71ad08 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -168,27 +168,6 @@ struct cs4271_private { int gpio_disable; }; -struct cs4271_clk_cfg { - unsigned int ratio; /* MCLK / sample rate */ - u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ - u8 mclk_master; /* ratio bit mask for Master mode */ - u8 mclk_slave; /* ratio bit mask for Slave mode */ -}; - -static struct cs4271_clk_cfg cs4271_clk_tab[] = { - {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1}, - {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1}, - {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1}, - {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}, - {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3} -}; - -#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) - /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -297,6 +276,45 @@ static int cs4271_put_deemph(struct snd_kcontrol *kcontrol, return cs4271_set_deemph(codec); } +struct cs4271_clk_cfg { + bool master; /* codec mode */ + u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */ + unsigned short ratio; /* MCLK / sample rate */ + u8 ratio_mask; /* ratio bit mask for Master mode */ +}; + +static struct cs4271_clk_cfg cs4271_clk_tab[] = { + {1, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_3}, + {1, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {1, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_15}, + {1, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_2}, + {1, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_3}, + {0, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_2X, 512, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_1}, + {0, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_2}, + {0, CS4271_MODE1_MODE_4X, 256, CS4271_MODE1_DIV_2}, +}; + +#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab) + static int cs4271_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -308,23 +326,28 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, unsigned int ratio, val; cs4271->rate = params_rate(params); + + /* Configure DAC */ + if (cs4271->rate < 50000) + val = CS4271_MODE1_MODE_1X; + else if (cs4271->rate < 100000) + val = CS4271_MODE1_MODE_2X; + else + val = CS4271_MODE1_MODE_4X; + ratio = cs4271->mclk / cs4271->rate; for (i = 0; i < CS4171_NR_RATIOS; i++) - if (cs4271_clk_tab[i].ratio == ratio) + if ((cs4271_clk_tab[i].master == cs4271->master) && + (cs4271_clk_tab[i].speed_mode == val) && + (cs4271_clk_tab[i].ratio == ratio)) break; - if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) { + if (i == CS4171_NR_RATIOS) { dev_err(codec->dev, "Invalid sample rate\n"); return -EINVAL; } - /* Configure DAC */ - val = cs4271_clk_tab[i].speed_mode; - - if (cs4271->master) - val |= cs4271_clk_tab[i].mclk_master; - else - val |= cs4271_clk_tab[i].mclk_slave; + val |= cs4271_clk_tab[i].ratio_mask; ret = snd_soc_update_bits(codec, CS4271_MODE1, CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);

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