diff options
author | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2012-09-22 11:26:27 -0400 |
---|---|---|
committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2012-09-22 11:26:27 -0400 |
commit | ddfb43f3881edb47aa0083651ad31983cdc42c33 (patch) | |
tree | 7128cb13b599d2903a4deb05bbae37508b1c261e /sound | |
parent | 2d6d649a2e0fa0268c0d03d5b1d330ca7907d33c (diff) | |
parent | 5698bd757d55b1bb87edd1a9744ab09c142abfc2 (diff) |
Merge tag 'v3.6-rc6' into for-3.7
Linux 3.6-rc6 has all our bug fixes.
Conflicts (trivial overlap):
sound/soc/omap/am3517evm.c
Diffstat (limited to 'sound')
-rw-r--r-- | sound/core/compress_offload.c | 8 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.c | 12 | ||||
-rw-r--r-- | sound/pci/hda/hda_codec.h | 1 | ||||
-rw-r--r-- | sound/pci/hda/hda_intel.c | 2 | ||||
-rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 6 | ||||
-rw-r--r-- | sound/pci/ice1712/prodigy_hifi.c | 3 | ||||
-rw-r--r-- | sound/soc/codecs/arizona.c | 2 | ||||
-rw-r--r-- | sound/soc/codecs/mc13783.c | 8 | ||||
-rw-r--r-- | sound/soc/codecs/wm8904.c | 2 | ||||
-rw-r--r-- | sound/soc/fsl/imx-sgtl5000.c | 2 | ||||
-rw-r--r-- | sound/soc/samsung/dma.c | 8 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 5 | ||||
-rw-r--r-- | sound/soc/spear/spear_pcm.c | 2 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_alc5632.c | 1 | ||||
-rw-r--r-- | sound/soc/tegra/tegra_pcm.c | 4 | ||||
-rw-r--r-- | sound/soc/ux500/ux500_msp_i2s.c | 25 | ||||
-rw-r--r-- | sound/usb/card.c | 4 | ||||
-rw-r--r-- | sound/usb/endpoint.c | 24 | ||||
-rw-r--r-- | sound/usb/endpoint.h | 3 | ||||
-rw-r--r-- | sound/usb/pcm.c | 70 |
20 files changed, 114 insertions, 78 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index ec2118d0e27a..eb60cb8dbb8a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -80,14 +80,12 @@ static int snd_compr_open(struct inode *inode, struct file *f) int maj = imajor(inode); int ret; - if (f->f_flags & O_WRONLY) + if ((f->f_flags & O_ACCMODE) == O_WRONLY) dirn = SND_COMPRESS_PLAYBACK; - else if (f->f_flags & O_RDONLY) + else if ((f->f_flags & O_ACCMODE) == O_RDONLY) dirn = SND_COMPRESS_CAPTURE; - else { - pr_err("invalid direction\n"); + else return -EINVAL; - } if (maj == snd_major) compr = snd_lookup_minor_data(iminor(inode), diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f560051a949e..1c65cc5e3a31 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1320,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->epss = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_EPSS); + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -2346,6 +2353,7 @@ int snd_hda_codec_reset(struct hda_codec *codec) } if (codec->patch_ops.free) codec->patch_ops.free(codec); + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); snd_hda_jack_tbl_clear(codec); codec->proc_widget_hook = NULL; codec->spec = NULL; @@ -2361,7 +2369,6 @@ int snd_hda_codec_reset(struct hda_codec *codec) codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; - memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); codec->slave_dig_outs = NULL; codec->spdif_status_reset = 0; module_put(codec->owner); @@ -3543,8 +3550,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, /* this delay seems necessary to avoid click noise at power-down */ if (power_state == AC_PWRST_D3) { /* transition time less than 10ms for power down */ - bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS); - msleep(epss ? 10 : 100); + msleep(codec->epss ? 10 : 100); } /* repeat power states setting at most 10 times*/ diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7fbc1bcaf1a9..e5a7e19a8071 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -862,6 +862,7 @@ struct hda_codec { unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */ unsigned int no_jack_detect:1; /* Machine has no jack-detection */ unsigned int pcm_format_first:1; /* PCM format must be set first */ + unsigned int epss:1; /* supporting EPSS? */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ int power_transition; /* power-state in transition */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 60882c62f180..c4763c52eaf6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2701,6 +2701,8 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1043, 0x1ac3, "ASUS X53S", POS_FIX_POSBUF), + SND_PCI_QUIRK(0x1043, 0x1b43, "ASUS K53E", POS_FIX_POSBUF), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea5775a1a7db..3d4722f0a1ca 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1075,7 +1075,7 @@ static struct snd_kcontrol_new stac_smux_mixer = { static const char * const slave_pfxs[] = { "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Speaker", "IEC958", + "Headphone", "Speaker", "IEC958", "PCM", NULL }; @@ -4543,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec, struct auto_pin_cfg *cfg = &spec->autocfg; int i; + if (cfg->speaker_outs == 0) + return; + for (i = 0; i < cfg->line_outs; i++) { if (presence) break; @@ -5531,6 +5534,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) snd_hda_codec_set_pincfg(codec, 0xf, 0x2181205e); } + codec->epss = 0; /* longer delay needed for D3 */ codec->no_trigger_sense = 1; codec->spec = spec; diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 764cc93dbca4..075d5aa1fee0 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -297,6 +297,7 @@ static int ak4396_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem } static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1); +static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { { @@ -307,7 +308,7 @@ static struct snd_kcontrol_new prodigy_hd2_controls[] __devinitdata = { .info = ak4396_dac_vol_info, .get = ak4396_dac_vol_get, .put = ak4396_dac_vol_put, - .tlv = { .p = db_scale_wm_dac }, + .tlv = { .p = ak4396_db_scale }, }, }; diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index b79578e7e10c..c167c896eaee 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -492,7 +492,7 @@ static const int arizona_44k1_bclk_rates[] = { 940800, 1411200, 1881600, - 2882400, + 2822400, 3763200, 5644800, 7526400, diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index d89e343ff108..bc955999c8aa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -661,7 +661,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_DAC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, @@ -672,7 +672,7 @@ static struct snd_soc_dai_driver mc13783_dai_async[] = { .id = MC13783_ID_STEREO_CODEC, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, @@ -694,14 +694,14 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { .id = MC13783_ID_SYNC, .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = MC13783_FORMATS, }, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = MC13783_RATES_RECORD, .formats = MC13783_FORMATS, diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 2b9766d55384..7c8df52a8d9d 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -100,7 +100,7 @@ static const struct reg_default wm8904_reg_defaults[] = { { 14, 0x0000 }, /* R14 - Power Management 2 */ { 15, 0x0000 }, /* R15 - Power Management 3 */ { 18, 0x0000 }, /* R18 - Power Management 6 */ - { 19, 0x945E }, /* R20 - Clock Rates 0 */ + { 20, 0x945E }, /* R20 - Clock Rates 0 */ { 21, 0x0C05 }, /* R21 - Clock Rates 1 */ { 22, 0x0006 }, /* R22 - Clock Rates 2 */ { 24, 0x0050 }, /* R24 - Audio Interface 0 */ diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index fb21b17f17f5..199408ec4261 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -94,7 +94,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) dev_err(&pdev->dev, "audmux internal port setup failed\n"); return ret; } - imx_audmux_v2_configure_port(ext_port, + ret = imx_audmux_v2_configure_port(ext_port, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38c10fe..b70964ea448c 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -34,9 +34,7 @@ static const struct snd_pcm_hardware dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U8 | @@ -248,15 +246,11 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: prtd->state &= ~ST_RUNNING; prtd->params->ops->stop(prtd->params->ch); break; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a18d115bc507..873e6e76ee87 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -313,8 +313,11 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, if (dapm->codec->driver->set_bias_level) ret = dapm->codec->driver->set_bias_level(dapm->codec, level); - } else + else + dapm->bias_level = level; + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 97c2cac8e92c..8c7f23729446 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -138,7 +138,7 @@ static void spear_pcm_free(struct snd_pcm *pcm) continue; buf = &substream->dma_buffer; - if (!buf && !buf->area) + if (!buf || !buf->area) continue; dma_free_writecombine(pcm->card->dev, buf->bytes, diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index e463529b38bb..76cb1b363b71 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -89,7 +89,6 @@ static struct snd_soc_jack_gpio tegra_alc5632_hp_jack_gpio = { .name = "Headset detection", .report = SND_JACK_HEADSET, .debounce_time = 150, - .invert = 1, }; static const struct snd_soc_dapm_widget tegra_alc5632_dapm_widgets[] = { diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 5658bcec1931..8d6900c1ee47 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -334,11 +334,11 @@ static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.dst_addr = dmap->addr; - slave_config.src_maxburst = 0; + slave_config.dst_maxburst = 4; } else { slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; slave_config.src_addr = dmap->addr; - slave_config.dst_maxburst = 0; + slave_config.src_maxburst = 4; } slave_config.slave_id = dmap->req_sel; diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 36be11e47ad6..1b7c2f58ce13 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -663,7 +663,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, struct msp_i2s_platform_data *platform_data) { - int ret = 0; struct resource *res = NULL; struct i2s_controller *i2s_cont; struct ux500_msp *msp; @@ -687,15 +686,14 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, if (res == NULL) { dev_err(&pdev->dev, "%s: ERROR: Unable to get resource!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } - msp->registers = ioremap(res->start, (res->end - res->start + 1)); + msp->registers = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); if (msp->registers == NULL) { dev_err(&pdev->dev, "%s: ERROR: ioremap failed!\n", __func__); - ret = -ENOMEM; - goto err_res; + return -ENOMEM; } msp->msp_state = MSP_STATE_IDLE; @@ -707,7 +705,7 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, dev_err(&pdev->dev, "%s: ERROR: Failed to allocate I2S-controller!\n", __func__); - goto err_i2s_cont; + return -ENOMEM; } i2s_cont->dev.parent = &pdev->dev; i2s_cont->data = (void *)msp; @@ -718,14 +716,6 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, msp->i2s_cont = i2s_cont; return 0; - -err_i2s_cont: - iounmap(msp->registers); - -err_res: - devm_kfree(&pdev->dev, msp); - - return ret; } void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, @@ -734,11 +724,6 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, dev_dbg(msp->dev, "%s: Enter (id = %d).\n", __func__, msp->id); device_unregister(&msp->i2s_cont->dev); - devm_kfree(&pdev->dev, msp->i2s_cont); - - iounmap(msp->registers); - - devm_kfree(&pdev->dev, msp); } MODULE_LICENSE("GPL v2"); diff --git a/sound/usb/card.c b/sound/usb/card.c index d5b5c3388e28..4a469f0cb6d4 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -553,7 +553,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, struct snd_usb_audio *chip) { struct snd_card *card; - struct list_head *p; + struct list_head *p, *n; if (chip == (void *)-1L) return; @@ -570,7 +570,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, snd_usb_stream_disconnect(p); } /* release the endpoint resources */ - list_for_each(p, &chip->ep_list) { + list_for_each_safe(p, n, &chip->ep_list) { snd_usb_endpoint_free(p); } /* release the midi resources */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index c41181202688..d6e2bb49c59c 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -141,7 +141,7 @@ int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep) * * For implicit feedback, next_packet_size() is unused. */ -static int next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -177,15 +177,6 @@ static void retire_inbound_urb(struct snd_usb_endpoint *ep, ep->retire_data_urb(ep->data_subs, urb); } -static void prepare_outbound_urb_sizes(struct snd_usb_endpoint *ep, - struct snd_urb_ctx *ctx) -{ - int i; - - for (i = 0; i < ctx->packets; ++i) - ctx->packet_size[i] = next_packet_size(ep); -} - /* * Prepare a PLAYBACK urb for submission to the bus. */ @@ -370,7 +361,6 @@ static void snd_complete_urb(struct urb *urb) goto exit_clear; } - prepare_outbound_urb_sizes(ep, ctx); prepare_outbound_urb(ep, ctx); } else { retire_inbound_urb(ep, ctx); @@ -799,7 +789,9 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, /** * snd_usb_endpoint_start: start an snd_usb_endpoint * - * @ep: the endpoint to start + * @ep: the endpoint to start + * @can_sleep: flag indicating whether the operation is executed in + * non-atomic context * * A call to this function will increment the use count of the endpoint. * In case it is not already running, the URBs for this endpoint will be @@ -809,7 +801,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, * * Returns an error if the URB submission failed, 0 in all other cases. */ -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep) { int err; unsigned int i; @@ -821,6 +813,11 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (++ep->use_count != 1) return 0; + /* just to be sure */ + deactivate_urbs(ep, 0, can_sleep); + if (can_sleep) + wait_clear_urbs(ep); + ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; @@ -850,7 +847,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) goto __error; if (usb_pipeout(ep->pipe)) { - prepare_outbound_urb_sizes(ep, urb->context); prepare_outbound_urb(ep, urb->context); } else { prepare_inbound_urb(ep, urb->context); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index ee2723fb174f..cbbbdf226d66 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -13,7 +13,7 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, struct audioformat *fmt, struct snd_usb_endpoint *sync_ep); -int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_start(struct snd_usb_endpoint *ep, int can_sleep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, int force, int can_sleep, int wait); int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); @@ -21,6 +21,7 @@ int snd_usb_endpoint_deactivate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct list_head *head); int snd_usb_endpoint_implict_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, struct snd_usb_endpoint *sender, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 62ec808ed792..f782ce19bf5a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -212,7 +212,7 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, } } -static int start_endpoints(struct snd_usb_substream *subs) +static int start_endpoints(struct snd_usb_substream *subs, int can_sleep) { int err; @@ -225,7 +225,7 @@ static int start_endpoints(struct snd_usb_substream *subs) snd_printdd(KERN_DEBUG "Starting data EP @%p\n", ep); ep->data_subs = subs; - err = snd_usb_endpoint_start(ep); + err = snd_usb_endpoint_start(ep, can_sleep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_DATA_EP_STARTED, &subs->flags); return err; @@ -236,10 +236,25 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->alt_idx != subs->sync_endpoint->alt_idx) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->alt_idx); + if (err < 0) { + snd_printk(KERN_ERR + "%d:%d:%d: cannot set interface (%d)\n", + subs->dev->devnum, + subs->sync_endpoint->iface, + subs->sync_endpoint->alt_idx, err); + return -EIO; + } + } + snd_printdd(KERN_DEBUG "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; - err = snd_usb_endpoint_start(ep); + err = snd_usb_endpoint_start(ep, can_sleep); if (err < 0) { clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); return err; @@ -544,13 +559,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) subs->last_frame_number = 0; runtime->delay = 0; - /* clear the pending deactivation on the target EPs */ - deactivate_endpoints(subs); - /* for playback, submit the URBs now; otherwise, the first hwptr_done * updates for all URBs would happen at the same time when starting */ if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) - return start_endpoints(subs); + return start_endpoints(subs, 1); return 0; } @@ -1032,6 +1044,7 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, struct urb *urb) { struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; + struct snd_usb_endpoint *ep = subs->data_endpoint; struct snd_urb_ctx *ctx = urb->context; unsigned int counts, frames, bytes; int i, stride, period_elapsed = 0; @@ -1043,7 +1056,11 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { - counts = ctx->packet_size[i]; + if (ctx->packet_size[i]) + counts = ctx->packet_size[i]; + else + counts = snd_usb_endpoint_next_packet_size(ep); + /* set up descriptor */ urb->iso_frame_desc[i].offset = frames * stride; urb->iso_frame_desc[i].length = counts * stride; @@ -1094,7 +1111,16 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + + /* update delay with exact number of samples queued */ + runtime->delay = subs->last_delay; runtime->delay += frames; + subs->last_delay = runtime->delay; + + /* realign last_frame_number */ + subs->last_frame_number = usb_get_current_frame_number(subs->dev); + subs->last_frame_number &= 0xFF; /* keep 8 LSBs */ + spin_unlock_irqrestore(&subs->lock, flags); urb->transfer_buffer_length = bytes; if (period_elapsed) @@ -1112,12 +1138,32 @@ static void retire_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime = subs->pcm_substream->runtime; int stride = runtime->frame_bits >> 3; int processed = urb->transfer_buffer_length / stride; + int est_delay; + + /* ignore the delay accounting when procssed=0 is given, i.e. + * silent payloads are procssed before handling the actual data + */ + if (!processed) + return; spin_lock_irqsave(&subs->lock, flags); - if (processed > runtime->delay) - runtime->delay = 0; + est_delay = snd_usb_pcm_delay(subs, runtime->rate); + /* update delay with exact number of samples played */ + if (processed > subs->last_delay) + subs->last_delay = 0; else - runtime->delay -= processed; + subs->last_delay -= processed; + runtime->delay = subs->last_delay; + + /* + * Report when delay estimate is off by more than 2ms. + * The error should be lower than 2ms since the estimate relies + * on two reads of a counter updated every ms. + */ + if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + snd_printk(KERN_DEBUG "delay: estimated %d, actual %d\n", + est_delay, subs->last_delay); + spin_unlock_irqrestore(&subs->lock, flags); } @@ -1175,7 +1221,7 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream switch (cmd) { case SNDRV_PCM_TRIGGER_START: - err = start_endpoints(subs); + err = start_endpoints(subs, 0); if (err < 0) return err; |