From 4e38b745af76cdd93bca33258e1e33a23458f92c Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 31 Oct 2012 16:07:43 +0800 Subject: ASoC: si476x: Add missing break for SNDRV_PCM_FORMAT_S8 switch case Signed-off-by: Axel Lin Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f2d61a187830..566ea3256e2d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -159,6 +159,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: width = SI476X_PCM_FORMAT_S8; + break; case SNDRV_PCM_FORMAT_S16_LE: width = SI476X_PCM_FORMAT_S16_LE; break; -- cgit v1.2.3 From 8af294b472067e9034fe288d912455cc0961d1b9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Feb 2013 17:48:15 +0000 Subject: ASoC: dapm: Fix handling of loops Currently if a path loops back on itself we correctly skip over it to avoid going into an infinite loop but this causes us to ignore the need to power up the path as we don't count the loop for the purposes of counting inputs and outputs. This means that internal loopbacks within a device that have powered devices on them won't be powered up. Fix this by treating any path that is currently in the process of being recursed as having a single input or output so that it is counted for the purposes of power decisions. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc-dapm.h | 1 + sound/soc/soc-dapm.c | 15 +++++++++++++++ 2 files changed, 16 insertions(+) (limited to 'sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e1ef63d4a5c4..44a30b108683 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -488,6 +488,7 @@ struct snd_soc_dapm_path { /* status */ u32 connect:1; /* source and sink widgets are connected */ u32 walked:1; /* path has been walked */ + u32 walking:1; /* path is in the process of being walked */ u32 weak:1; /* path ignored for power management */ int (*connected)(struct snd_soc_dapm_widget *source, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 258acadb9e7d..f3255517de79 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,6 +821,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) { widget->outputs = snd_soc_dapm_suspend_check(widget); + path->walking = 0; return widget->outputs; } } @@ -831,6 +832,9 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -838,6 +842,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, if (path->sink && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -847,11 +852,14 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_output_ep(path->sink, list); + + path->walking = 0; } } @@ -931,6 +939,9 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->weak) continue; + if (path->walking) + return 1; + if (path->walked) continue; @@ -938,6 +949,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, if (path->source && path->connect) { path->walked = 1; + path->walking = 1; /* do we need to add this widget to the list ? */ if (list) { @@ -947,11 +959,14 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, dev_err(widget->dapm->dev, "ASoC: could not add widget %s\n", widget->name); + path->walking = 0; return con; } } con += is_connected_input_ep(path->source, list); + + path->walking = 0; } } -- cgit v1.2.3 From f4b828128ab64fd9dc5eec9525b38fbfeafa5c0e Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 12 Mar 2013 00:23:15 +0800 Subject: ASoC: wm_adsp: fix possible memory leak in wm_adsp_load_coeff() 'file' is malloced in wm_adsp_load_coeff() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3f7e75f8628..9af1bddc4c62 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -828,7 +828,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } adsp_dbg(dsp, "%s.%d: Writing %d bytes at %x\n", @@ -865,7 +866,7 @@ out_fw: wm_adsp_buf_free(&buf_list); out: kfree(file); - return 0; + return ret; } int wm_adsp1_init(struct wm_adsp *adsp) -- cgit v1.2.3 From e8b18addee32d1f389573b4c116e67ae230216ad Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 12 Mar 2013 00:35:14 +0800 Subject: ASoC: core: fix possible memory leak in snd_soc_bytes_put() 'data' is malloced in snd_soc_bytes_put() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b7e84a7cd9ee..93341deaa4b9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,7 +3140,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, if (params->mask) { ret = regmap_read(codec->control_data, params->base, &val); if (ret != 0) - return ret; + goto out; val &= params->mask; @@ -3158,13 +3158,15 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ((u32 *)data)[0] |= cpu_to_be32(val); break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } } ret = regmap_raw_write(codec->control_data, params->base, data, len); +out: kfree(data); return ret; -- cgit v1.2.3 From b6e51600f4e983e757b1b6942becaa1ae7d82e67 Mon Sep 17 00:00:00 2001 From: Sascha Hauer Date: Sun, 10 Mar 2013 19:33:03 +0100 Subject: ASoC: imx-ssi: Fix occasional AC97 reset failure Signed-off-by: Sascha Hauer Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/fsl/imx-ssi.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 55464a5b0706..810c7eeb7b03 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -496,6 +496,8 @@ static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_reset) imx_ssi->ac97_reset(ac97); + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) @@ -504,6 +506,9 @@ static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) if (imx_ssi->ac97_warm_reset) imx_ssi->ac97_warm_reset(ac97); + + /* First read sometimes fails, do a dummy read */ + imx_ssi_ac97_read(ac97, 0); } struct snd_ac97_bus_ops soc_ac97_ops = { -- cgit v1.2.3 From 34913fd950dc1817d466d76bdccd63443fdcbb12 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Sun, 10 Mar 2013 19:33:06 +0100 Subject: ASoC: pcm030 audio fabric: remove __init from probe Remove probe function from the init section. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8e52c1485df3..eb4373840bb6 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -51,7 +51,7 @@ static struct snd_soc_card pcm030_card = { .num_links = ARRAY_SIZE(pcm030_fabric_dai), }; -static int __init pcm030_fabric_probe(struct platform_device *op) +static int pcm030_fabric_probe(struct platform_device *op) { struct device_node *np = op->dev.of_node; struct device_node *platform_np; -- cgit v1.2.3 From 7f08a89862b96d84c6dfe6c242eb010084e51d3b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 14 Mar 2013 21:26:24 +0100 Subject: ASoC: dapm: Fix pointer dereference in is_connected_output_ep() *path is not yet initialized when we check if the widget is connected. The compiler also warns about this: sound/soc/soc-dapm.c: In function 'is_connected_output_ep': sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f3255517de79..ab621b1db105 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -821,7 +821,6 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) { widget->outputs = snd_soc_dapm_suspend_check(widget); - path->walking = 0; return widget->outputs; } } -- cgit v1.2.3 From 31b6945a899a30f9dffa9cba8ed2e494784810a9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 17 Mar 2013 10:23:40 +0100 Subject: ALSA: hda - Fix missing beep detach in patch_conexant.c This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 941bf6c766ec..1051a88f5304 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3191,11 +3191,17 @@ static int cx_auto_build_controls(struct hda_codec *codec) return 0; } +static void cx_auto_free(struct hda_codec *codec) +{ + snd_hda_detach_beep_device(codec); + snd_hda_gen_free(codec); +} + static const struct hda_codec_ops cx_auto_patch_ops = { .build_controls = cx_auto_build_controls, .build_pcms = snd_hda_gen_build_pcms, .init = snd_hda_gen_init, - .free = snd_hda_gen_free, + .free = cx_auto_free, .unsol_event = snd_hda_jack_unsol_event, #ifdef CONFIG_PM .check_power_status = snd_hda_gen_check_power_status, -- cgit v1.2.3 From a86b1a2cd2f81f74e815e07f756edd7bc5b6f034 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Mar 2013 11:00:44 +0100 Subject: ALSA: hda/cirrus - Fix the digital beep registration The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1051a88f5304..2a89d1eefeb6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1142,7 +1142,7 @@ static int patch_cxt5045(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -1921,7 +1921,7 @@ static int patch_cxt5051(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -3099,7 +3099,7 @@ static int patch_cxt5066(struct hda_codec *codec) } if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); return 0; } @@ -3397,7 +3397,7 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->patch_ops = cx_auto_patch_ops; if (spec->beep_amp) - snd_hda_attach_beep_device(codec, spec->beep_amp); + snd_hda_attach_beep_device(codec, get_amp_nid_(spec->beep_amp)); /* Some laptops with Conexant chips show stalls in S3 resume, * which falls into the single-cmd mode. -- cgit v1.2.3 From 039eb75350acd1131a18a9bd12a0d4e1fb17892e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Mar 2013 16:55:49 +0100 Subject: ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver I forgot to update spec->gpio_data in the automute hook, so it will be overridden at the init sequence, thus the machine is still silent when no headphone jack is plugged at boot time. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 60d08f669f0c..0d9c58f13560 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -168,10 +168,10 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); if (spec->gpio_eapd_hp) { - unsigned int gpio = spec->gen.hp_jack_present ? + spec->gpio_data = spec->gen.hp_jack_present ? spec->gpio_eapd_hp : spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, - AC_VERB_SET_GPIO_DATA, gpio); + AC_VERB_SET_GPIO_DATA, spec->gpio_data); } } -- cgit v1.2.3 From 61ac51301e6c6d4ed977d7674ce2b8e713619a9b Mon Sep 17 00:00:00 2001 From: Torstein Hegge Date: Tue, 19 Mar 2013 17:12:14 +0100 Subject: ALSA: usb: Parse UAC2 extension unit like for UAC1 UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in the same way when parsing the unit. Otherwise parse_audio_unit() fails when it sees an extension unit on a UAC2 device. UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1. Signed-off-by: Torstein Hegge Acked-by: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 638e7f738018..8eb84c0f7bf1 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -725,7 +725,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC1_PROCESSING_UNIT: case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ - /* UAC2_EFFECT_UNIT */ { + /* UAC2_EFFECT_UNIT */ + case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; if (state->mixer->protocol == UAC_VERSION_2 && @@ -2052,6 +2053,8 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_extension_unit(state, unitid, p1); else /* UAC_VERSION_2 */ return parse_audio_processing_unit(state, unitid, p1); + case UAC2_EXTENSION_UNIT_V2: + return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); return -EINVAL; -- cgit v1.2.3 From 4d7b86c98e445b075c2c4c3757eb6d3d6efbe72e Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 19 Mar 2013 21:09:24 +0100 Subject: ALSA: snd-usb: mixer: propagate errors up the call chain In check_input_term() and parse_audio_feature_unit(), propagate the error value that has been returned by a failing function instead of -EINVAL. That helps cleaning up the error pathes in the mixer. Signed-off-by: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 8eb84c0f7bf1..45cc0aff9c3e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,8 +715,9 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ case UAC2_CLOCK_SELECTOR: { struct uac_selector_unit_descriptor *d = p1; /* call recursively to retrieve the channel info */ - if (check_input_term(state, d->baSourceID[0], term) < 0) - return -ENODEV; + err = check_input_term(state, d->baSourceID[0], term); + if (err < 0) + return err; term->type = d->bDescriptorSubtype << 16; /* virtual type */ term->id = id; term->name = uac_selector_unit_iSelector(d); @@ -1357,8 +1358,9 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void return err; /* determine the input source type and name */ - if (check_input_term(state, hdr->bSourceID, &iterm) < 0) - return -EINVAL; + err = check_input_term(state, hdr->bSourceID, &iterm); + if (err < 0) + return err; master_bits = snd_usb_combine_bytes(bmaControls, csize); /* master configuration quirks */ -- cgit v1.2.3 From 83ea5d18d74f032a760fecde78c0210f66f7f70c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 19 Mar 2013 21:09:25 +0100 Subject: ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls() Creation of individual mixer controls may fail, but that shouldn't cause the entire mixer creation to fail. Even worse, if the mixer creation fails, that will error out the entire device probing. All the functions called by parse_audio_unit() should return -EINVAL if they find descriptors that are unsupported or believed to be malformed, so we can safely handle this error code as a non-fatal condition in snd_usb_mixer_controls(). That fixes a long standing bug which is commonly worked around by adding quirks which make the driver ignore entire interfaces. Some of them might now be unnecessary. Signed-off-by: Daniel Mack Reported-and-tested-by: Rodolfo Thomazelli Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 45cc0aff9c3e..ca4739c3f650 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2123,7 +2123,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; } else { /* UAC_VERSION_2 */ struct uac2_output_terminal_descriptor *desc = p; @@ -2135,12 +2135,12 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.type = le16_to_cpu(desc->wTerminalType); state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; /* for UAC2, use the same approach to also add the clock selectors */ err = parse_audio_unit(&state, desc->bCSourceID); - if (err < 0) + if (err < 0 && err != -EINVAL) return err; } } -- cgit v1.2.3 From 4480764f57ba494e3f64003e13223c0b5ec6a2ca Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Tue, 19 Mar 2013 14:58:43 -0700 Subject: ASoC:: max98090: Remove executable bit Source files shouldn't have the executable bit set. Signed-off-by: Joe Perches Signed-off-by: Mark Brown --- include/sound/max98090.h | 0 sound/soc/codecs/max98090.c | 0 sound/soc/codecs/max98090.h | 0 3 files changed, 0 insertions(+), 0 deletions(-) mode change 100755 => 100644 include/sound/max98090.h mode change 100755 => 100644 sound/soc/codecs/max98090.c mode change 100755 => 100644 sound/soc/codecs/max98090.h (limited to 'sound') diff --git a/include/sound/max98090.h b/include/sound/max98090.h old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h old mode 100755 new mode 100644 -- cgit v1.2.3 From 59d9cc2a5073ab4b8c8f8bdbacf230a538abc55d Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 18 Mar 2013 18:57:23 +0100 Subject: ASoC: spear_pcm: Update to new pcm_new() API Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only rtd") updated the pcm_new() callback to take the rtd as the only parameter. The spear PCM driver (which was merged much later) still uses the old API. This patch updates the driver to the new API. Signed-off-by: Lars-Peter Clausen Acked-by: Rajeev Kumar Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/spear/spear_pcm.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index 9b76cc5a1148..5e7aebe1e664 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -149,9 +149,9 @@ static void spear_pcm_free(struct snd_pcm *pcm) static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); -static int spear_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int spear_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; int ret; if (!card->dev->dma_mask) @@ -159,16 +159,16 @@ static int spear_pcm_new(struct snd_card *card, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, spear_pcm_hardware.buffer_bytes_max); if (ret) return ret; } - if (dai->driver->capture.channels_min) { - ret = spear_pcm_preallocate_dma_buffer(pcm, + if (rtd->cpu_dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(rtd->pcm, SNDRV_PCM_STREAM_CAPTURE, spear_pcm_hardware.buffer_bytes_max); if (ret) -- cgit v1.2.3 From f7ba716f1e704a00d682a8697108f9c86497c551 Mon Sep 17 00:00:00 2001 From: Silviu-Mihai Popescu Date: Sat, 16 Mar 2013 13:45:34 +0200 Subject: ASoC: core: fix invalid free of devm_ allocated data The objects allocated by devm_* APIs are managed by devres and are freed when the device is detached. Hence there is no need to use kfree() explicitly. Signed-off-by: Silviu-Mihai Popescu Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93341deaa4b9..507d251916af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4199,7 +4199,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4208,7 +4207,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "ASoC: Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } } -- cgit v1.2.3 From a686fd141e20244ad75f80ad54706da07d7bb90a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Mar 2013 15:42:00 +0100 Subject: ALSA: hda - Fix typo in checking IEC958 emphasis bit There is a typo in convert_to_spdif_status() about checking the emphasis IEC958 status bit. It should check the given value instead of the resultant value. Reported-by: Martin Weishart Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a9ebcf9e3710..ecdf30eb5879 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3144,7 +3144,7 @@ static unsigned int convert_to_spdif_status(unsigned short val) if (val & AC_DIG1_PROFESSIONAL) sbits |= IEC958_AES0_PROFESSIONAL; if (sbits & IEC958_AES0_PROFESSIONAL) { - if (sbits & AC_DIG1_EMPHASIS) + if (val & AC_DIG1_EMPHASIS) sbits |= IEC958_AES0_PRO_EMPHASIS_5015; } else { if (val & AC_DIG1_EMPHASIS) -- cgit v1.2.3 From eb49faa6a4703698fa5d8b304b01e7f59e7d1f11 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Mar 2013 09:19:11 +0100 Subject: ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader The current DSP loader code abuses snd_hda_lock_devices() for ensuring the DSP loader not conflicting with the other normal operations. But this trick obviously doesn't work for the PM resume since the streams are kept opened there where snd_hda_lock_devices() returns -EBUSY. That means we need another lock mechanism instead of abuse. This patch provides the new lock state to azx_dev. Theoretically it's possible that the DSP loader conflicts with the stream that has been already assigned for another PCM. If it's running, the DSP loader should simply fail. If not -- it's the case for PM resume --, we should assign this stream temporarily to the DSP loader, and take it back to the PCM after finishing DSP loading. If the PCM is operated during the DSP loading, it should get an error, too. Reported-and-tested-by: Dylan Reid Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 132 ++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 109 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4cea6bb6fade..418bfc0eb0a3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -415,6 +415,8 @@ struct azx_dev { unsigned int opened :1; unsigned int running :1; unsigned int irq_pending :1; + unsigned int prepared:1; + unsigned int locked:1; /* * For VIA: * A flag to ensure DMA position is 0 @@ -426,8 +428,25 @@ struct azx_dev { struct timecounter azx_tc; struct cyclecounter azx_cc; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct mutex dsp_mutex; +#endif }; +/* DSP lock helpers */ +#ifdef CONFIG_SND_HDA_DSP_LOADER +#define dsp_lock_init(dev) mutex_init(&(dev)->dsp_mutex) +#define dsp_lock(dev) mutex_lock(&(dev)->dsp_mutex) +#define dsp_unlock(dev) mutex_unlock(&(dev)->dsp_mutex) +#define dsp_is_locked(dev) ((dev)->locked) +#else +#define dsp_lock_init(dev) do {} while (0) +#define dsp_lock(dev) do {} while (0) +#define dsp_unlock(dev) do {} while (0) +#define dsp_is_locked(dev) 0 +#endif + /* CORB/RIRB */ struct azx_rb { u32 *buf; /* CORB/RIRB buffer @@ -527,6 +546,10 @@ struct azx { /* card list (for power_save trigger) */ struct list_head list; + +#ifdef CONFIG_SND_HDA_DSP_LOADER + struct azx_dev saved_azx_dev; +#endif }; #define CREATE_TRACE_POINTS @@ -1793,15 +1816,25 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) dev = chip->capture_index_offset; nums = chip->capture_streams; } - for (i = 0; i < nums; i++, dev++) - if (!chip->azx_dev[dev].opened) { - res = &chip->azx_dev[dev]; - if (res->assigned_key == key) - break; + for (i = 0; i < nums; i++, dev++) { + struct azx_dev *azx_dev = &chip->azx_dev[dev]; + dsp_lock(azx_dev); + if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { + res = azx_dev; + if (res->assigned_key == key) { + res->opened = 1; + res->assigned_key = key; + dsp_unlock(azx_dev); + return azx_dev; + } } + dsp_unlock(azx_dev); + } if (res) { + dsp_lock(res); res->opened = 1; res->assigned_key = key; + dsp_unlock(res); } return res; } @@ -2009,6 +2042,12 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct azx_dev *azx_dev = get_azx_dev(substream); int ret; + dsp_lock(azx_dev); + if (dsp_is_locked(azx_dev)) { + ret = -EBUSY; + goto unlock; + } + mark_runtime_wc(chip, azx_dev, substream, false); azx_dev->bufsize = 0; azx_dev->period_bytes = 0; @@ -2016,8 +2055,10 @@ static int azx_pcm_hw_params(struct snd_pcm_substream *substream, ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto unlock; mark_runtime_wc(chip, azx_dev, substream, true); + unlock: + dsp_unlock(azx_dev); return ret; } @@ -2029,16 +2070,21 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; /* reset BDL address */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); - azx_sd_writel(azx_dev, SD_CTL, 0); - azx_dev->bufsize = 0; - azx_dev->period_bytes = 0; - azx_dev->format_val = 0; + dsp_lock(azx_dev); + if (!dsp_is_locked(azx_dev)) { + azx_sd_writel(azx_dev, SD_BDLPL, 0); + azx_sd_writel(azx_dev, SD_BDLPU, 0); + azx_sd_writel(azx_dev, SD_CTL, 0); + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + } snd_hda_codec_cleanup(apcm->codec, hinfo, substream); mark_runtime_wc(chip, azx_dev, substream, false); + azx_dev->prepared = 0; + dsp_unlock(azx_dev); return snd_pcm_lib_free_pages(substream); } @@ -2055,6 +2101,12 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_hda_spdif_out_of_nid(apcm->codec, hinfo->nid); unsigned short ctls = spdif ? spdif->ctls : 0; + dsp_lock(azx_dev); + if (dsp_is_locked(azx_dev)) { + err = -EBUSY; + goto unlock; + } + azx_stream_reset(chip, azx_dev); format_val = snd_hda_calc_stream_format(runtime->rate, runtime->channels, @@ -2065,7 +2117,8 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) snd_printk(KERN_ERR SFX "%s: invalid format_val, rate=%d, ch=%d, format=%d\n", pci_name(chip->pci), runtime->rate, runtime->channels, runtime->format); - return -EINVAL; + err = -EINVAL; + goto unlock; } bufsize = snd_pcm_lib_buffer_bytes(substream); @@ -2084,7 +2137,7 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) azx_dev->no_period_wakeup = runtime->no_period_wakeup; err = azx_setup_periods(chip, substream, azx_dev); if (err < 0) - return err; + goto unlock; } /* wallclk has 24Mhz clock source */ @@ -2101,8 +2154,14 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) if ((chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) && stream_tag > chip->capture_streams) stream_tag -= chip->capture_streams; - return snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag, + err = snd_hda_codec_prepare(apcm->codec, hinfo, stream_tag, azx_dev->format_val, substream); + + unlock: + if (!err) + azx_dev->prepared = 1; + dsp_unlock(azx_dev); + return err; } static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -2117,6 +2176,9 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) azx_dev = get_azx_dev(substream); trace_azx_pcm_trigger(chip, azx_dev, cmd); + if (dsp_is_locked(azx_dev) || !azx_dev->prepared) + return -EPIPE; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: rstart = 1; @@ -2621,17 +2683,27 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format, struct azx_dev *azx_dev; int err; - if (snd_hda_lock_devices(bus)) - return -EBUSY; + azx_dev = azx_get_dsp_loader_dev(chip); + + dsp_lock(azx_dev); + spin_lock_irq(&chip->reg_lock); + if (azx_dev->running || azx_dev->locked) { + spin_unlock_irq(&chip->reg_lock); + err = -EBUSY; + goto unlock; + } + azx_dev->prepared = 0; + chip->saved_azx_dev = *azx_dev; + azx_dev->locked = 1; + spin_unlock_irq(&chip->reg_lock); err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), byte_size, bufp); if (err < 0) - goto unlock; + goto err_alloc; mark_pages_wc(chip, bufp, true); - azx_dev = azx_get_dsp_loader_dev(chip); azx_dev->bufsize = byte_size; azx_dev->period_bytes = byte_size; azx_dev->format_val = format; @@ -2649,13 +2721,20 @@ static int azx_load_dsp_prepare(struct hda_bus *bus, unsigned int format, goto error; azx_setup_controller(chip, azx_dev); + dsp_unlock(azx_dev); return azx_dev->stream_tag; error: mark_pages_wc(chip, bufp, false); snd_dma_free_pages(bufp); -unlock: - snd_hda_unlock_devices(bus); + err_alloc: + spin_lock_irq(&chip->reg_lock); + if (azx_dev->opened) + *azx_dev = chip->saved_azx_dev; + azx_dev->locked = 0; + spin_unlock_irq(&chip->reg_lock); + unlock: + dsp_unlock(azx_dev); return err; } @@ -2677,9 +2756,10 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, struct azx *chip = bus->private_data; struct azx_dev *azx_dev = azx_get_dsp_loader_dev(chip); - if (!dmab->area) + if (!dmab->area || !azx_dev->locked) return; + dsp_lock(azx_dev); /* reset BDL address */ azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); @@ -2692,7 +2772,12 @@ static void azx_load_dsp_cleanup(struct hda_bus *bus, snd_dma_free_pages(dmab); dmab->area = NULL; - snd_hda_unlock_devices(bus); + spin_lock_irq(&chip->reg_lock); + if (azx_dev->opened) + *azx_dev = chip->saved_azx_dev; + azx_dev->locked = 0; + spin_unlock_irq(&chip->reg_lock); + dsp_unlock(azx_dev); } #endif /* CONFIG_SND_HDA_DSP_LOADER */ @@ -3481,6 +3566,7 @@ static int azx_first_init(struct azx *chip) } for (i = 0; i < chip->num_streams; i++) { + dsp_lock_init(&chip->azx_dev[i]); /* allocate memory for the BDL for each stream */ err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci), -- cgit v1.2.3 From 55a63d4da3b8850480a1c5b222f77c739e30e346 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 21 Mar 2013 17:20:12 +0100 Subject: ALSA: hda - Fix DAC assignment for independent HP The generic parser should evaluate the availability of the independent HP when specified. Otherwise a DAC without the direct connection to the corresponding pin may be assigned for the HP, but the driver doesn't check it at all. The problem was actually seen on some machines with VT1708s or equivalent codec, where DAC0 is assigned to HP although it can be connected only via aamix. This patch adds the badness evaluation for the independent HP to make it working properly. Reported-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 46 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 78897d05d80f..43c2ea539561 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -995,6 +995,8 @@ enum { BAD_NO_EXTRA_SURR_DAC = 0x101, /* Primary DAC shared with main surrounds */ BAD_SHARED_SURROUND = 0x100, + /* No independent HP possible */ + BAD_NO_INDEP_HP = 0x40, /* Primary DAC shared with main CLFE */ BAD_SHARED_CLFE = 0x10, /* Primary DAC shared with extra surrounds */ @@ -1392,6 +1394,43 @@ static int check_aamix_out_path(struct hda_codec *codec, int path_idx) return snd_hda_get_path_idx(codec, path); } +/* check whether the independent HP is available with the current config */ +static bool indep_hp_possible(struct hda_codec *codec) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct nid_path *path; + int i, idx; + + if (cfg->line_out_type == AUTO_PIN_HP_OUT) + idx = spec->out_paths[0]; + else + idx = spec->hp_paths[0]; + path = snd_hda_get_path_from_idx(codec, idx); + if (!path) + return false; + + /* assume no path conflicts unless aamix is involved */ + if (!spec->mixer_nid || !is_nid_contained(path, spec->mixer_nid)) + return true; + + /* check whether output paths contain aamix */ + for (i = 0; i < cfg->line_outs; i++) { + if (spec->out_paths[i] == idx) + break; + path = snd_hda_get_path_from_idx(codec, spec->out_paths[i]); + if (path && is_nid_contained(path, spec->mixer_nid)) + return false; + } + for (i = 0; i < cfg->speaker_outs; i++) { + path = snd_hda_get_path_from_idx(codec, spec->speaker_paths[i]); + if (path && is_nid_contained(path, spec->mixer_nid)) + return false; + } + + return true; +} + /* fill the empty entries in the dac array for speaker/hp with the * shared dac pointed by the paths */ @@ -1545,6 +1584,9 @@ static int fill_and_eval_dacs(struct hda_codec *codec, badness += BAD_MULTI_IO; } + if (spec->indep_hp && !indep_hp_possible(codec)) + badness += BAD_NO_INDEP_HP; + /* re-fill the shared DAC for speaker / headphone */ if (cfg->line_out_type != AUTO_PIN_HP_OUT) refill_shared_dacs(codec, cfg->hp_outs, @@ -1758,6 +1800,10 @@ static int parse_output_paths(struct hda_codec *codec) cfg->speaker_pins, val); } + /* clear indep_hp flag if not available */ + if (spec->indep_hp && !indep_hp_possible(codec)) + spec->indep_hp = 0; + kfree(best_cfg); return 0; } -- cgit v1.2.3 From 417a1178f1bf3cdc606376b3ded3a22489fbb3eb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 15 Mar 2013 11:26:15 +0100 Subject: ASoC: dma-sh7760: Fix compile error The dma-sh7760 currently fails with the following compile error: sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe': sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *' This is due the misnaming of the snd_soc_platform_driver type name and 'ops' field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC Multi-Component Support"). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/sh/dma-sh7760.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 19eff8fc4fdd..1a8b03e4b41b 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -342,8 +342,8 @@ static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_platform sh7760_soc_platform = { - .pcm_ops = &camelot_pcm_ops, +static struct snd_soc_platform_driver sh7760_soc_platform = { + .ops = &camelot_pcm_ops, .pcm_new = camelot_pcm_new, .pcm_free = camelot_pcm_free, }; -- cgit v1.2.3 From 0eaa6cca1f75e12e4f5ec62cbe887330fe3b5fe9 Mon Sep 17 00:00:00 2001 From: Joonyoung Shim Date: Tue, 26 Mar 2013 14:41:05 +0900 Subject: ASoC: core: Fix to check return value of snd_soc_update_bits_locked() It can be 0 or 1 return value of snd_soc_update_bits_locked() when it is success. So just check return value is negative. Signed-off-by: Joonyoung Shim Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 507d251916af..ff4b45a5d796 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2963,7 +2963,7 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val = val << shift; ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); - if (ret != 0) + if (ret < 0) return ret; if (snd_soc_volsw_is_stereo(mc)) { -- cgit v1.2.3 From fa40ef208c955bfe21f53913f51f297ac3237e95 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 27 Mar 2013 16:39:01 +0000 Subject: ASoC: compress: Cancel delayed power down if needed When a new stream is being opened it is necessary to cancel any delayed power down of the audio. [Fixed unused variable -- broonie] Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b5b3db71e253..ed0bfb0ddb96 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -211,19 +211,27 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, if (platform->driver->compr_ops && platform->driver->compr_ops->set_params) { ret = platform->driver->compr_ops->set_params(cstream, params); if (ret < 0) - goto out; + goto err; } if (rtd->dai_link->compr_ops && rtd->dai_link->compr_ops->set_params) { ret = rtd->dai_link->compr_ops->set_params(cstream); if (ret < 0) - goto out; + goto err; } snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_START); -out: + /* cancel any delayed stream shutdown that is pending */ + rtd->pop_wait = 0; + mutex_unlock(&rtd->pcm_mutex); + + cancel_delayed_work_sync(&rtd->delayed_work); + + return ret; + +err: mutex_unlock(&rtd->pcm_mutex); return ret; } -- cgit v1.2.3 From 2ef5692efad330b67a234e2c49edad38538751e7 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 28 Mar 2013 05:20:22 -0400 Subject: ALSA: hda - bug fix on return value when getting HDMI ELD info In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0. Otherwise it will be returned uninitialized as non-zero after ELD info is got successfully. Thus hdmi_present_sense() will always assume ELD info is invalid by mistake, and /proc file system cannot show the proper ELD info. Signed-off-by: Mengdong Lin Cc: stable@vger.kernel.org Acked-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 7dd846380a50..d0d7ac1e99d2 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -320,7 +320,7 @@ int snd_hdmi_get_eld(struct hda_codec *codec, hda_nid_t nid, unsigned char *buf, int *eld_size) { int i; - int ret; + int ret = 0; int size; /* -- cgit v1.2.3 From 10250911c6f3531e9c2f4f1c6017782bc9bcd6d4 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Thu, 28 Mar 2013 05:21:28 -0400 Subject: ALSA: hda - bug fix on HDMI ELD debug message This patch let ELD debug message show 'pin_eld->monitor_present' which reflects the real pin response to verb GET_PIN_SENSE. 'eld->monitor_present' should not be used here because 'eld' is a temp structure now and so its "monitor_present" is not set. Signed-off-by: Mengdong Lin Acked-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827d0a95..de8ac5c07fd0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1196,7 +1196,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, eld->monitor_present, eld->eld_valid); + codec->addr, pin_nid, pin_eld->monitor_present, eld->eld_valid); if (eld->eld_valid) { if (snd_hdmi_get_eld(codec, pin_nid, eld->eld_buffer, -- cgit v1.2.3 From 690a863ff03d9a29ace2b752b8f802fba78a842f Mon Sep 17 00:00:00 2001 From: Torstein Hegge Date: Tue, 26 Mar 2013 22:10:05 +0100 Subject: ALSA: usb: Work around CM6631 sample rate change bug The C-Media CM6631 USB receiver doesn't respond to changes in sample rate while the interface is active. The same behavior is observed in other UAC2 hardware like the VIA VT1731. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all UAC2 devices, as it should not affect a standards compliant device, but it is only necessary for C-Media CM6631, VIA VT1731 and possibly others. Failure to read sample rate from the device is not handled as an error in set_sample_rate_v2(), as (permanent or intermittent) failure to read sample rate isn't essential for a successful sample rate set. Signed-off-by: Torstein Hegge Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 45 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 35 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 5e634a2eb282..9e2703a25156 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -253,7 +253,7 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, { struct usb_device *dev = chip->dev; unsigned char data[4]; - int err, crate; + int err, cur_rate, prev_rate; int clock = snd_usb_clock_find_source(chip, fmt->clock); if (clock < 0) @@ -266,6 +266,19 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return -ENXIO; } + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", + dev->devnum, iface, fmt->altsetting); + prev_rate = 0; + } else { + prev_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); + } + data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; @@ -280,19 +293,31 @@ static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface, return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - data, sizeof(data))) < 0) { + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + data, sizeof(data)); + if (err < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n", dev->devnum, iface, fmt->altsetting); - return err; + cur_rate = 0; + } else { + cur_rate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); } - crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24); - if (crate != rate) - snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate); + if (cur_rate != rate) { + snd_printd(KERN_WARNING + "current rate %d is different from the runtime rate %d\n", + cur_rate, rate); + } + + /* Some devices doesn't respond to sample rate changes while the + * interface is active. */ + if (rate != prev_rate) { + usb_set_interface(dev, iface, 0); + usb_set_interface(dev, iface, fmt->altsetting); + } return 0; } -- cgit v1.2.3 From a9b977ecd3dbc5d4f0fe0b3d5c66d284859b1f2a Mon Sep 17 00:00:00 2001 From: Prathyush K Date: Tue, 2 Apr 2013 16:53:01 +0530 Subject: ASoC: Samsung: return error if drvdata is not set This patch fixes a possible crash in case drvdata for the secondary device is not set. Signed-off-by: Prathyush K Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index d7231e336a7c..f1fc06419560 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1107,6 +1107,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) if (samsung_dai_type == TYPE_SEC) { sec_dai = dev_get_drvdata(&pdev->dev); + if (!sec_dai) { + dev_err(&pdev->dev, "Unable to get drvdata\n"); + return -EFAULT; + } snd_soc_register_dai(&sec_dai->pdev->dev, &sec_dai->i2s_dai_drv); asoc_dma_platform_register(&pdev->dev); -- cgit v1.2.3 From c6f9b1eb0e5df468891eff17f981b76c86f95f3a Mon Sep 17 00:00:00 2001 From: Prathyush K Date: Tue, 2 Apr 2013 16:53:02 +0530 Subject: ASoC: Samsung: set drvdata before adding secondary device Currently, a new platform device is created for secondary device by calling platform_device_register_resndata and then the drvdata is set for this device. The following patch has been added to driver core: "driver core: fix possible missing of device probe". This results in the added device getting probed immediately but the drvdata for the secondary device is not yet set. This patch removes the platform_device_register_resndata call and instead calls platform_device_alloc, platform_set_drvdata and platform_device_add which fixes the above issue. Signed-off-by: Prathyush K Signed-off-by: Padmavathi Venna Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index f1fc06419560..6bbeb0bf1a73 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -972,6 +972,7 @@ static const struct snd_soc_dai_ops samsung_i2s_dai_ops = { static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) { struct i2s_dai *i2s; + int ret; i2s = devm_kzalloc(&pdev->dev, sizeof(struct i2s_dai), GFP_KERNEL); if (i2s == NULL) @@ -996,15 +997,17 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.capture.channels_max = 2; i2s->i2s_dai_drv.capture.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.capture.formats = SAMSUNG_I2S_FMTS; + dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ - i2s->pdev = platform_device_register_resndata(NULL, - "samsung-i2s-sec", -1, NULL, 0, NULL, 0); + i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); if (IS_ERR(i2s->pdev)) return NULL; - } - /* Pre-assign snd_soc_dai_set_drvdata */ - dev_set_drvdata(&i2s->pdev->dev, i2s); + platform_set_drvdata(i2s->pdev, i2s); + ret = platform_device_add(i2s->pdev); + if (ret < 0) + return NULL; + } return i2s; } -- cgit v1.2.3 From 5aa995e83ac7727b7705431e6eb2b317c59b95ba Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 3 Apr 2013 11:00:01 +0200 Subject: ASoC: tegra: Don't claim to support PCM pause and resume The tegra dmaengine driver does not support pausing and resuming a DMA stream. The tegra PCM driver still claims to support pause and resume though and implements them by stopping and restarting the stream. This is not what an application using pause/resume would expect. Usually applications have support for working around PCMs which do not support suspend and resume, so don't set the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME flags for the tegra PCM and use the default snd_dmaengine_pcm_trigger callback. Signed-off-by: Lars-Peter Clausen Reviewed-by: Stephen Warren Tested-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_pcm.c | 24 +----------------------- 1 file changed, 1 insertion(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index c925ab0adeb6..5e2c55c5b255 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -43,8 +43,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED, .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels_min = 2, @@ -127,26 +125,6 @@ static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int tegra_pcm_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: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_START); - - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - return snd_dmaengine_pcm_trigger(substream, - SNDRV_PCM_TRIGGER_STOP); - default: - return -EINVAL; - } - return 0; -} - static int tegra_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { @@ -164,7 +142,7 @@ static struct snd_pcm_ops tegra_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = tegra_pcm_hw_params, .hw_free = tegra_pcm_hw_free, - .trigger = tegra_pcm_trigger, + .trigger = snd_dmaengine_pcm_trigger, .pointer = snd_dmaengine_pcm_pointer, .mmap = tegra_pcm_mmap, }; -- cgit v1.2.3 From 1d87caa69c04008e09f5ff47b5e6acb6116febc7 Mon Sep 17 00:00:00 2001 From: Rainer Koenig Date: Thu, 4 Apr 2013 08:40:38 +0200 Subject: ALSA: hda - Enabling Realtek ALC 671 codec * Added the device ID to the modalias list and assinged ALC662 patches for it * Added 4 port support for the device ID 0671 in alc662_parse_auto_config Signed-off-by: Rainer Koenig Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 563c24df4d6f..f15c36bde540 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3440,7 +3440,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec) const hda_nid_t *ssids; if (codec->vendor_id == 0x10ec0272 || codec->vendor_id == 0x10ec0663 || - codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670) + codec->vendor_id == 0x10ec0665 || codec->vendor_id == 0x10ec0670 || + codec->vendor_id == 0x10ec0671) ssids = alc663_ssids; else ssids = alc662_ssids; @@ -3894,6 +3895,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 }, { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 }, { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 }, + { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 }, { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 }, { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, -- cgit v1.2.3 From aeb3a97222832e5457c4b72d72235098ce4bfe8d Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 4 Apr 2013 11:47:13 +0200 Subject: ALSA: hda - fix typo in proc output Rename "Digitial In" to "Digital In". This function is only used for proc output, so should not cause any problems to change. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ecdf30eb5879..4aba7646dd9c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -173,7 +173,7 @@ const char *snd_hda_get_jack_type(u32 cfg) "Line Out", "Speaker", "HP Out", "CD", "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" + "SPDIF In", "Digital In", "Reserved", "Other" }; return jack_types[(cfg & AC_DEFCFG_DEVICE) -- cgit v1.2.3 From 8fc24426f15d967d585af7062b7be3c46bbce571 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Apr 2013 15:35:24 +0200 Subject: Revert "ALSA: hda - Allow power_save_controller option override DCAPS" This reverts commit 6ab317419c62850a71e2adfd1573e5ee87d8774f. The commit [6ab317419c: ALSA: hda - Allow power_save_controller option override DCAPS] changed the behavior of power_save_controller so that it can override the driver capability. This assumed that this option is rarely changed dynamically unlike power_save option. Too naive. It turned out that the user-space power-management tool tries to set power_save_controller option to 1 together with power_save option without knowing what's actually doing. This enabled forcibly the runtime PM of the controller, which is known to be broken om many chips thus disabled as default. So, the only sane fix is to revert this commit again. It was intended to ease debugging/testing for runtime PM enablement, but obviously we need another way for it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171 Reported-and-tested-by: Nikita Tsukanov Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 5 ++--- sound/pci/hda/hda_intel.c | 6 ++---- 2 files changed, 4 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4499bd948860..95731a08f257 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -890,9 +890,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. enable_msi - Enable Message Signaled Interrupt (MSI) (default = off) power_save - Automatic power-saving timeout (in second, 0 = disable) - power_save_controller - Support runtime D3 of HD-audio controller - (-1 = on for supported chip (default), false = off, - true = force to on even for unsupported hardware) + power_save_controller - Reset HD-audio controller in power-saving mode + (default = on) align_buffer_size - Force rounding of buffer/period sizes to multiples of 128 bytes. This is more efficient in terms of memory access but isn't required by the HDA spec and prevents diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 418bfc0eb0a3..bcd40ee488e3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -134,8 +134,8 @@ MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " * this may give more power-saving, but will take longer time to * wake up. */ -static int power_save_controller = -1; -module_param(power_save_controller, bint, 0644); +static bool power_save_controller = 1; +module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #endif /* CONFIG_PM */ @@ -2931,8 +2931,6 @@ static int azx_runtime_idle(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - if (power_save_controller > 0) - return 0; if (!power_save_controller || !(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) return -EBUSY; -- cgit v1.2.3 From 868211db6df96ddae411fcd800502725beef8387 Mon Sep 17 00:00:00 2001 From: Jiri Slaby Date: Thu, 4 Apr 2013 22:32:10 +0200 Subject: ALSA: hda/generic - fix uninitialized variable changed is not initialized in path_power_down_sync, but it is expected to be false in case no change happened in the loop. So set it to false. Signed-off-by: Jiri Slaby Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 43c2ea539561..2dbe767be16b 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -740,7 +740,7 @@ EXPORT_SYMBOL_HDA(snd_hda_activate_path); static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) { struct hda_gen_spec *spec = codec->spec; - bool changed; + bool changed = false; int i; if (!spec->power_down_unused || path->active) -- cgit v1.2.3 From 889d66848b12d891248b03abcb2a42047f8e172a Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Fri, 5 Apr 2013 20:49:46 +0200 Subject: ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_* The usb_control_msg() function expects __u16 types and performs the endianness conversions by itself. However, in three places, a conversion is performed before it is handed over to usb_control_msg(), which leads to a double conversion (= no conversion): * snd_usb_nativeinstruments_boot_quirk() * snd_nativeinstruments_control_get() * snd_nativeinstruments_control_put() Caught by sparse: sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types) sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types) sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types) sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value sound/usb/quirks.c:502:35: got restricted __le16 [usertype] Signed-off-by: Eldad Zack Acked-by: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 4 ++-- sound/usb/quirks.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 497d2741d119..ebe91440a068 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -509,7 +509,7 @@ static int snd_nativeinstruments_control_get(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_rcvctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0, cpu_to_le16(wIndex), + 0, wIndex, &tmp, sizeof(tmp), 1000); up_read(&mixer->chip->shutdown_rwsem); @@ -540,7 +540,7 @@ static int snd_nativeinstruments_control_put(struct snd_kcontrol *kcontrol, else ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), bRequest, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, - cpu_to_le16(wValue), cpu_to_le16(wIndex), + wValue, wIndex, NULL, 0, 1000); up_read(&mixer->chip->shutdown_rwsem); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5325a3869bb7..9c5ab22358b1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -486,7 +486,7 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, - cpu_to_le16(1), 0, NULL, 0, 1000); + 1, 0, NULL, 0, 1000); if (ret < 0) return ret; -- cgit v1.2.3 From f1ca493b0b5e8f42d3b2dc8877860db2983f47b6 Mon Sep 17 00:00:00 2001 From: Alban Bedel Date: Tue, 9 Apr 2013 17:13:59 +0200 Subject: ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is running The Charge Pump needs the DSP clock to work properly, without it the bypass to HP/LINEOUT is not working properly. This requirement is not mentioned in the datasheet but has been confirmed by Mark Brown from Wolfson. Signed-off-by: Alban Bedel Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8903.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 134e41c870b9..f8a31ad0b203 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1083,6 +1083,8 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "ROP", NULL, "Right Speaker PGA" }, { "RON", NULL, "Right Speaker PGA" }, + { "Charge Pump", NULL, "CLK_DSP" }, + { "Left Headphone Output PGA", NULL, "Charge Pump" }, { "Right Headphone Output PGA", NULL, "Charge Pump" }, { "Left Line Output PGA", NULL, "Charge Pump" }, -- cgit v1.2.3 From f6f629f8332ea70255f6c60c904270640a21a114 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Apr 2013 13:19:26 +0100 Subject: ASoC: wm5102: Correct lookup of arizona struct in SYSCLK event Reported-by: Ryo Tsutsui Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm5102.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b82bbf584146..34d0201d6a78 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -584,7 +584,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct arizona *arizona = dev_get_drvdata(codec->dev); + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct regmap *regmap = codec->control_data; const struct reg_default *patch = NULL; int i, patch_size; -- cgit v1.2.3 From 0fe09a45c4848b5b5607b968d959fdc1821c161d Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Fri, 19 Apr 2013 10:01:04 -0700 Subject: vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helper This is my example conversion of a few existing mmap users. The pcm mmap case is one of the more straightforward ones. Acked-by: Takashi Iwai Signed-off-by: Linus Torvalds --- sound/core/pcm_native.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 71ae86ca64ac..eb560fa32321 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3222,18 +3222,10 @@ EXPORT_SYMBOL_GPL(snd_pcm_lib_default_mmap); int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - long size; - unsigned long offset; + struct snd_pcm_runtime *runtime = substream->runtime;; area->vm_page_prot = pgprot_noncached(area->vm_page_prot); - area->vm_flags |= VM_IO; - size = area->vm_end - area->vm_start; - offset = area->vm_pgoff << PAGE_SHIFT; - if (io_remap_pfn_range(area, area->vm_start, - (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, - size, area->vm_page_prot)) - return -EAGAIN; - return 0; + return vm_iomap_memory(area, runtime->dma_addr, runtime->dma_bytes); } EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); -- cgit v1.2.3