November 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/cs8409: Schedule delayed work for jack detect on resume
by Vitaly Rodionov 25 Nov '21

25 Nov '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> CS42L42 runs jack detect on resume, however this requires unsol events, and unsol events are ignored whilst the power state is not set to ON. The power state is set to ON only after the resume finishes. Schedule a delayed work timer to run jack detect after the resume call finishes. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409.c | 79 +++++++++++++++++++++++++++++------- sound/pci/hda/patch_cs8409.h | 1 + 2 files changed, 65 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 31ff11ab868e..88213e95f0b3 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -634,6 +634,30 @@ static void cs42l42_run_jack_detect(struct sub_codec *cs42l42) cs8409_i2c_write(cs42l42, 0x1120, 0xc0); } +static void cs42l42_run_jack_detect_all(struct hda_codec *codec) +{ + struct cs8409_spec *spec = codec->spec; + struct sub_codec *cs42l42; + int i; + + for (i = 0; i < spec->num_scodecs; i++) { + cs42l42 = spec->scodecs[i]; + cs42l42_enable_jack_detect(cs42l42); + if (!cs42l42->hp_jack_in) + cs42l42_run_jack_detect(cs42l42); + } +} + +/* + * cs42l42_jack_detect_worker - Worker that retries jack detect + */ +static void cs42l42_jack_detect_worker(struct work_struct *work) +{ + struct cs8409_spec *spec = container_of(work, struct cs8409_spec, jack_detect_work.work); + + cs42l42_run_jack_detect_all(spec->codec); +} + static int cs42l42_handle_tip_sense(struct sub_codec *cs42l42, unsigned int reg_ts_status) { int status_changed = 0; @@ -749,8 +773,6 @@ static void cs42l42_resume(struct sub_codec *cs42l42) if (cs42l42->full_scale_vol) cs8409_i2c_write(cs42l42, 0x2001, 0x01); - - cs42l42_enable_jack_detect(cs42l42); } #ifdef CONFIG_PM @@ -800,6 +822,7 @@ static void cs8409_free(struct hda_codec *codec) /* Cancel i2c clock disable timer, and disable clock if left enabled */ cancel_delayed_work_sync(&spec->i2c_clk_work); + cancel_delayed_work_sync(&spec->jack_detect_work); cs8409_disable_i2c_clock(codec); snd_hda_gen_free(codec); @@ -863,6 +886,7 @@ static int cs8409_cs42l42_suspend(struct hda_codec *codec) /* Cancel i2c clock disable timer, and disable clock if left enabled */ cancel_delayed_work_sync(&spec->i2c_clk_work); + cancel_delayed_work_sync(&spec->jack_detect_work); cs8409_disable_i2c_clock(codec); snd_hda_shutup_pins(codec); @@ -970,6 +994,8 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, spec->scodecs[CS8409_CODEC0]->codec = codec; codec->patch_ops = cs8409_cs42l42_patch_ops; + INIT_DELAYED_WORK(&spec->jack_detect_work, cs42l42_jack_detect_worker); + spec->gen.suppress_auto_mute = 1; spec->gen.no_primary_hp = 1; spec->gen.suppress_vmaster = 1; @@ -1029,9 +1055,16 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, case HDA_FIXUP_ACT_INIT: cs8409_cs42l42_hw_init(codec); spec->init_done = 1; - if (spec->init_done && spec->build_ctrl_done - && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + if (spec->init_done && spec->build_ctrl_done) { + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); + } + } break; case HDA_FIXUP_ACT_BUILD: spec->build_ctrl_done = 1; @@ -1040,9 +1073,16 @@ void cs8409_cs42l42_fixups(struct hda_codec *codec, const struct hda_fixup *fix, * been already plugged in. * Run immediately after init. */ - if (spec->init_done && spec->build_ctrl_done - && !spec->scodecs[CS8409_CODEC0]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[CS8409_CODEC0]); + if (spec->init_done && spec->build_ctrl_done) { + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); + } + } break; default: break; @@ -1178,7 +1218,6 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac { struct cs8409_spec *spec = codec->spec; struct snd_kcontrol_new *kctrl; - int i; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: @@ -1193,6 +1232,8 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac spec->scodecs[CS8409_CODEC1]->codec = codec; spec->num_scodecs = 2; + INIT_DELAYED_WORK(&spec->jack_detect_work, cs42l42_jack_detect_worker); + codec->patch_ops = cs8409_dolphin_patch_ops; /* GPIO 1,5 out, 0,4 in */ @@ -1237,9 +1278,13 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac dolphin_hw_init(codec); spec->init_done = 1; if (spec->init_done && spec->build_ctrl_done) { - for (i = 0; i < spec->num_scodecs; i++) { - if (!spec->scodecs[i]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[i]); + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); } } break; @@ -1251,9 +1296,13 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac * Run immediately after init. */ if (spec->init_done && spec->build_ctrl_done) { - for (i = 0; i < spec->num_scodecs; i++) { - if (!spec->scodecs[i]->hp_jack_in) - cs42l42_run_jack_detect(spec->scodecs[i]); + /* No point in running jack detect until we have fully resumed */ + if (codec->core.dev.power.power_state.event != PM_EVENT_ON) { + codec_warn(codec, "Not ready to detect jack, deferring...\n"); + schedule_delayed_work(&spec->jack_detect_work, msecs_to_jiffies(25)); + return; + } else { + cs42l42_run_jack_detect_all(codec); } } break; diff --git a/sound/pci/hda/patch_cs8409.h b/sound/pci/hda/patch_cs8409.h index ade2b838590c..632d3ec8322d 100644 --- a/sound/pci/hda/patch_cs8409.h +++ b/sound/pci/hda/patch_cs8409.h @@ -330,6 +330,7 @@ struct cs8409_spec { unsigned int i2c_clck_enabled; unsigned int dev_addr; struct delayed_work i2c_clk_work; + struct delayed_work jack_detect_work; unsigned int playback_started:1; unsigned int capture_started:1; -- 2.25.1

2 1

[PATCH] dt-bindings: sound: nvidia, tegra-audio: Convert multiple txt bindings to yaml
by David Heidelberg 25 Nov '21

25 Nov '21

Convert Tegra audio complex with the * ALC5632 * MAX98090 * RT5640 * RT5677 * SGTL5000 * TrimSlice * WM8753 * WM8903 * WM9712 codec to the YAML format. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- .../sound/nvidia,tegra-audio-alc5632.txt | 48 ----- .../sound/nvidia,tegra-audio-max98090.txt | 53 ----- .../sound/nvidia,tegra-audio-rt5640.txt | 52 ----- .../sound/nvidia,tegra-audio-rt5677.txt | 67 ------ .../sound/nvidia,tegra-audio-sgtl5000.txt | 42 ---- .../sound/nvidia,tegra-audio-trimslice.txt | 21 -- .../sound/nvidia,tegra-audio-wm8753.txt | 40 ---- .../sound/nvidia,tegra-audio-wm8903.txt | 62 ------ .../sound/nvidia,tegra-audio-wm9712.txt | 60 ------ .../bindings/sound/nvidia,tegra-audio.yaml | 190 ++++++++++++++++++ 10 files changed, 190 insertions(+), 445 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt delete mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt deleted file mode 100644 index 57f40f93453e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-alc5632.txt +++ /dev/null @@ -1,48 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-alc5632" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the ALC5632's pins as documented in the binding for the device - and: - - * Headset Stereophone - * Int Spk - * Headset Mic - * Digital Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller -- nvidia,audio-codec : The phandle of the ALC5632 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-alc5632-paz00", - "nvidia,tegra-audio-alc5632"; - - nvidia,model = "Compal PAZ00"; - - nvidia,audio-routing = - "Int Spk", "SPK_OUTP", - "Int Spk", "SPK_OUTN", - "Headset Mic","MICBIAS1", - "MIC1_N", "Headset Mic", - "MIC1_P", "Headset Mic", - "Headset Stereophone", "HP_OUT_R", - "Headset Stereophone", "HP_OUT_L"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&alc5632>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt deleted file mode 100644 index c3495beba358..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ /dev/null @@ -1,53 +0,0 @@ -NVIDIA Tegra audio complex, with MAX98090 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-max98090" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the MAX98090's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - * Int Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the MAX98090 audio codec. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-max98090-venice2", - "nvidia,tegra-audio-max98090"; - nvidia,model = "NVIDIA Tegra Venice2"; - - nvidia,audio-routing = - "Headphones", "HPR", - "Headphones", "HPL", - "Speakers", "SPKR", - "Speakers", "SPKL", - "Mic Jack", "MICBIAS", - "IN34", "Mic Jack"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&acodec>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt deleted file mode 100644 index 7788808dcd0b..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5640.txt +++ /dev/null @@ -1,52 +0,0 @@ -NVIDIA Tegra audio complex, with RT5640 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5640" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the RT5640's pins (as documented in its binding), and the jacks - on the board: - - * Headphones - * Speakers - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5640 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5640-dalmore", - "nvidia,tegra-audio-rt5640"; - nvidia,model = "NVIDIA Tegra Dalmore"; - - nvidia,audio-routing = - "Headphones", "HPOR", - "Headphones", "HPOL", - "Speakers", "SPORP", - "Speakers", "SPORN", - "Speakers", "SPOLP", - "Speakers", "SPOLN"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5640>; - - nvidia,hp-det-gpios = <&gpio 143 0>; /* GPIO PR7 */ - - clocks = <&tegra_car 216>, <&tegra_car 217>, <&tegra_car 120>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt deleted file mode 100644 index a4589cda214e..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt +++ /dev/null @@ -1,67 +0,0 @@ -NVIDIA Tegra audio complex, with RT5677 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-rt5677" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the RT5677's pins (as documented in its binding), and the jacks - on the board: - - * Headphone - * Speaker - * Headset Mic - * Internal Mic 1 - * Internal Mic 2 - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the RT5677 audio codec. This binding - assumes that AIF1 on the CODEC is connected to Tegra. - -Optional properties: -- nvidia,hp-det-gpios : The GPIO that detects headphones are plugged in -- nvidia,hp-en-gpios : The GPIO that enables headphone amplifier -- nvidia,mic-present-gpios: The GPIO that mic jack is plugged in -- nvidia,dmic-clk-en-gpios : The GPIO that gates DMIC clock signal - -Example: - -sound { - compatible = "nvidia,tegra-audio-rt5677-ryu", - "nvidia,tegra-audio-rt5677"; - nvidia,model = "NVIDIA Tegra Ryu"; - - nvidia,audio-routing = - "Headphone", "LOUT2", - "Headphone", "LOUT1", - "Headset Mic", "MICBIAS1", - "IN1P", "Headset Mic", - "IN1N", "Headset Mic", - "DMIC L1", "Internal Mic 1", - "DMIC R1", "Internal Mic 1", - "DMIC L2", "Internal Mic 2", - "DMIC R2", "Internal Mic 2", - "Speaker", "PDM1L", - "Speaker", "PDM1R"; - - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&rt5677>; - - nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; - nvidia,mic-present-gpios = <&gpio TEGRA_GPIO(O, 5) GPIO_ACTIVE_LOW>; - nvidia,hp-en-gpios = <&rt5677 1 GPIO_ACTIVE_HIGH>; - nvidia,dmic-clk-en-gpios = <&rt5677 2 GPIO_ACTIVE_HIGH>; - - clocks = <&tegra_car TEGRA124_CLK_PLL_A>, - <&tegra_car TEGRA124_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA124_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt deleted file mode 100644 index 5da7da4ea07a..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt +++ /dev/null @@ -1,42 +0,0 @@ -NVIDIA Tegra audio complex, with SGTL5000 CODEC - -Required properties: -- compatible : "nvidia,tegra-audio-sgtl5000" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the SGTL5000's pins (as documented in its binding), and the jacks - on the board: - - * Headphone Jack - * Line In Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's - connected to the CODEC. -- nvidia,audio-codec : The phandle of the SGTL5000 audio codec. - -Example: - -sound { - compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", - "nvidia,tegra-audio-sgtl5000"; - nvidia,model = "Toradex Apalis T30"; - nvidia,audio-routing = - "Headphone Jack", "HP_OUT", - "LINE_IN", "Line In Jack", - "MIC_IN", "Mic Jack"; - nvidia,i2s-controller = <&tegra_i2s2>; - nvidia,audio-codec = <&sgtl5000>; - clocks = <&tegra_car TEGRA30_CLK_PLL_A>, - <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, - <&tegra_car TEGRA30_CLK_EXTERN1>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt deleted file mode 100644 index ef1fe7358279..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-trimslice.txt +++ /dev/null @@ -1,21 +0,0 @@ -NVIDIA Tegra audio complex for TrimSlice - -Required properties: -- compatible : "nvidia,tegra-audio-trimslice" -- clocks : Must contain an entry for each entry in clock-names. -- clock-names : Must include the following entries: - "pll_a" (The Tegra clock of that name), - "pll_a_out0" (The Tegra clock of that name), - "mclk" (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Example: - -sound { - compatible = "nvidia,tegra-audio-trimslice"; - nvidia,i2s-controller = <&tegra_i2s1>; - nvidia,audio-codec = <&codec>; - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt deleted file mode 100644 index 96f6a57dd6b4..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8753.txt +++ /dev/null @@ -1,40 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8753" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the WM8753's pins as documented in the binding for the WM8753, - and the jacks on the board: - - * Headphone Jack - * Mic Jack - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8753 audio codec -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8753-whistler", - "nvidia,tegra-audio-wm8753" - nvidia,model = "tegra-wm8753-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "LOUT1", - "Headphone Jack", "ROUT1"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8753>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt deleted file mode 100644 index bbd581a8c5bc..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm8903.txt +++ /dev/null @@ -1,62 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm8903" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the WM8903's pins (documented in the WM8903 binding document), - and the jacks on the board: - - * Headphone Jack - * Int Spk - * Mic Jack - * Int Mic - -- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller -- nvidia,audio-codec : The phandle of the WM8903 audio codec - -Optional properties: -- nvidia,spkr-en-gpios : The GPIO that enables the speakers -- nvidia,hp-mute-gpios : The GPIO that mutes the headphones -- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in -- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone -- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone -- nvidia,headset : The Mic Jack represents state of the headset microphone pin - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm8903-harmony", - "nvidia,tegra-audio-wm8903" - nvidia,model = "tegra-wm8903-harmony"; - - nvidia,audio-routing = - "Headphone Jack", "HPOUTR", - "Headphone Jack", "HPOUTL", - "Int Spk", "ROP", - "Int Spk", "RON", - "Int Spk", "LOP", - "Int Spk", "LON", - "Mic Jack", "MICBIAS", - "IN1L", "Mic Jack"; - - nvidia,i2s-controller = <&i2s1>; - nvidia,audio-codec = <&wm8903>; - - nvidia,spkr-en-gpios = <&codec 2 0>; - nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */ - nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */ - nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */ - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; - diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt deleted file mode 100644 index 436f6cd9d07c..000000000000 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-wm9712.txt +++ /dev/null @@ -1,60 +0,0 @@ -NVIDIA Tegra audio complex - -Required properties: -- compatible : "nvidia,tegra-audio-wm9712" -- clocks : Must contain an entry for each entry in clock-names. - See ../clocks/clock-bindings.txt for details. -- clock-names : Must include the following entries: - - pll_a - - pll_a_out0 - - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk) -- nvidia,model : The user-visible name of this sound complex. -- nvidia,audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the connection's sink, - the second being the connection's source. Valid names for sources and - sinks are the WM9712's pins, and the jacks on the board: - - WM9712 pins: - - * MONOOUT - * HPOUTL - * HPOUTR - * LOUT2 - * ROUT2 - * OUT3 - * LINEINL - * LINEINR - * PHONE - * PCBEEP - * MIC1 - * MIC2 - * Mic Bias - - Board connectors: - - * Headphone - * LineIn - * Mic - -- nvidia,ac97-controller : The phandle of the Tegra AC97 controller - - -Example: - -sound { - compatible = "nvidia,tegra-audio-wm9712-colibri_t20", - "nvidia,tegra-audio-wm9712"; - nvidia,model = "Toradex Colibri T20"; - - nvidia,audio-routing = - "Headphone", "HPOUTL", - "Headphone", "HPOUTR", - "LineIn", "LINEINL", - "LineIn", "LINEINR", - "Mic", "MIC1"; - - nvidia,ac97-controller = <&ac97>; - - clocks = <&tegra_car 112>, <&tegra_car 113>, <&tegra_car 93>; - clock-names = "pll_a", "pll_a_out0", "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml new file mode 100644 index 000000000000..6b8876bfa40f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio.yaml @@ -0,0 +1,190 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/nvidia,tegra-audio.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: NVIDIA Tegra audio complex + +maintainers: + - Jon Hunter <jonathanh(a)nvidia.com> + - Thierry Reding <thierry.reding(a)gmail.com> + +properties: + compatible: + oneOf: + - items: + - pattern: '^[a-z0-9]+,tegra-audio-alc5632(-[a-z0-9])+' + - const: nvidia,tegra-audio-alc5632 + - items: + - enum: + - nvidia,tegra-audio-max98090-nyan-big + - nvidia,tegra-audio-max98090-nyan-blaze + - const: nvidia,tegra-audio-max98090-nyan + - const: nvidia,tegra-audio-max98090 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-max98090(-[a-z0-9])+' + - const: nvidia,tegra-audio-max98090 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-rt56(39|40)(-[a-z0-9])+' + - const: nvidia,tegra-audio-rt5640 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-rt5677(-[a-z0-9])+' + - const: nvidia,tegra-audio-rt5677 + - items: + - enum: + - toradex,tegra-audio-sgtl5000-apalis_t30 + - toradex,tegra-audio-sgtl5000-colibri_t30 + - toradex,tegra-audio-sgtl5000-apalis_tk1 + - const: nvidia,tegra-audio-sgtl5000 + - const: nvidia,tegra-audio-trimslice + - items: + - pattern: '^[a-z0-9]+,tegra-audio-wm8753(-[a-z0-9])+' + - const: nvidia,tegra-audio-wm8753 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-(plutux|wm8903(-[a-z0-9])+)' + - const: nvidia,tegra-audio-wm8903 + - items: + - pattern: '^[a-z0-9]+,tegra-audio-wm9712(-[a-z0-9])+' + - const: nvidia,tegra-audio-wm9712 + + clocks: + items: + - description: PLL A clock + - description: PLL A OUT0 clock + - description: The Tegra cdev1/extern1 clock, which feeds the card's mclk + + clock-names: + items: + - const: pll_a + - const: pll_a_out0 + - const: mclk + + assigned-clocks: true + + assigned-clock-parents: true + + assigned-clock-rates: true + + nvidia,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + nvidia,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the pins (documented in the binding document), + and the jacks on the board: + ALC5632: + * Headset Stereophone + * Int Spk + * Headset Mic + * Digital Mic + MAXX98090: + * Headphones + * SPeakers + * Mic Jack + * Int Mic + RT5640: + * Headphones + * Speakers + * Mic Jack + RT5677: + * Headphone + * Speaker + * Headset Mic + * Internal Mic 1 + * Internal Mic 2 + SGTL5000: + * Headphone Jack + * Line In Jack + * Mic Jack + WM8753: + * Headphone Jack + * Mic Jack + WM8903: + * Headphone Jack + * Int Spk + * Mic Jack + * Int Mic + WM9712: + * Headphone + * LineIn + * Mic + + nvidia,ac97-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the AC97 controller + + nvidia,i2s-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the Tegra I2S controller + + nvidia,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of audio codec + + nvidia,spkr-en-gpios: + maxItems: 1 + description: The GPIO that enables the speakers + + nvidia,hp-mute-gpios: + maxItems: 1 + description: The GPIO that mutes the headphones + + nvidia,hp-det-gpios: + maxItems: 1 + description: The GPIO that detect headphones are plugged in + + nvidia,mic-det-gpios: + maxItems: 1 + description: The GPIO that detect microphone is plugged in + + nvidia,int-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the internal microphone + + nvidia,ext-mic-en-gpios: + maxItems: 1 + description: The GPIO that enables the external microphone + + nvidia,headset: + type: boolean + description: The Mic Jack represents state of the headset microphone pin + +additionalProperties: false + +required: + - compatible + - clocks + - clock-names + +oneOf: + - required: + - nvidia,audio-codec + - nvidia,i2s-controller + - required: + - nvidia,ac97-controller + +examples: + - | + #include <dt-bindings/clock/tegra30-car.h> + + sound { + compatible = "toradex,tegra-audio-sgtl5000-apalis_t30", + "nvidia,tegra-audio-sgtl5000"; + nvidia,model = "Toradex Apalis T30 SGTL5000"; + nvidia,audio-routing = + "Headphone Jack", "HP_OUT", + "LINE_IN", "Line In Jack", + "MIC_IN", "Mic Jack"; + nvidia,i2s-controller = <&tegra_i2s2>; + nvidia,audio-codec = <&codec>; + clocks = <&tegra_car TEGRA30_CLK_PLL_A>, + <&tegra_car TEGRA30_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA30_CLK_EXTERN1>; + clock-names = "pll_a", "pll_a_out0", "mclk"; + }; -- 2.33.0

3 5

[PATCH v2] ASoC: rockchip: i2s_tdm: Dup static DAI template
by Nicolas Frattaroli 24 Nov '21

24 Nov '21

Previously, the DAI template was used directly, which lead to fun bugs such as "why is my channels_max changing?" when one instantiated more than one i2s_tdm IP block in a device tree. This change makes it so that we instead duplicate the template struct, and then use that. Fixes: 081068fd6414 ("ASoC: rockchip: add support for i2s-tdm controller") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s_tdm.c | 38 ++++++++++++++++----------- 1 file changed, 22 insertions(+), 16 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 17b9b287853a..4328c10ea830 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1312,17 +1312,12 @@ static const struct of_device_id rockchip_i2s_tdm_match[] = { static struct snd_soc_dai_driver i2s_tdm_dai = { .probe = rockchip_i2s_tdm_dai_probe, - .playback = { - .stream_name = "Playback", - }, - .capture = { - .stream_name = "Capture", - }, .ops = &rockchip_i2s_tdm_dai_ops, }; -static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) +static int rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) { + struct snd_soc_dai_driver *dai; struct property *dma_names; const char *dma_name; u64 formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | @@ -1337,19 +1332,28 @@ static void rockchip_i2s_tdm_init_dai(struct rk_i2s_tdm_dev *i2s_tdm) i2s_tdm->has_capture = true; } + dai = devm_kmemdup(i2s_tdm->dev, &i2s_tdm_dai, + sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + if (i2s_tdm->has_playback) { - i2s_tdm_dai.playback.channels_min = 2; - i2s_tdm_dai.playback.channels_max = 8; - i2s_tdm_dai.playback.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.playback.formats = formats; + dai->playback.stream_name = "Playback"; + dai->playback.channels_min = 2; + dai->playback.channels_max = 8; + dai->playback.rates = SNDRV_PCM_RATE_8000_192000; + dai->playback.formats = formats; } if (i2s_tdm->has_capture) { - i2s_tdm_dai.capture.channels_min = 2; - i2s_tdm_dai.capture.channels_max = 8; - i2s_tdm_dai.capture.rates = SNDRV_PCM_RATE_8000_192000; - i2s_tdm_dai.capture.formats = formats; + dai->capture.stream_name = "Capture"; + dai->capture.channels_min = 2; + dai->capture.channels_max = 8; + dai->capture.rates = SNDRV_PCM_RATE_8000_192000; + dai->capture.formats = formats; } + + return 0; } static int rockchip_i2s_tdm_path_check(struct rk_i2s_tdm_dev *i2s_tdm, @@ -1541,7 +1545,9 @@ static int rockchip_i2s_tdm_probe(struct platform_device *pdev) spin_lock_init(&i2s_tdm->lock); i2s_tdm->soc_data = (struct rk_i2s_soc_data *)of_id->data; - rockchip_i2s_tdm_init_dai(i2s_tdm); + ret = rockchip_i2s_tdm_init_dai(i2s_tdm); + if (ret) + return ret; i2s_tdm->frame_width = 64; -- 2.34.0

1 1

[PATCH v2 1/3] ASoC: SOF: sof-audio: setup sched widgets during pipeline complete step
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Older firmware prior to ABI 3.19 has a dependency where the scheduler widgets need to be setup last. Moving the call to sof_widget_setup() before the pipeline_complete() call also helps remove the need for the 'reverse' direction when walking through the widget list - this was only working because of the topology macros but the topology does not require any order. Fixes: 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0f2566f7c094..f4e142ec0fbd 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -637,16 +637,25 @@ const struct sof_ipc_pipe_new *snd_sof_pipeline_find(struct snd_sof_dev *sdev, int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) { + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct snd_sof_widget *swidget; struct snd_sof_route *sroute; int ret; /* restore pipeline components */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { /* only set up the widgets belonging to static pipelines */ if (!verify && swidget->dynamic_pipeline_widget) continue; + /* + * For older firmware, skip scheduler widgets in this loop, + * sof_widget_setup() will be called in the 'complete pipeline' loop + */ + if (v->abi_version < SOF_ABI_VER(3, 19, 0) && + swidget->id == snd_soc_dapm_scheduler) + continue; + /* update DAI config. The IPC will be sent in sof_widget_setup() */ if (WIDGET_IS_DAI(swidget->id)) { struct snd_sof_dai *dai = swidget->private; @@ -694,6 +703,12 @@ int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) if (!verify && swidget->dynamic_pipeline_widget) continue; + if (v->abi_version < SOF_ABI_VER(3, 19, 0)) { + ret = sof_widget_setup(sdev, swidget); + if (ret < 0) + return ret; + } + swidget->complete = snd_sof_complete_pipeline(sdev, swidget); break; @@ -722,7 +737,7 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) * sroute->setup because during suspend all streams are suspended and during topology * loading the sound card unavailable to open PCMs. */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { if (swidget->dynamic_pipeline_widget) continue; base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

2 6

[PATCH] ASoC: SOF: handle paused streams during system suspend
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> During system suspend, paused streams do not get suspended. Therefore, we need to explicitly free these PCMs in the DSP and free the associated DAPM widgets so that they can be set up again during resume. Fixes: 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/pcm.c | 5 ++- sound/soc/sof/sof-audio.c | 74 +++++++++++++++++++++++++++++++++++++-- sound/soc/sof/sof-audio.h | 2 ++ 3 files changed, 76 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 31dd79b794f1..0ceb1a9cbf73 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -100,9 +100,8 @@ void snd_sof_pcm_period_elapsed(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(snd_sof_pcm_period_elapsed); -static int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream, - struct snd_sof_dev *sdev, - struct snd_sof_pcm *spcm) +int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream, struct snd_sof_dev *sdev, + struct snd_sof_pcm *spcm) { struct sof_ipc_stream stream; struct sof_ipc_reply reply; diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0f2566f7c094..038cdb4a9aa2 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -129,6 +129,14 @@ int sof_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) case snd_soc_dapm_buffer: ipc_free.hdr.cmd |= SOF_IPC_TPLG_BUFFER_FREE; break; + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + { + struct snd_sof_dai *dai = swidget->private; + + dai->configured = false; + fallthrough; + } default: ipc_free.hdr.cmd |= SOF_IPC_TPLG_COMP_FREE; break; @@ -705,6 +713,55 @@ int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) return 0; } +/* + * Free the PCM, its associated widgets and set the prepared flag to false for all PCMs that + * did not get suspended(ex: paused streams) so the widgets can be set up again during resume. + */ +static int sof_tear_down_left_over_pipelines(struct snd_sof_dev *sdev) +{ + struct snd_sof_widget *swidget; + struct snd_sof_pcm *spcm; + int dir, ret; + + /* + * free all PCMs and their associated DAPM widgets if their connected DAPM widget + * list is not NULL. This should only be true for paused streams at this point. + * This is equivalent to the handling of FE DAI suspend trigger for running streams. + */ + list_for_each_entry(spcm, &sdev->pcm_list, list) + for_each_pcm_streams(dir) { + struct snd_pcm_substream *substream = spcm->stream[dir].substream; + + if (!substream || !substream->runtime) + continue; + + if (spcm->stream[dir].list) { + ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm); + if (ret < 0) + return ret; + + ret = sof_widget_list_free(sdev, spcm, dir); + if (ret < 0) { + dev_err(sdev->dev, "failed to free widgets during suspend\n"); + return ret; + } + } + } + + /* + * free any left over DAI widgets. This is equivalent to the handling of suspend trigger + * for the BE DAI for running streams. + */ + list_for_each_entry(swidget, &sdev->widget_list, list) + if (WIDGET_IS_DAI(swidget->id) && swidget->use_count == 1) { + ret = sof_widget_free(sdev, swidget); + if (ret < 0) + return ret; + } + + return 0; +} + /* * For older firmware, this function doesn't free widgets for static pipelines during suspend. * It only resets use_count for all widgets. @@ -719,8 +776,8 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) /* * This function is called during suspend and for one-time topology verification during * first boot. In both cases, there is no need to protect swidget->use_count and - * sroute->setup because during suspend all streams are suspended and during topology - * loading the sound card unavailable to open PCMs. + * sroute->setup because during suspend all running streams are suspended and during + * topology loading the sound card unavailable to open PCMs. */ list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { if (swidget->dynamic_pipeline_widget) @@ -739,6 +796,19 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) return ret; } + /* + * Tear down all pipelines associated with PCMs that did not get suspended + * and unset the prepare flag so that they can be set up again during resume. + * Skip this step for older firmware. + */ + if (!verify && v->abi_version >= SOF_ABI_VER(3, 19, 0)) { + ret = sof_tear_down_left_over_pipelines(sdev); + if (ret < 0) { + dev_err(sdev->dev, "failed to tear down paused pipelines\n"); + return ret; + } + } + list_for_each_entry(sroute, &sdev->route_list, list) sroute->setup = false; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 389d56ac3aba..1c4f59d34717 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -265,4 +265,6 @@ int sof_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); /* PCM */ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, int dir); int sof_widget_list_free(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, int dir); +int sof_pcm_dsp_pcm_free(struct snd_pcm_substream *substream, struct snd_sof_dev *sdev, + struct snd_sof_pcm *spcm); #endif base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

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[PATCH 1/2] ASoC: SOF: sof-audio: setup sched widgets during pipeline complete step
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Older firmware prior to ABI 3.19 has a dependency where the scheduler widgets need to be setup last. Moving the call to sof_widget_setup() before the pipeline_complete() call also helps remove the need for the 'reverse' direction when walking through the widget list - this was only working because of the topology macros but the topology does not require any order. Fixes: 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/sof-audio.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 0f2566f7c094..f4e142ec0fbd 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -637,16 +637,25 @@ const struct sof_ipc_pipe_new *snd_sof_pipeline_find(struct snd_sof_dev *sdev, int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) { + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct snd_sof_widget *swidget; struct snd_sof_route *sroute; int ret; /* restore pipeline components */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { /* only set up the widgets belonging to static pipelines */ if (!verify && swidget->dynamic_pipeline_widget) continue; + /* + * For older firmware, skip scheduler widgets in this loop, + * sof_widget_setup() will be called in the 'complete pipeline' loop + */ + if (v->abi_version < SOF_ABI_VER(3, 19, 0) && + swidget->id == snd_soc_dapm_scheduler) + continue; + /* update DAI config. The IPC will be sent in sof_widget_setup() */ if (WIDGET_IS_DAI(swidget->id)) { struct snd_sof_dai *dai = swidget->private; @@ -694,6 +703,12 @@ int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) if (!verify && swidget->dynamic_pipeline_widget) continue; + if (v->abi_version < SOF_ABI_VER(3, 19, 0)) { + ret = sof_widget_setup(sdev, swidget); + if (ret < 0) + return ret; + } + swidget->complete = snd_sof_complete_pipeline(sdev, swidget); break; @@ -722,7 +737,7 @@ int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) * sroute->setup because during suspend all streams are suspended and during topology * loading the sound card unavailable to open PCMs. */ - list_for_each_entry_reverse(swidget, &sdev->widget_list, list) { + list_for_each_entry(swidget, &sdev->widget_list, list) { if (swidget->dynamic_pipeline_widget) continue; base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

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[PATCH v4 0/4] ASoC: SOF: Intel: power optimizations with HDaudio SPIB register
by Pierre-Louis Bossart 24 Nov '21

24 Nov '21

This patchset was initially provided in a larger series that was split in two [1]. This part only provides support for the SPIB register support, added on Intel platforms since Skylake (2015). The use of the SPIB register helps reduce power consumption - though to a smaller degree than DMI_L1. This hardware capability is however incompatible with userspace-initiated rewinds typically used by PulseAudio. In the past (2015..2017) Intel suggested an API extension to let applications disable rewinds. At the time the feedback was that such a capability was too Intel-specific and SPIB remained unused except for loading DSP code. We now see devices with smaller batteries being released, and it's time to revisit Linux support for SPIB to extend battery life. In this update the rewinds are disabled via an opt-in kernel parameter. In the previous reviews, there was consensus that a Kconfig option was too complicated for distributions to set, and we are missing a TBD API to expose such capabilities to user-space. The debate on whether or not to use rewinds, and the impact of disabling rewinds, will likely be closed when Intel releases the 'deep-buffer' support, currently under development [2][3]. With this solution, rewinds will not be needed, ever. When an application deals with content that is not latency-sensitive (e.g. music playback), it will be able to reduce power consumption by selecting a different PCM device with increased buffering capabilities. Low-latency streams will be handled by the 'regular' path. In other words, the impossible compromise between power and latency will be handled with different PCM devices/profiles for the same endpoint, and we can push the design of capability negotiation to a later time when all the building blocks (firmware topology, kernel, userspace) are ready - we still have firmware xruns, DPCM race conditions to solve, and a need to describe these alternate PCM devices with UCM using 'modifiers'. [1] https://lore.kernel.org/alsa-devel/20210610205326.1176400-1-pierre-louis.bo… [2] https://github.com/thesofproject/linux/pull/3146 [3] https://github.com/thesofproject/sof/pull/4611 Changes since v3: Added Takashi's Reviewed-by tags on patches 1 and 2 Used SYNC_APPLPTR instead of EXPLICIT_SYNC as suggested by Takashi Added a missing .ack callback for IceLake Changes since v2: Added check on appl_ptr (Takashi) Moved rewind detection to deal with SYNC_PTR (Takashi) Pierre-Louis Bossart (2): ALSA: pcm: unconditionally check if appl_ptr is in 0..boundary range ALSA: pcm: introduce INFO_NO_REWINDS flag Ranjani Sridharan (2): ASoC: SOF: pcm: add .ack callback support ASoC: SOF: Intel: add .ack support for HDaudio platforms include/uapi/sound/asound.h | 2 +- sound/core/pcm_lib.c | 17 +++++++++++++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/hda-pcm.c | 41 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/hda-stream.c | 2 ++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/tgl.c | 1 + sound/soc/sof/ops.h | 10 ++++++++ sound/soc/sof/pcm.c | 9 +++++++ sound/soc/sof/sof-priv.h | 3 +++ 12 files changed, 86 insertions(+), 3 deletions(-) -- 2.25.1

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[PATCH] ASoC: SOF: hda: reset DAI widget before reconfiguring it
by Kai Vehmanen 24 Nov '21

24 Nov '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> It is not unusual for ALSA/ASoC hw_params callbacks to be invoked multiple times. Reset and free the DAI widget before reconfiguring it to keep the DAI widget use_count balanced. Fixes: 0acb48dd31e3 ("ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1e1e9659ea86..54dd18dce8d0 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -58,6 +58,13 @@ int hda_ctrl_dai_widget_setup(struct snd_soc_dapm_widget *w) return -EINVAL; } + /* DAI already configured, reset it before reconfiguring it */ + if (sof_dai->configured) { + ret = hda_ctrl_dai_widget_free(w); + if (ret < 0) + return ret; + } + config = &sof_dai->dai_config[sof_dai->current_config]; /* base-commit: 65c16dd2942f4476a3f96a2625b1475c6137c09a -- 2.33.0

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[PATCH] ASoC: Intel: soc-acpi: add entry for ESSX8336 on CML
by Pierre-Louis Bossart 24 Nov '21

24 Nov '21

We have configurations for this codec on APL, GLK, JSL and TGL, somehow the information that some designs rely on CometLake was not shared. BugLink: https://github.com/thesofproject/linux/issues/3248 Fixes: 790049fb6623 ("ASoC: Intel: soc-acpi: apl/glk/tgl: add entry for devices based on ES8336 codec") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index b4eb0c97edf1..4eebc79d4b48 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -81,6 +81,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { .sof_fw_filename = "sof-cml.ri", .sof_tplg_filename = "sof-cml-da7219-max98390.tplg", }, + { + .id = "ESSX8336", + .drv_name = "sof-essx8336", + .sof_fw_filename = "sof-cml.ri", + .sof_tplg_filename = "sof-cml-es8336.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines); -- 2.25.1

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[PATCH 4.19 271/323] ALSA: ISA: not for M68K
by Greg Kroah-Hartman 24 Nov '21

24 Nov '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 3c05f1477e62ea5a0a8797ba6a545b1dc751fb31 ] On m68k, compiling drivers under SND_ISA causes build errors: ../sound/core/isadma.c: In function 'snd_dma_program': ../sound/core/isadma.c:33:17: error: implicit declaration of function 'claim_dma_lock' [-Werror=implicit-function-declaration] 33 | flags = claim_dma_lock(); | ^~~~~~~~~~~~~~ ../sound/core/isadma.c:41:9: error: implicit declaration of function 'release_dma_lock' [-Werror=implicit-function-declaration] 41 | release_dma_lock(flags); | ^~~~~~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_playback_prepare': ../sound/isa/sb/sb16_main.c:253:72: error: 'DMA_AUTOINIT' undeclared (first use in this function) 253 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); | ^~~~~~~~~~~~ ../sound/isa/sb/sb16_main.c:253:72: note: each undeclared identifier is reported only once for each function it appears in ../sound/isa/sb/sb16_main.c: In function 'snd_sb16_capture_prepare': ../sound/isa/sb/sb16_main.c:322:71: error: 'DMA_AUTOINIT' undeclared (first use in this function) 322 | snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); | ^~~~~~~~~~~~ and more... Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-m68k(a)lists.linux-m68k.org Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Link: https://lore.kernel.org/r/20211016062602.3588-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/Makefile | 2 ++ sound/isa/Kconfig | 2 +- sound/pci/Kconfig | 1 + 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/Makefile b/sound/core/Makefile index ee4a4a6b99ba7..d123587c0fd8f 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -9,7 +9,9 @@ ifneq ($(CONFIG_SND_PROC_FS),) snd-y += info.o snd-$(CONFIG_SND_OSSEMUL) += info_oss.o endif +ifneq ($(CONFIG_M68K),y) snd-$(CONFIG_ISA_DMA_API) += isadma.o +endif snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o snd-$(CONFIG_SND_VMASTER) += vmaster.o snd-$(CONFIG_SND_JACK) += ctljack.o jack.o diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index d7db1eeebc844..f8f3433925bb4 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -21,7 +21,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" depends on ISA || COMPILE_TEST - depends on ISA_DMA_API + depends on ISA_DMA_API && !M68K default y help Support for sound devices connected via the ISA bus. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 4105d9f653d90..bbaf46dc3f804 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -278,6 +278,7 @@ config SND_CS46XX_NEW_DSP config SND_CS5530 tristate "CS5530 Audio" depends on ISA_DMA_API && (X86_32 || COMPILE_TEST) + depends on !M68K select SND_SB16_DSP help Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. -- 2.33.0

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