March 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [asoc:for-next 32/53] sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type
by kbuild test robot 01 Mar '18

01 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 90d3c54575fc6488e912870c3d1a8b4af6a5a7b7 commit: bdc6f82ed1a68bdedea3c899e4ac35c88e02e63a [32/53] Merge remote-tracking branches 'asoc/topic/msm8916-wcd-digital', 'asoc/topic/nau8540', 'asoc/topic/nau8810' and 'asoc/topic/nau8824' into asoc-next config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout bdc6f82ed1a68bdedea3c899e4ac35c88e02e63a # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/intel/boards/cht_bsw_nau8824.c: In function 'cht_codec_init': >> sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type [-Werror=incompatible-pointer-types] nau8824_enable_jack_detect(codec, jack); ^~~~~ In file included from sound/soc/intel/boards/cht_bsw_nau8824.c:31:0: sound/soc/intel/boards/../../codecs/nau8824.h:474:5: note: expected 'struct snd_soc_component *' but argument is of type 'struct snd_soc_codec *' int nau8824_enable_jack_detect(struct snd_soc_component *component, ^~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/nau8824_enable_jack_detect +137 sound/soc/intel/boards/cht_bsw_nau8824.c 5b00ce63 John Hsu 2018-02-06 101 5b00ce63 John Hsu 2018-02-06 102 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) 5b00ce63 John Hsu 2018-02-06 103 { 5b00ce63 John Hsu 2018-02-06 104 struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); 5b00ce63 John Hsu 2018-02-06 105 struct snd_soc_jack *jack = &ctx->jack; 5b00ce63 John Hsu 2018-02-06 106 struct snd_soc_codec *codec = runtime->codec; 5b00ce63 John Hsu 2018-02-06 107 struct snd_soc_dai *codec_dai = runtime->codec_dai; 5b00ce63 John Hsu 2018-02-06 108 int ret, jack_type; 5b00ce63 John Hsu 2018-02-06 109 5b00ce63 John Hsu 2018-02-06 110 /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ 5b00ce63 John Hsu 2018-02-06 111 ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); 5b00ce63 John Hsu 2018-02-06 112 if (ret < 0) { 5b00ce63 John Hsu 2018-02-06 113 dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); 5b00ce63 John Hsu 2018-02-06 114 return ret; 5b00ce63 John Hsu 2018-02-06 115 } 5b00ce63 John Hsu 2018-02-06 116 5b00ce63 John Hsu 2018-02-06 117 /* NAU88L24 supports 4 butons headset detection 5b00ce63 John Hsu 2018-02-06 118 * KEY_MEDIA 5b00ce63 John Hsu 2018-02-06 119 * KEY_VOICECOMMAND 5b00ce63 John Hsu 2018-02-06 120 * KEY_VOLUMEUP 5b00ce63 John Hsu 2018-02-06 121 * KEY_VOLUMEDOWN 5b00ce63 John Hsu 2018-02-06 122 */ 5b00ce63 John Hsu 2018-02-06 123 jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | 5b00ce63 John Hsu 2018-02-06 124 SND_JACK_BTN_2 | SND_JACK_BTN_3; 5b00ce63 John Hsu 2018-02-06 125 ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, 5b00ce63 John Hsu 2018-02-06 126 cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); 5b00ce63 John Hsu 2018-02-06 127 if (ret) { 5b00ce63 John Hsu 2018-02-06 128 dev_err(runtime->dev, 5b00ce63 John Hsu 2018-02-06 129 "Headset Jack creation failed %d\n", ret); 5b00ce63 John Hsu 2018-02-06 130 return ret; 5b00ce63 John Hsu 2018-02-06 131 } 5b00ce63 John Hsu 2018-02-06 132 snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); 5b00ce63 John Hsu 2018-02-06 133 snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); 5b00ce63 John Hsu 2018-02-06 134 snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); 5b00ce63 John Hsu 2018-02-06 135 snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); 5b00ce63 John Hsu 2018-02-06 136 5b00ce63 John Hsu 2018-02-06 @137 nau8824_enable_jack_detect(codec, jack); 5b00ce63 John Hsu 2018-02-06 138 5b00ce63 John Hsu 2018-02-06 139 return ret; 5b00ce63 John Hsu 2018-02-06 140 } 5b00ce63 John Hsu 2018-02-06 141 :::::: The code at line 137 was first introduced by commit :::::: 5b00ce63e07fac692400db80d3e1fbd1f1f65701 ASoC: Intel: add nau8824 sound card :::::: TO: John Hsu <KCHSU0(a)nuvoton.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] Applied "ASoC: rt5514: Fix rt5514_spi_burst_read() buffer passing" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: rt5514: Fix rt5514_spi_burst_read() buffer passing has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 29838a5c1e77ed93d41fb9837390e243b9b73518 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven <geert(a)linux-m68k.org> Date: Tue, 30 Jan 2018 14:19:42 +0100 Subject: [PATCH] ASoC: rt5514: Fix rt5514_spi_burst_read() buffer passing MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit rt5514_dsp_voice_wake_up_put() passes "(u8 *)&buf" to rt5514_spi_burst_read(), where the cast is needed to silence a compiler warning: sound/soc/codecs/rt5514.c: In function ‘rt5514_dsp_voice_wake_up_put’: sound/soc/codecs/rt5514.c:357: warning: passing argument 2 of ‘rt5514_spi_burst_read’ from incompatible pointer type However, as the variable is declared as an array ("u8 buf[8]"), the function should just pass "buf" instead. Fixes: fc9cab05837639ce ("ASoC: rt5514: The DSP clock can be calibrated by the other clock source") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 2d79571098ac..b358704ee19c 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -353,8 +353,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, msleep(20); #if IS_ENABLED(CONFIG_SND_SOC_RT5514_SPI) rt5514_spi_burst_read(RT5514_PLL3_CALIB_CTRL6 | - RT5514_DSP_MAPPING, - (u8 *)&buf, sizeof(buf)); + RT5514_DSP_MAPPING, buf, sizeof(buf)); #else dev_err(component->dev, "There is no SPI driver for" " loading the firmware\n"); -- 2.16.2

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[alsa-devel] Applied "ASoC: rt5514: Fix uninitialized calibration value" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: rt5514: Fix uninitialized calibration value has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 890861aa85043cc9d99a01e1bd26883722e95243 Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven <geert(a)linux-m68k.org> Date: Tue, 30 Jan 2018 14:19:43 +0100 Subject: [PATCH] ASoC: rt5514: Fix uninitialized calibration value MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit With gcc-4.1.2, if CONFIG_SND_SOC_RT5514_SPI is not set: sound/soc/codecs/rt5514.c: In function ‘rt5514_dsp_voice_wake_up_put’: sound/soc/codecs/rt5514.c:363: warning: ‘buf[0]’ is used uninitialized in this function sound/soc/codecs/rt5514.c:363: warning: ‘buf[1]’ is used uninitialized in this function sound/soc/codecs/rt5514.c:363: warning: ‘buf[2]’ is used uninitialized in this function sound/soc/codecs/rt5514.c:363: warning: ‘buf[3]’ is used uninitialized in this function Fix this by initializing the buffer with zeroes, to avoid random bits being written to the calibration register later. Fixes: fc9cab05837639ce ("ASoC: rt5514: The DSP clock can be calibrated by the other clock source") Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5514.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index b358704ee19c..e8a66b03faab 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -357,6 +357,7 @@ static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, #else dev_err(component->dev, "There is no SPI driver for" " loading the firmware\n"); + memset(buf, 0, sizeof(buf)); #endif rt5514->pll3_cal_value = buf[0] | buf[1] << 8 | buf[2] << 16 | buf[3] << 24; -- 2.16.2

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[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake fixes
by Guneshwor Singh 01 Mar '18

01 Mar '18

This includes fixes for Skylake+ driver which are improving widget type check, fixes for simultaneous wake-on-voice (WoV) and capture use-case, fix to improve firmware download latency and lastly, fix to use uuid instead of module id for library modules. Guneshwor Singh (1): ASoC: Intel: Skylake: Refine skl widget type check Pawse, GuruprasadX (2): ASoC: Intel: Skylake: Add BE DAI for WoV usecase ASoC: Intel: board: Add BE DAI link for WoV and update DAPM machine map. Sanyog Kale (1): ASoC: Intel: Skylake: Disable clock and power gating during fw/lib download Sriram Periyasamy (1): ASoC: Intel: Skylake: Find module id from UUID for bind params sound/soc/intel/boards/bxt_rt298.c | 15 +++++ sound/soc/intel/skylake/skl-messages.c | 7 ++- sound/soc/intel/skylake/skl-pcm.c | 20 +++++- sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-topology.c | 107 +++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.h | 13 +++- sound/soc/intel/skylake/skl.c | 21 +++++++ sound/soc/intel/skylake/skl.h | 4 ++ 8 files changed, 176 insertions(+), 14 deletions(-) -- 2.16.1

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[alsa-devel] Applied "ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1310e737a68741c9b06cf63be0084bae6018b308 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 25 Feb 2018 11:46:55 +0100 Subject: [PATCH] ASoC: rt5651: Enable LDO and micbias1 supplies for jack-type detection To determine if a plugged in jack is a headset (speakers + mic) or headphones (mic contact shorted to ground) we use the micbias1 OVer Current Detect (OVCD) functionality. For this to work we need to have a micbias current to actually cause an overcurrent condition when headphones are plugged in, so jack-type detection requires both the LDO and micbias1 supplies to be on. Before this commit there were 2 issues with the handling of this: 1) The LDO supply was force-enabled twice and never disabled again even though it only needs to be forced on when doing jack-type detection 2) micbias1 was not force-enabled, and thus may be off when doing jack-type detection This commit fixes both by force-enabling the LDO and micbias1 supplies before checking for an overcurrent condition and disabling them afterwards. Note that both supplies will still get turned on normally (and OVCD will protect against overcurrent) when the micbias1 is enabled normally because the user has activated a sound stream recording from the mic. Tested-by: Carlo Caione <carlo(a)endlessm.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt5651.c | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index cbf9f56ecb14..be942886700d 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1561,6 +1561,28 @@ static int rt5651_set_bias_level(struct snd_soc_component *component, return 0; } +static void rt5651_enable_micbias1_for_ovcd(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "micbias1"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void rt5651_disable_micbias1_for_ovcd(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "micbias1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "LDO"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + static irqreturn_t rt5651_irq(int irq, void *data) { struct rt5651_priv *rt5651 = data; @@ -1574,7 +1596,6 @@ static irqreturn_t rt5651_irq(int irq, void *data) static int rt5651_set_jack(struct snd_soc_component *component, struct snd_soc_jack *hp_jack, void *data) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5651_priv *rt5651 = snd_soc_component_get_drvdata(component); int ret; @@ -1616,9 +1637,6 @@ static int rt5651_set_jack(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5651_PWR_ANLG2, RT5651_PWR_JD_M, RT5651_PWR_JD_M); - snd_soc_dapm_force_enable_pin(dapm, "LDO"); - snd_soc_dapm_sync(dapm); - snd_soc_component_update_bits(component, RT5651_MICBIAS, 0x38, 0x38); rt5651->hp_jack = hp_jack; @@ -1819,13 +1837,10 @@ static const struct dmi_system_id rt5651_quirk_table[] = { static int rt5651_jack_detect(struct snd_soc_component *component, int jack_insert) { - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); int jack_type; if (jack_insert) { - snd_soc_dapm_force_enable_pin(dapm, "LDO"); - snd_soc_dapm_sync(dapm); - + rt5651_enable_micbias1_for_ovcd(component); snd_soc_component_update_bits(component, RT5651_MICBIAS, RT5651_MIC1_OVCD_MASK | RT5651_MIC1_OVTH_MASK | @@ -1842,6 +1857,7 @@ static int rt5651_jack_detect(struct snd_soc_component *component, int jack_inse jack_type = SND_JACK_HEADSET; snd_soc_component_update_bits(component, RT5651_IRQ_CTRL2, RT5651_MB1_OC_CLR, 0); + rt5651_disable_micbias1_for_ovcd(component); } else { /* jack out */ jack_type = 0; -- 2.16.2

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[alsa-devel] xhci: Not enough bandwidth for new device state
by Waldemar Brodkorb 01 Mar '18

01 Mar '18

Hi, I am getting these messages from a Debian 9.3 system using Linux 4.9.x: [ 547.352746] usb 3-5.2: 3:1: usb_set_interface failed (-28) [ 548.352865] usb 3-5.3: Not enough bandwidth for new device state. [ 548.352868] usb 3-5.3: Not enough bandwidth for altsetting 1 I have connected a 7-Port USB 3.0 Hub to my machine. I have 6 different Headsets and USB Soundcards connected to the HUB. As soon as I start a C++ application which opens all ALSA devices I get these errors and the USB stopps working correctly. I think the maximum count of USB sound devices, which behave stable is three or four. Is this some miscalculation of the Linux kernel regarding the USB bandwidth usage? How I can determine how much bandwidth the USB sound card driver preallocates? Or are five USB soundcards really too much for an USB 3.0 bus? Thanks in advance for any advise, best regards Waldemar P.s.: I sent this message to LKML, but maybe it is more appropriate to ask here.

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[alsa-devel] Applied "ASoC: Intel: sst: Fix error-code check in sst_pause_stream()" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Fix error-code check in sst_pause_stream() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6f5062773747b3d3eb9f957030ac1be98e983ac2 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:38 +0100 Subject: [PATCH] ASoC: Intel: sst: Fix error-code check in sst_pause_stream() The value returned by sst_prepare_and_post_msg() is a negated SST_ERR_* value, so we must check for -SST_ERR_INVALID_STREAM_ID. Note that sst_pause_resume() already has the correct check. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index 7ee6aeb7e0af..b082b0922a7a 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -253,7 +253,7 @@ int sst_pause_stream(struct intel_sst_drv *sst_drv_ctx, int str_id) if (retval == 0) { str_info->prev = str_info->status; str_info->status = STREAM_PAUSED; - } else if (retval == SST_ERR_INVALID_STREAM_ID) { + } else if (retval == -SST_ERR_INVALID_STREAM_ID) { retval = -EINVAL; mutex_lock(&sst_drv_ctx->sst_lock); sst_clean_stream(str_info); -- 2.16.2

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[alsa-devel] Applied "ASoC: Intel: sst: Remove 2 unused members from stream_info struct" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove 2 unused members from stream_info struct has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bd47469c9985b50831a55bdda5a36f4672452764 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:39 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove 2 unused members from stream_info struct Remove the unused ops and str_id members from the stream_info struct. While at it also remove some kernel-doc comments for members which have already been removed in the past. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 7 ------- sound/soc/intel/atom/sst/sst_pvt.c | 1 - 2 files changed, 8 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index e02e2b4cc08f..1f3107909044 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -181,21 +181,15 @@ struct sst_block { * * @status : stream current state * @prev : stream prev state - * @ops : stream operation pb/cp/drm... - * @bufs: stream buffer list * @lock : stream mutex for protecting state * @pcm_substream : PCM substream * @period_elapsed : PCM period elapsed callback * @sfreq : stream sampling freq - * @str_type : stream type * @cumm_bytes : cummulative bytes decoded - * @str_type : stream type - * @src : stream source */ struct stream_info { unsigned int status; unsigned int prev; - unsigned int ops; struct mutex lock; void *pcm_substream; @@ -212,7 +206,6 @@ struct stream_info { unsigned int num_ch; unsigned int pipe_id; - unsigned int str_id; unsigned int task_id; }; diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index b1e6b8f34a6a..d7d38b0f68db 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -365,7 +365,6 @@ void sst_init_stream(struct stream_info *stream, { stream->status = STREAM_INIT; stream->prev = STREAM_UN_INIT; - stream->ops = ops; } int sst_validate_strid( -- 2.16.2

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[alsa-devel] Applied "ASoC: Intel: sst: Remove unnecessary sst_init_stream() function" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove unnecessary sst_init_stream() function has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8cf732bbd8d6bc9a28ac48c7bac0ccc9f55cb151 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:40 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove unnecessary sst_init_stream() function sst_init_stream() has only one caller and all its function arguments are unused. Inline it on the one call site and remove it. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 2 -- sound/soc/intel/atom/sst/sst_pvt.c | 7 ------- sound/soc/intel/atom/sst/sst_stream.c | 5 ++--- 3 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 1f3107909044..164b0f674c20 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -494,8 +494,6 @@ int sst_prepare_and_post_msg(struct intel_sst_drv *sst, void sst_process_pending_msg(struct work_struct *work); int sst_assign_pvt_id(struct intel_sst_drv *sst_drv_ctx); -void sst_init_stream(struct stream_info *stream, - int codec, int sst_id, int ops, u8 slot); int sst_validate_strid(struct intel_sst_drv *sst_drv_ctx, int str_id); struct stream_info *get_stream_info(struct intel_sst_drv *sst_drv_ctx, int str_id); diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index d7d38b0f68db..af93244b4868 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -360,13 +360,6 @@ int sst_assign_pvt_id(struct intel_sst_drv *drv) return local; } -void sst_init_stream(struct stream_info *stream, - int codec, int sst_id, int ops, u8 slot) -{ - stream->status = STREAM_INIT; - stream->prev = STREAM_UN_INIT; -} - int sst_validate_strid( struct intel_sst_drv *sst_drv_ctx, int str_id) { diff --git a/sound/soc/intel/atom/sst/sst_stream.c b/sound/soc/intel/atom/sst/sst_stream.c index b082b0922a7a..7cf5dd1993bc 100644 --- a/sound/soc/intel/atom/sst/sst_stream.c +++ b/sound/soc/intel/atom/sst/sst_stream.c @@ -83,6 +83,8 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) pipe_id = str_params->device_type; task_id = str_params->task; + sst_drv_ctx->streams[str_id].status = STREAM_INIT; + sst_drv_ctx->streams[str_id].prev = STREAM_UN_INIT; sst_drv_ctx->streams[str_id].pipe_id = pipe_id; sst_drv_ctx->streams[str_id].task_id = task_id; sst_drv_ctx->streams[str_id].num_ch = num_ch; @@ -100,9 +102,6 @@ int sst_alloc_stream_mrfld(struct intel_sst_drv *sst_drv_ctx, void *params) pipe_id, task_id); /* allocate device type context */ - sst_init_stream(&sst_drv_ctx->streams[str_id], alloc_param.codec_type, - str_id, alloc_param.operation, 0); - dev_dbg(sst_drv_ctx->dev, "Alloc for str %d pipe %#x\n", str_id, pipe_id); ret = sst_prepare_and_post_msg(sst_drv_ctx, task_id, IPC_CMD, -- 2.16.2

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[alsa-devel] Applied "ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states" to the asoc tree
by Mark Brown 01 Mar '18

01 Mar '18

The patch ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From de9833476de7eb1d4b8bdc599cbbd616fb0e1a28 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:41 +0100 Subject: [PATCH] ASoC: Intel: sst: Remove unused STREAM_DECODE and STREAM_RESET states STREAM_DECODE is completely unused, status == STREAM_RESET was checked for, but never set, remove both. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst.h | 4 +--- sound/soc/intel/atom/sst/sst_drv_interface.c | 19 ------------------- 2 files changed, 1 insertion(+), 22 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.h b/sound/soc/intel/atom/sst/sst.h index 164b0f674c20..f4fd442080b2 100644 --- a/sound/soc/intel/atom/sst/sst.h +++ b/sound/soc/intel/atom/sst/sst.h @@ -65,9 +65,7 @@ enum sst_stream_states { STREAM_UN_INIT = 0, /* Freed/Not used stream */ STREAM_RUNNING = 1, /* Running */ STREAM_PAUSED = 2, /* Paused stream */ - STREAM_DECODE = 3, /* stream is in decoding only state */ - STREAM_INIT = 4, /* stream init, waiting for data */ - STREAM_RESET = 5, /* force reset on recovery */ + STREAM_INIT = 3, /* stream init, waiting for data */ }; enum sst_ram_type { diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 71af5449be90..6a8b253c58d2 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -238,16 +238,7 @@ static int sst_cdev_close(struct device *dev, unsigned int str_id) return -EINVAL; } - if (stream->status == STREAM_RESET) { - dev_dbg(dev, "stream in reset state...\n"); - stream->status = STREAM_UN_INIT; - - retval = 0; - goto put; - } - retval = sst_free_stream(ctx, str_id); -put: stream->compr_cb_param = NULL; stream->compr_cb = NULL; @@ -256,7 +247,6 @@ static int sst_cdev_close(struct device *dev, unsigned int str_id) dev_dbg(dev, "End\n"); return retval; - } static int sst_cdev_ack(struct device *dev, unsigned int str_id, @@ -486,16 +476,7 @@ static int sst_close_pcm_stream(struct device *dev, unsigned int str_id) return -EINVAL; } - if (stream->status == STREAM_RESET) { - /* silently fail here as we have cleaned the stream earlier */ - dev_dbg(ctx->dev, "stream in reset state...\n"); - - retval = 0; - goto put; - } - retval = free_stream_context(ctx, str_id); -put: stream->pcm_substream = NULL; stream->status = STREAM_UN_INIT; stream->period_elapsed = NULL; -- 2.16.2

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