diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 51a3d381a59e3f47e89f56803538e89c5354116a..9890cf2066ffef53e0773961cd0837d42ee4b740 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; } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 82ebeb9544fe54296ab3043d9a25e34cee04f415..93fa59cc60ef9ba0f5b2315c6461d3966608994c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5326,6 +5326,82 @@ static int patch_stac92hd73xx(struct hda_codec *codec) 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 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) AC_VERB_SET_CONNECT_SEL, num_dacs); } + stac92hd83xxx_set_system_btl_amp(codec); + codec->proc_widget_hook = stac92hd_proc_hook; return 0; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index c53955fe17b65875e8354063e897c6501edc7b50..de799cd1ba727e7b1fa45cad05f5e7aa9d0d80d3 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; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index e7a40d16df905748c74fe385618c9b41c6d5ad44..bc22ee93a75daf3c852ce65f9767df519445251c 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)); diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 7a1825418ee49a47d0d4bcbbb8ce8b28d298b143..99c046ba46bb6ed637d356579a2f4af9d1841bc3 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); diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index fe15bb26e48474e6f9aacb15961d62b59cea704d..25f27ec1dd6e4bba2590b869745d575c1d5b3545 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", }, }; diff --git a/sound/usb/card.h b/sound/usb/card.h index 1febf2f23754afc63eb5469450ac8da6a97f23b3..ae4251d5abf7bc9a1b9c15dc9d5d8e947627337a 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 f49756c1b83709716d8a60a8341dccb7c5f10f4f..cff3a3c465d736f17d73a4cf6c01c54c936469c3 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 3c650ab3c91de8e2edb87e3e0b03ce716b17d894..961c9a2506865b9a84cb337606e80b21705f2b04 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 8deeaad10f10caa4827288cc41230ce1cc6eb77e..e184349aee83f1e5a790deb43238186dab2b3355 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);