From 335e3b8687fa6832bd6a033f2c705786e4bfb92c Mon Sep 17 00:00:00 2001 From: Vitaliy Kulikov Date: Fri, 22 Oct 2010 18:38:31 -0500 Subject: ALSA: hda - Change BTL amp level on some HP notebooks Some HP laptops have lower amplifier levels for speakers in comparison with headphone outputs. This patch changes the BTL amp level for these machines to balance both the speaker and headphone output levels. Signed-off-by: Vitaliy Kulikov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 78 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 82ebeb9544fe..93fa59cc60ef 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5326,6 +5326,82 @@ again: return 0; } +static int stac92hd83xxx_set_system_btl_amp(struct hda_codec *codec) +{ + if (codec->vendor_id != 0x111d7605 && + codec->vendor_id != 0x111d76d1) + return 0; + + switch (codec->subsystem_id) { + case 0x103c1618: + case 0x103c1619: + case 0x103c161a: + case 0x103c161b: + case 0x103c161c: + case 0x103c161d: + case 0x103c161e: + case 0x103c161f: + case 0x103c1620: + case 0x103c1621: + case 0x103c1622: + case 0x103c1623: + + case 0x103c162a: + case 0x103c162b: + + case 0x103c1630: + case 0x103c1631: + + case 0x103c1633: + + case 0x103c1635: + + case 0x103c164f: + + case 0x103c1676: + case 0x103c1677: + case 0x103c1678: + case 0x103c1679: + case 0x103c167a: + case 0x103c167b: + case 0x103c167c: + case 0x103c167d: + case 0x103c167e: + case 0x103c167f: + case 0x103c1680: + case 0x103c1681: + case 0x103c1682: + case 0x103c1683: + case 0x103c1684: + case 0x103c1685: + case 0x103c1686: + case 0x103c1687: + case 0x103c1688: + case 0x103c1689: + case 0x103c168a: + case 0x103c168b: + case 0x103c168c: + case 0x103c168d: + case 0x103c168e: + case 0x103c168f: + case 0x103c1690: + case 0x103c1691: + case 0x103c1692: + + case 0x103c3587: + case 0x103c3588: + case 0x103c3589: + case 0x103c358a: + + case 0x103c3667: + case 0x103c3668: + /* set BTL amp level to 13.43dB for louder speaker output */ + return snd_hda_codec_write_cache(codec, codec->afg, 0, + 0x7F4, 0x14); + } + return 0; +} + static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5452,6 +5528,8 @@ again: AC_VERB_SET_CONNECT_SEL, num_dacs); } + stac92hd83xxx_set_system_btl_amp(codec); + codec->proc_widget_hook = stac92hd_proc_hook; return 0; -- cgit v1.2.3 From 3342b9680fbd6535f65148f5e95e581bc75be9cc Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Tue, 26 Oct 2010 12:25:38 +0200 Subject: sound/oss/sb_ess.c: delete double assignment Delete successive assignments to the same location. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ expression i; @@ *i = ...; i = ...; // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/oss/sb_ess.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 51a3d381a59e..9890cf2066ff 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -1721,7 +1721,6 @@ printk (KERN_INFO "FKS: es_rec_set_recmask mask = %x\n", mask); left = value & 0x000000ff; right = (value & 0x0000ff00) >> 8; } else { /* Turn it off (3) */ - left = 0; left = 0; right = 0; } -- cgit v1.2.3 From e94be5f3628565d0968d668341d9dd468700516d Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 26 Oct 2010 20:07:43 +0100 Subject: ASoC: fsl - fix build error in pcm030-audio-fabric.c Fix build error:- sound/soc/fsl/pcm030-audio-fabric.c:27:33: fatal error: sound/soc-of-simple.h: No such file or directory Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/soc/fsl/pcm030-audio-fabric.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index fe15bb26e484..25f27ec1dd6e 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -24,7 +24,6 @@ #include #include #include -#include #include "mpc5200_dma.h" #include "mpc5200_psc_ac97.h" @@ -49,7 +48,7 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", .platform_name = "mpc5200-pcm-audio", - ..codec_name = "wm9712-codec", + .codec_name = "wm9712-codec", }, }; -- cgit v1.2.3 From 836f5394061830a9d1ece4aafc437c098774522f Mon Sep 17 00:00:00 2001 From: Arnaud Lacombe Date: Tue, 26 Oct 2010 16:04:34 -0400 Subject: ASoC: sound/ad73311: add missing __devexit marker This fixes the following warning: sound/soc/codecs/ad73311.c:50:12: warning: 'ad73311_remove' defined but not used Signed-off-by: Arnaud Lacombe Signed-off-by: Takashi Iwai --- sound/soc/codecs/ad73311.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index c53955fe17b6..de799cd1ba72 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -47,7 +47,7 @@ static int ad73311_probe(struct platform_device *pdev) &soc_codec_dev_ad73311, &ad73311_dai, 1); } -static int ad73311_remove(struct platform_device *pdev) +static int __devexit ad73311_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; -- cgit v1.2.3 From f3607aef0d3370ae9edbfecfc7182233397cb0aa Mon Sep 17 00:00:00 2001 From: Arnaud Lacombe Date: Tue, 26 Oct 2010 16:04:35 -0400 Subject: ASoC: sound/max98088: add missing __devexit marker This fixes the following warning: sound/soc/codecs/max98088.c:2054:12: warning: 'max98088_i2c_remove' defined but not used Signed-off-by: Arnaud Lacombe Signed-off-by: Takashi Iwai --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index e7a40d16df90..bc22ee93a75d 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -2051,7 +2051,7 @@ static int max98088_i2c_probe(struct i2c_client *i2c, return ret; } -static int max98088_i2c_remove(struct i2c_client *client) +static int __devexit max98088_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); kfree(i2c_get_clientdata(client)); -- cgit v1.2.3 From 0d040df9984c8fcb6a777a8f6d5dc513eaefd2de Mon Sep 17 00:00:00 2001 From: Arnaud Lacombe Date: Tue, 26 Oct 2010 16:04:36 -0400 Subject: ASoC: sound/wm9090: add missing __devexit marker This fixes the following warning: sound/soc/codecs/wm9090.c:668:12: warning: 'wm9090_i2c_remove' defined but not used Signed-off-by: Arnaud Lacombe Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm9090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 7a1825418ee4..99c046ba46bb 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -665,7 +665,7 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, return ret; } -static int wm9090_i2c_remove(struct i2c_client *i2c) +static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); -- cgit v1.2.3 From 89e1e66d6be8a520cdcd26043cda2cc870a34015 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 26 Oct 2010 17:14:41 +0200 Subject: ALSA: usb-audio: automatically detect feedback format There are two USB Audio Class specifications (v1 and v2), but neither of them clearly defines the feedback format for high-speed UAC v1 devices. Add to this whatever the Creative and M-Audio firmware writers have been smoking, and it becomes impossible to predict the exact feedback format used by a particular device. Therefore, automatically detect the feedback format by looking at the magnitude of the first received feedback value. Also, this allows us to get rid of some special cases for E-Mu devices. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/card.h | 2 + sound/usb/pcm.c | 2 + sound/usb/proc.c | 5 ++ sound/usb/urb.c | 170 +++++++++++++++++++------------------------------------ 4 files changed, 67 insertions(+), 112 deletions(-) (limited to 'sound') diff --git a/sound/usb/card.h b/sound/usb/card.h index 1febf2f23754..ae4251d5abf7 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -62,12 +62,14 @@ struct snd_usb_substream { unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ + int freqshift; /* how much to shift the feedback value to get Q16.16 */ unsigned int freqmax; /* maximum sampling rate, used for buffer management */ unsigned int phase; /* phase accumulator */ unsigned int maxpacksize; /* max packet size in bytes */ unsigned int maxframesize; /* max packet size in frames */ unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ + unsigned int syncmaxsize; /* sync endpoint packet size */ unsigned int fill_max: 1; /* fill max packet size always */ unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f49756c1b837..cff3a3c465d7 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -237,6 +237,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->datainterval = fmt->datainterval; subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; + subs->syncmaxsize = 0; subs->fill_max = 0; /* we need a sync pipe in async OUT or adaptive IN mode */ @@ -283,6 +284,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->syncinterval = get_endpoint(alts, 1)->bInterval - 1; else subs->syncinterval = 3; + subs->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); } /* always fill max packet size */ diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 3c650ab3c91d..961c9a250686 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -132,6 +132,11 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn ? get_full_speed_hz(subs->freqm) : get_high_speed_hz(subs->freqm), subs->freqm >> 16, subs->freqm & 0xffff); + if (subs->freqshift != INT_MIN) + snd_iprintf(buffer, " Feedback Format = %d.%d\n", + (subs->syncmaxsize > 3 ? 32 : 24) + - (16 - subs->freqshift), + 16 - subs->freqshift); } else { snd_iprintf(buffer, " Status: Stop\n"); } diff --git a/sound/usb/urb.c b/sound/usb/urb.c index 8deeaad10f10..e184349aee83 100644 --- a/sound/usb/urb.c +++ b/sound/usb/urb.c @@ -225,6 +225,7 @@ int snd_usb_init_substream_urbs(struct snd_usb_substream *subs, else subs->freqn = get_usb_high_speed_rate(rate); subs->freqm = subs->freqn; + subs->freqshift = INT_MIN; /* calculate max. frequency */ if (subs->maxpacksize) { /* whatever fits into a max. size packet */ @@ -513,11 +514,10 @@ static int retire_paused_capture_urb(struct snd_usb_substream *subs, /* - * prepare urb for full speed playback sync pipe + * prepare urb for playback sync pipe * * set up the offset and length to receive the current frequency. */ - static int prepare_playback_sync_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) @@ -525,103 +525,78 @@ static int prepare_playback_sync_urb(struct snd_usb_substream *subs, struct snd_urb_ctx *ctx = urb->context; urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 3; + urb->iso_frame_desc[0].length = min(4u, ctx->subs->syncmaxsize); urb->iso_frame_desc[0].offset = 0; return 0; } /* - * prepare urb for high speed playback sync pipe + * process after playback sync complete * - * set up the offset and length to receive the current frequency. - */ - -static int prepare_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - struct snd_urb_ctx *ctx = urb->context; - - urb->dev = ctx->subs->dev; /* we need to set this at each time */ - urb->iso_frame_desc[0].length = 4; - urb->iso_frame_desc[0].offset = 0; - return 0; -} - -/* - * process after full speed playback sync complete - * - * retrieve the current 10.14 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). + * Full speed devices report feedback values in 10.14 format as samples per + * frame, high speed devices in 16.16 format as samples per microframe. + * Because the Audio Class 1 spec was written before USB 2.0, many high speed + * devices use a wrong interpretation, some others use an entirely different + * format. Therefore, we cannot predict what format any particular device uses + * and must detect it automatically. */ static int retire_playback_sync_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { unsigned int f; + int shift; unsigned long flags; - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 3) { - f = combine_triple((u8*)urb->transfer_buffer) << 2; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } - } - - return 0; -} + if (urb->iso_frame_desc[0].status != 0 || + urb->iso_frame_desc[0].actual_length < 3) + return 0; -/* - * process after high speed playback sync complete - * - * retrieve the current 12.13 frequency from pipe, and set it. - * the value is referred in prepare_playback_urb(). - */ -static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; + f = le32_to_cpup(urb->transfer_buffer); + if (urb->iso_frame_desc[0].actual_length == 3) + f &= 0x00ffffff; + else + f &= 0x0fffffff; + if (f == 0) + return 0; - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); + if (unlikely(subs->freqshift == INT_MIN)) { + /* + * The first time we see a feedback value, determine its format + * by shifting it left or right until it matches the nominal + * frequency value. This assumes that the feedback does not + * differ from the nominal value more than +50% or -25%. + */ + shift = 0; + while (f < subs->freqn - subs->freqn / 4) { + f <<= 1; + shift++; + } + while (f > subs->freqn + subs->freqn / 2) { + f >>= 1; + shift--; } + subs->freqshift = shift; } + else if (subs->freqshift >= 0) + f <<= subs->freqshift; + else + f >>= -subs->freqshift; - return 0; -} - -/* - * process after E-Mu 0202/0404/Tracker Pre high speed playback sync complete - * - * These devices return the number of samples per packet instead of the number - * of samples per microframe. - */ -static int retire_playback_sync_urb_hs_emu(struct snd_usb_substream *subs, - struct snd_pcm_runtime *runtime, - struct urb *urb) -{ - unsigned int f; - unsigned long flags; - - if (urb->iso_frame_desc[0].status == 0 && - urb->iso_frame_desc[0].actual_length == 4) { - f = combine_quad((u8*)urb->transfer_buffer) & 0x0fffffff; - f >>= subs->datainterval; - if (f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax) { - spin_lock_irqsave(&subs->lock, flags); - subs->freqm = f; - spin_unlock_irqrestore(&subs->lock, flags); - } + if (likely(f >= subs->freqn - subs->freqn / 8 && f <= subs->freqmax)) { + /* + * If the frequency looks valid, set it. + * This value is referred to in prepare_playback_urb(). + */ + spin_lock_irqsave(&subs->lock, flags); + subs->freqm = f; + spin_unlock_irqrestore(&subs->lock, flags); + } else { + /* + * Out of range; maybe the shift value is wrong. + * Reset it so that we autodetect again the next time. + */ + subs->freqshift = INT_MIN; } return 0; @@ -878,21 +853,6 @@ static struct snd_urb_ops audio_urb_ops[2] = { }, }; -static struct snd_urb_ops audio_urb_ops_high_speed[2] = { - { - .prepare = prepare_nodata_playback_urb, - .retire = retire_playback_urb, - .prepare_sync = prepare_playback_sync_urb_hs, - .retire_sync = retire_playback_sync_urb_hs, - }, - { - .prepare = prepare_capture_urb, - .retire = retire_capture_urb, - .prepare_sync = prepare_capture_sync_urb_hs, - .retire_sync = retire_capture_sync_urb, - }, -}; - /* * initialize the substream instance. */ @@ -909,23 +869,9 @@ void snd_usb_init_substream(struct snd_usb_stream *as, subs->direction = stream; subs->dev = as->chip->dev; subs->txfr_quirk = as->chip->txfr_quirk; - if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { - subs->ops = audio_urb_ops[stream]; - } else { - subs->ops = audio_urb_ops_high_speed[stream]; - switch (as->chip->usb_id) { - case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ - case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ - case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ - subs->ops.retire_sync = retire_playback_sync_urb_hs_emu; - break; - case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra 8 */ - case USB_ID(0x0763, 0x2081): /* M-Audio Fast Track Ultra 8R */ - subs->ops.prepare_sync = prepare_playback_sync_urb; - subs->ops.retire_sync = retire_playback_sync_urb; - break; - } - } + subs->ops = audio_urb_ops[stream]; + if (snd_usb_get_speed(subs->dev) >= USB_SPEED_HIGH) + subs->ops.prepare_sync = prepare_capture_sync_urb_hs; snd_usb_set_pcm_ops(as->pcm, stream); -- cgit v1.2.3