February 2020 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5682: Enable PLL2 function
by derek.fang＠realtek.com 13 Feb '20

13 Feb '20

From: "derek.fang" <derek.fang(a)realtek.com> Enable RT5682 PLL2 function to implement the more complex frequency conversion. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 211 ++++++++++++++++++++++++++++++++++++---------- sound/soc/codecs/rt5682.h | 39 ++++++++- 2 files changed, 203 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 82a6366..9fbb386 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -64,9 +64,9 @@ struct rt5682_priv { int bclk[RT5682_AIFS]; int master[RT5682_AIFS]; - int pll_src; - int pll_in; - int pll_out; + int pll_src[RT5682_PLLS]; + int pll_in[RT5682_PLLS]; + int pll_out[RT5682_PLLS]; int jack_type; }; @@ -75,6 +75,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, + {RT5682_PLL2_INTERNAL, 0x8266}, }; static const struct reg_default rt5682_reg[] = { @@ -222,7 +223,7 @@ static const struct reg_default rt5682_reg[] = { {0x0148, 0x0000}, {0x0149, 0x0000}, {0x0150, 0x79a1}, - {0x0151, 0x0000}, + {0x0156, 0xaaaa}, {0x0160, 0x4ec0}, {0x0161, 0x0080}, {0x0162, 0x0200}, @@ -928,10 +929,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); usleep_range(15000, 20000); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, RT5682_PWR_CBJ); @@ -1298,6 +1299,21 @@ static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, return 0; } +static int is_sys_clk_from_pll2(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL2) + return 1; + else + return 0; +} + static int is_using_asrc(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_widget *sink) { @@ -1612,9 +1628,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, - 0, NULL, 0), + 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + NULL, 0), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1796,7 +1814,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*PLL*/ {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"ADC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"ADC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"DAC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, @@ -2053,8 +2075,10 @@ static int rt5682_hw_params(struct snd_pcm_substream *substream, RT5682_I2S1_DL_MASK, len_1); if (rt5682->master[RT5682_AIF1]) { snd_soc_component_update_bits(component, - RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, - pre_div << RT5682_I2S_M_DIV_SFT); + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK | + RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); } if (params_channels(params) == 1) /* mono mode */ snd_soc_component_update_bits(component, @@ -2227,61 +2251,157 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, unsigned int freq_out) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct rl6231_pll_code pll_code; + struct rl6231_pll_code pll_code, pll2f_code, pll2b_code; + unsigned int pll2_fout1; int ret; - if (source == rt5682->pll_src && freq_in == rt5682->pll_in && - freq_out == rt5682->pll_out) + if (source == rt5682->pll_src[pll_id] && + freq_in == rt5682->pll_in[pll_id] && + freq_out == rt5682->pll_out[pll_id]) return 0; if (!freq_in || !freq_out) { dev_dbg(component->dev, "PLL disabled\n"); - rt5682->pll_in = 0; - rt5682->pll_out = 0; + rt5682->pll_in[pll_id] = 0; + rt5682->pll_out[pll_id] = 0; snd_soc_component_update_bits(component, RT5682_GLB_CLK, RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); return 0; } - switch (source) { - case RT5682_PLL1_S_MCLK: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); - break; - case RT5682_PLL1_S_BCLK1: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); - break; - default: - dev_err(component->dev, "Unknown PLL Source %d\n", source); - return -EINVAL; - } + if (pll_id == RT5682_PLL2) { + switch (source) { + case RT5682_PLL2_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL2_SRC_MASK, + RT5682_PLL2_SRC_MCLK); + break; + default: + dev_err(component->dev, "Unknown PLL2 Source %d\n", + source); + return -EINVAL; + } - ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); - if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); - return ret; + /** + * PLL2 concatenates 2 PLL units. + * We suggest the Fout of the front PLL is 3.84MHz. + */ + pll2_fout1 = 3840000; + ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + dev_dbg(component->dev, "PLL2F: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + freq_in, pll2_fout1, + pll2f_code.m_bp, + (pll2f_code.m_bp ? 0 : pll2f_code.m_code), + pll2f_code.n_code, pll2f_code.k_code); + + ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + pll2_fout1); + return ret; + } + dev_dbg(component->dev, "PLL2B: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + pll2_fout1, freq_out, + pll2b_code.m_bp, + (pll2b_code.m_bp ? 0 : pll2b_code.m_code), + pll2b_code.n_code, pll2b_code.k_code); + + snd_soc_component_write(component, RT5682_PLL2_CTRL_1, + pll2f_code.k_code << RT5682_PLL2F_K_SFT | + pll2b_code.k_code << RT5682_PLL2B_K_SFT | + pll2b_code.m_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_2, + pll2f_code.m_code << RT5682_PLL2F_M_SFT | + pll2b_code.n_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_3, + pll2f_code.n_code << RT5682_PLL2F_N_SFT); + snd_soc_component_update_bits(component, RT5682_PLL2_CTRL_4, + RT5682_PLL2B_M_BP_MASK | RT5682_PLL2F_M_BP_MASK | 0xf, + (pll2b_code.m_bp ? 1 : 0) << RT5682_PLL2B_M_BP_SFT | + (pll2f_code.m_bp ? 1 : 0) << RT5682_PLL2F_M_BP_SFT | + 0xf); + } else { + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL1 Source %d\n", + source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); } - dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", - pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), - pll_code.n_code, pll_code.k_code); + rt5682->pll_in[pll_id] = freq_in; + rt5682->pll_out[pll_id] = freq_out; + rt5682->pll_src[pll_id] = source; + + return 0; +} + +static int rt5682_set_bclk1_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - snd_soc_component_write(component, RT5682_PLL_CTRL_1, - pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); - snd_soc_component_write(component, RT5682_PLL_CTRL_2, - (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | - pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + rt5682->bclk[dai->id] = ratio; - rt5682->pll_in = freq_in; - rt5682->pll_out = freq_out; - rt5682->pll_src = source; + switch (ratio) { + case 256: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_256); + break; + case 128: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_128); + break; + case 64: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_32); + break; + default: + dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio); + return -EINVAL; + } return 0; } -static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +static int rt5682_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio) { struct snd_soc_component *component = dai->component; struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); @@ -2300,7 +2420,7 @@ static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) RT5682_I2S2_BCLK_MS2_32); break; default: - dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio); return -EINVAL; } @@ -2389,12 +2509,13 @@ static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, .set_tdm_slot = rt5682_set_tdm_slot, + .set_bclk_ratio = rt5682_set_bclk1_ratio, }; static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, - .set_bclk_ratio = rt5682_set_bclk_ratio, + .set_bclk_ratio = rt5682_set_bclk2_ratio, }; static struct snd_soc_dai_driver rt5682_dai[] = { diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 4d3a8c4..465c99b 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -177,7 +177,7 @@ #define RT5682_TEST_MODE_CTRL_4 0x0148 #define RT5682_TEST_MODE_CTRL_5 0x0149 #define RT5682_PLL1_INTERNAL 0x0150 -#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_PLL2_INTERNAL 0x0156 #define RT5682_STO_NG2_CTRL_1 0x0160 #define RT5682_STO_NG2_CTRL_2 0x0161 #define RT5682_STO_NG2_CTRL_3 0x0162 @@ -738,7 +738,7 @@ #define RT5682_ADC_OSR_D_24 (0x7 << 12) #define RT5682_ADC_OSR_D_32 (0x8 << 12) #define RT5682_ADC_OSR_D_48 (0x9 << 12) -#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 8) #define RT5682_I2S_M_DIV_SFT 8 #define RT5682_I2S_M_D_1 (0x0 << 8) #define RT5682_I2S_M_D_2 (0x1 << 8) @@ -820,6 +820,12 @@ #define RT5682_TDM_DF_PCM_B (0x3 << 11) #define RT5682_TDM_DF_PCM_A_N (0x6 << 11) #define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_BCLK_MS1_MASK (0x3 << 9) +#define RT5682_TDM_BCLK_MS1_SFT 9 +#define RT5682_TDM_BCLK_MS1_32 (0x0 << 9) +#define RT5682_TDM_BCLK_MS1_64 (0x1 << 9) +#define RT5682_TDM_BCLK_MS1_128 (0x2 << 9) +#define RT5682_TDM_BCLK_MS1_256 (0x3 << 9) #define RT5682_TDM_CL_MASK (0x3 << 4) #define RT5682_TDM_CL_16 (0x0 << 4) #define RT5682_TDM_CL_20 (0x1 << 4) @@ -1049,6 +1055,28 @@ #define RT5682_PWR_CLK1M_PD (0x0 << 8) #define RT5682_PWR_CLK1M_PU (0x1 << 8) +/* PLL2 M/N/K Code Control 1 (0x009b) */ +#define RT5682_PLL2F_K_MASK (0x1f << 8) +#define RT5682_PLL2F_K_SFT 8 +#define RT5682_PLL2B_K_MASK (0xf << 4) +#define RT5682_PLL2B_K_SFT 4 +#define RT5682_PLL2B_M_MASK (0xf << 0) + +/* PLL2 M/N/K Code Control 2 (0x009c) */ +#define RT5682_PLL2F_M_MASK (0x3f << 8) +#define RT5682_PLL2F_M_SFT 8 +#define RT5682_PLL2B_N_MASK (0x3f << 0) + +/* PLL2 M/N/K Code Control 2 (0x009d) */ +#define RT5682_PLL2F_N_MASK (0x7f << 8) +#define RT5682_PLL2F_N_SFT 8 + +/* PLL2 M/N/K Code Control 2 (0x009e) */ +#define RT5682_PLL2B_M_BP_MASK (0x1 << 11) +#define RT5682_PLL2B_M_BP_SFT 11 +#define RT5682_PLL2F_M_BP_MASK (0x1 << 7) +#define RT5682_PLL2F_M_BP_SFT 7 + /* RC Clock Control (0x009f) */ #define RT5682_POW_IRQ (0x1 << 15) #define RT5682_POW_JDH (0x1 << 14) @@ -1315,6 +1343,13 @@ enum { RT5682_PLL1_S_MCLK, RT5682_PLL1_S_BCLK1, RT5682_PLL1_S_RCCLK, + RT5682_PLL2_S_MCLK, +}; + +enum { + RT5682_PLL1, + RT5682_PLL2, + RT5682_PLLS, }; enum { -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: doc: fix snd_hda_intel driver name
by Radoslaw Smigielski 13 Feb '20

13 Feb '20

Update driver name snd-hda-intel to proper, existing driver name snd_hda_intel in Documentation/sound/hd-audio/notes.rst. Signed-off-by: Radoslaw Smigielski <radoslaw.smigielski(a)interia.pl> --- Documentation/sound/hd-audio/notes.rst | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index 0f3109d9abc8..56ccc15f4d26 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -17,12 +17,12 @@ methods for the HD-audio hardware. The HD-audio component consists of two parts: the controller chip and the codec chips on the HD-audio bus. Linux provides a single driver -for all controllers, snd-hda-intel. Although the driver name contains +for all controllers, snd_hda_intel. Although the driver name contains a word of a well-known hardware vendor, it's not specific to it but for all controller chips by other companies. Since the HD-audio controllers are supposed to be compatible, the single snd-hda-driver should work in most cases. But, not surprisingly, there are known -bugs and issues specific to each controller type. The snd-hda-intel +bugs and issues specific to each controller type. The snd_hda_intel driver has a bunch of workarounds for these as described below. A controller may have multiple codecs. Usually you have one audio @@ -31,7 +31,7 @@ multiple audio codecs, e.g. for analog and digital outputs, and the driver might not work properly because of conflict of mixer elements. This should be fixed in future if such hardware really exists. -The snd-hda-intel driver has several different codec parsers depending +The snd_hda_intel driver has several different codec parsers depending on the codec. It has a generic parser as a fallback, but this functionality is fairly limited until now. Instead of the generic parser, usually the codec-specific parser (coded in patch_*.c) is used @@ -226,7 +226,7 @@ the external amplifier bits. Thus a headphone output has a slightly better chance. Before making a bug report, double-check whether the mixer is set up -correctly. The recent version of snd-hda-intel driver provides mostly +correctly. The recent version of snd_hda_intel driver provides mostly "Master" volume control as well as "Front" volume (where Front indicates the front-channels). In addition, there can be individual "Headphone" and "Speaker" controls. @@ -596,7 +596,7 @@ For example, if you have two cards, one for an on-board analog and one for an HDMI video board, you may pass patch option like below: :: - options snd-hda-intel patch=on-board-patch,hdmi-patch + options snd_hda_intel patch=on-board-patch,hdmi-patch Power-Saving -- 2.24.1

2 3

[alsa-devel] [PATCH v2 0/5] soundwire: stream: fix state machines and transitions
by Pierre-Louis Bossart 13 Feb '20

13 Feb '20

The existing stream support works fine with simple cases, but does not map well with ALSA transitions for underflows/resume where prepare() can be called multiple times. Concurrency with multiple devices per links or multiple streams enabled on the same link also needs to be fixed. These patches are the result of hours of validation on the Intel side and should benefit other implementations since there is nothing hardware-specific. The Intel-specific changes being reviewed do depend on those stream changes though to be functional. Changes since v1: Removed spurious code block change flagged by Vinod No change (replies provided in v1 thread) Github link issue is public, no reason to remove it Bandwidth computation on ALSA prepare/start (for resume cases) handled internally in stream layer. Kept emacs comment formatting. No additional code/test for concurrent streams (not supported due to locking) Bard Liao (1): soundwire: stream: only prepare stream when it is configured. Pierre-Louis Bossart (2): soundwire: stream: update state machine and add state checks soundwire: stream: do not update parameters during DISABLED-PREPARED transition Rander Wang (2): soundwire: stream: fix support for multiple Slaves on the same link soundwire: stream: don't program ports when a stream that has not been prepared Documentation/driver-api/soundwire/stream.rst | 61 +++++++++---- drivers/soundwire/stream.c | 90 ++++++++++++++++--- 2 files changed, 124 insertions(+), 27 deletions(-) -- 2.20.1

2 7

[alsa-devel] [PATCH] drm/mediatek: fix race condition of plugged_cb and codec_dev
by Tzung-Bi Shih 13 Feb '20

13 Feb '20

hdmi_conn_detect and mtk_hdmi_audio_hook_plugged_cb woule be called by different threads. plugged_cb and codec_dev are in danger of race condition. Fixes by: - Checks NULLs first. - Uses WRITE_ONCE() to prevent store tearing (i.e. write to plugged_cb after codec_dev). - Uses codec_dev as a signal to report HDMI jack status. The patch intends to not use mutex to protect the shared variables. In runtime, mtk_hdmi_audio_hook_plugged_cb would be called only once if no one rebinds the sound card. In contrast, hdmi_conn_detect could be called multiple times. Uses a mutex within hdmi_conn_detect calling path is a waste (no lock contention). Fixes: 5d3c64477392 ("drm/mediatek: support HDMI jack status reporting") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- drivers/gpu/drm/mediatek/mtk_hdmi.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index 03aeb73005ef..746567c91bb7 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -1201,7 +1201,7 @@ mtk_hdmi_update_plugged_status(struct mtk_hdmi *hdmi) { bool connected = mtk_cec_hpd_high(hdmi->cec_dev); - if (hdmi->plugged_cb && hdmi->codec_dev) + if (hdmi->codec_dev) hdmi->plugged_cb(hdmi->codec_dev, connected); return connected ? @@ -1669,8 +1669,12 @@ static int mtk_hdmi_audio_hook_plugged_cb(struct device *dev, void *data, { struct mtk_hdmi *hdmi = data; - hdmi->plugged_cb = fn; - hdmi->codec_dev = codec_dev; + if (!fn || !codec_dev) + return -EINVAL; + + /* Use WRITE_ONCE() to prevent store tearing. */ + WRITE_ONCE(hdmi->plugged_cb, fn); + WRITE_ONCE(hdmi->codec_dev, codec_dev); mtk_hdmi_update_plugged_status(hdmi); return 0; -- 2.25.0.225.g125e21ebc7-goog

1 1

[alsa-devel] [PATCH v2 0/3] ASoC: mediatek: mt8173-rt5650: HDMI jack reporting
by Tzung-Bi Shih 13 Feb '20

13 Feb '20

The series supports HDMI jack reporting to mt8173-rt5650. The 1st patch is less related. It stops running in probe() if failed to register audio driver. The 2nd patch supports jack reporting in DRM driver. The 3rd patch supports jack reporting in mt8173-rt5650 ASoC machine driver. Changes from v1: - refactor the 2nd patch to simplify the code Tzung-Bi Shih (3): drm/mediatek: exit earlier if failed to register audio driver drm/mediatek: support HDMI jack status reporting ASoC: mediatek: mt8173-rt5650: support HDMI jack reporting drivers/gpu/drm/mediatek/mtk_hdmi.c | 45 ++++++++++++++++++++--- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 17 ++++++++- 2 files changed, 55 insertions(+), 7 deletions(-) -- 2.25.0.341.g760bfbb309-goog

3 11

[alsa-devel] [PATCH] ALSA: usb-audio: Add clock validity quirk for Denon MC7000/MCX8000
by Alexander Tsoy 13 Feb '20

13 Feb '20

It should be safe to ignore clock validity check result if the following conditions are met: - only one single sample rate is supported; - the terminal is directly connected to the clock source; - the clock type is internal. This is to deal with some Denon DJ controllers that always reports that clock is invalid. Tested-by: Tobias Oszlanyi <toszlanyi(a)yahoo.de> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/clock.c | 91 ++++++++++++++++++++++++++++++++-------------- sound/usb/clock.h | 4 +- sound/usb/format.c | 3 +- 3 files changed, 66 insertions(+), 32 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 018b1ecb5404..a48313dfa967 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,8 +151,34 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } +/* + * Assume the clock is valid if clock source supports only one single sample + * rate, the terminal is connected directly to it (there is no clock selector) + * and clock type is internal. This is to deal with some Denon DJ controllers + * that always reports that clock is invalid. + */ +static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, + struct audioformat *fmt, + int source_id) +{ + if (fmt->protocol == UAC_VERSION_2) { + struct uac_clock_source_descriptor *cs_desc = + snd_usb_find_clock_source(chip->ctrl_intf, source_id); + + if (!cs_desc) + return false; + + return (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT); + } + + return false; +} + static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, - int protocol, + struct audioformat *fmt, int source_id) { int err; @@ -160,7 +186,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; u32 bmControls; - if (protocol == UAC_VERSION_3) { + if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); @@ -194,10 +220,14 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, return false; } - return data ? true : false; + if (data) + return true; + else + return uac_clock_source_is_valid_quirk(chip, fmt, source_id); } -static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, +static int __uac_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, unsigned long *visited, bool validate) { struct uac_clock_source_descriptor *source; @@ -217,7 +247,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_2, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -248,8 +278,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac_clock_find_source(chip, selector->baCSourceID[ret - 1], - visited, validate); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], + visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -260,8 +291,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -281,14 +313,16 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, /* FIXME: multipliers only act as pass-thru element for now */ multiplier = snd_usb_find_clock_multiplier(chip->ctrl_intf, entity_id); if (multiplier) - return __uac_clock_find_source(chip, multiplier->bCSourceID, - visited, validate); + return __uac_clock_find_source(chip, fmt, + multiplier->bCSourceID, + visited, validate); return -EINVAL; } -static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, - unsigned long *visited, bool validate) +static int __uac3_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, + unsigned long *visited, bool validate) { struct uac3_clock_source_descriptor *source; struct uac3_clock_selector_descriptor *selector; @@ -307,7 +341,7 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source_v3(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_3, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -338,7 +372,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[ret - 1], + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -350,8 +385,9 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -372,7 +408,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, multiplier = snd_usb_find_clock_multiplier_v3(chip->ctrl_intf, entity_id); if (multiplier) - return __uac3_clock_find_source(chip, multiplier->bCSourceID, + return __uac3_clock_find_source(chip, fmt, + multiplier->bCSourceID, visited, validate); return -EINVAL; @@ -389,18 +426,18 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, * * Returns the clock source UnitID (>=0) on success, or an error. */ -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate) +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate) { DECLARE_BITMAP(visited, 256); memset(visited, 0, sizeof(visited)); - switch (protocol) { + switch (fmt->protocol) { case UAC_VERSION_2: - return __uac_clock_find_source(chip, entity_id, visited, + return __uac_clock_find_source(chip, fmt, fmt->clock, visited, validate); case UAC_VERSION_3: - return __uac3_clock_find_source(chip, entity_id, visited, + return __uac3_clock_find_source(chip, fmt, fmt->clock, visited, validate); default: return -EINVAL; @@ -501,8 +538,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * automatic clock selection if the current clock is not * valid. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, true); + clock = snd_usb_clock_find_source(chip, fmt, true); if (clock < 0) { /* We did not find a valid clock, but that might be * because the current sample rate does not match an @@ -510,8 +546,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * and we will do another validation after setting the * rate. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, false); + clock = snd_usb_clock_find_source(chip, fmt, false); if (clock < 0) return clock; } @@ -577,7 +612,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, validation: /* validate clock after rate change */ - if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + if (!uac_clock_source_is_valid(chip, fmt, clock)) return -ENXIO; return 0; } diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 076e31b79ee0..68df0fbe09d0 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -6,7 +6,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate); -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate); +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate); #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 9260136e4c9b..a77d6b1c403d 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -325,8 +325,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0, ret_l6; - int clock = snd_usb_clock_find_source(chip, fp->protocol, - fp->clock, false); + int clock = snd_usb_clock_find_source(chip, fp, false); if (clock < 0) { dev_err(&dev->dev, -- 2.24.1

2 1

Re: [alsa-devel] Welcome to the "Alsa-devel" mailing list
by Srinivas Pulukuru 13 Feb '20

13 Feb '20

This is my first post to this mailing list, so apologize for any inconvenience if this is a wrong place to post such issues. If this is a wrong place please forward it to the right mailing list or let me know and I'll repost it there. ****************************** I are trying to create a 16 channel shared TDM frame that is shared between 8 different playback sources, to be sent out to the default hw:0 card. I used dshare plugin for this purpose, please see the attached asound.conf. I tried using the share plugin but the performance is really bad and is not practical for our use given the number of channels that we are using. when starting the playback the poll function called from snd_pcm_wait_nocheck in pcm.c hangs trying to access the /dev/snd/timer and never returns. It is able to access the first couple of times then hangs after that. ==> snd_pcm_write_areas ==> snd_pcm_wait_nocheck ==> poll() I see that the snd_pcm_direct_t->spcm->hw.ptr that tracks the frames in time is stuck and is not getting incremented in this case. I have seen several old posts regarding this issue and not sure which version has this been fixed or it still an issue. I have tried using dmix plugin too with the same result. My system doesn't have a sound card at the back end of the processor(IMX8M). I just need a shared buffer passed to the SAI interface that gets sent out of the SAI datalines (I2S) as a TDM frame. The codec drivers have been replaced with null terminated codecs as there at no DAC's connected to the processor. The SAI I2S datalines are connected to a FPGA that demultiplexes and processes the TDM frame to be sent out to multiple audio outputs. I am using alsa 1.1.2 and kernel imx8m 4.14.98 . I have tried to move to later version of alsa that I was able to build. I went till 1.1.7 but I still get the same issue. I have posted more details in the following link in case you need more details including the asound.conf and the batch file that runs 8 different pcm files to 8 different virtual sources connected to the same HW card. https://unix.stackexchange.com/questions/567295/alsa-16-channel-audio-playb… ******************************** Thanks, Srinivas Srinivas Pulukuru | Sr. Firmware Engineer Phone: 800-237-2041 Email: spulukuru(a)crestron.com Crestron Plano 7250 Dallas Parkway, Suite 600, Plano, Texas 75024 -----Original Message----- From: Alsa-devel <alsa-devel-bounces(a)alsa-project.org> On Behalf Of alsa-devel-request(a)alsa-project.org Sent: Wednesday, February 12, 2020 11:29 PM To: Srinivas Pulukuru <spulukuru(a)crestron.com> Subject: Welcome to the "Alsa-devel" mailing list Welcome to the Alsa-devel(a)alsa-project.org mailing list! To post to this list, send your message to: alsa-devel(a)alsa-project.org General information about the mailing list is at: https://urldefense.proofpoint.com/v2/url?u=https-3A__mailman.alsa-2Dproject… If you ever want to unsubscribe or change your options (eg, switch to or from digest mode, change your password, etc.), visit your subscription page at: https://urldefense.proofpoint.com/v2/url?u=https-3A__mailman.alsa-2Dproject… You can also make such adjustments via email by sending a message to: Alsa-devel-request(a)alsa-project.org with the word `help' in the subject or body (don't include the quotes), and you will get back a message with instructions. You must know your password to change your options (including changing the password, itself) or to unsubscribe without confirmation. It is: psr9611p Normally, Mailman will remind you of your alsa-project.org mailing list passwords once every month, although you can disable this if you prefer. This reminder will also include instructions on how to unsubscribe or change your account options. There is also a button on your options page that will email your current password to you.

1 0

[alsa-devel] [PATCH] ASoC: dt-bindings: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 13 Feb '20

13 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/simple-card.txt | 351 -------------- .../bindings/sound/simple-card.yaml | 459 ++++++++++++++++++ 2 files changed, 459 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6e37b..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,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. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000000..7c9ced323787 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,459 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle-array + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle-array + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + clocks: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/phandle-array + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": true + "#size-cells": true + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + simple-audio-card,routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + sink, the second being the source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + "^simple-audio-card,cpu(@.*)?": + description: CPU node + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@.*)?": + description: Codec node + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" +# use common *-gpio for +# simple-audio-card,hp-det-gpio: +# simple-audio-card,mic-det-gpio: + + "^simple-audio-card,dai-link(@.*)?": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + # use common *-gpio for + # hp-det-gpio: + # mic-det-gpio: + + patternProperties: + "^cpu(@.*)?": + $ref: "#/definitions/dai" + "^codec(@.*)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.17.1

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[alsa-devel] USB Type-C monitor flashes once when play a video file after unplug and re-plug the monitor
by AJ_Cheng＠compal.com 13 Feb '20

13 Feb '20

Hi Sirs, Here is chromebook SW team from Compal. As the mail title, we hit issue that the external monitor will flash once when play video after hot pluging. We can reproduce not only on chromebook but also ubuntu 16.04. There has higher failure rate with Dell Solomon dock and Dell S2718D monitor. We found adding the delay in "sound/pci/hda/patch_hdmi.c " can fix this issue.(as the attachment) May need your help to review and advice. Thanks. Here is the issue number in gitlab for more detail. https://gitlab.freedesktop.org/drm/intel/issues/318 AJ Cheng NID/NID1 e-mail: AJ_Cheng(a)compal.com<mailto:AJ_Cheng@compal.com> Tel: +886-2-8797-8599 ext. 17561 Mobile : +886-932827829 COMPAL Electronics, Inc.

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[alsa-devel] [PATCH 0/3] Add new module driver for new ASRC
by Shengjiu Wang 13 Feb '20

13 Feb '20

Add new module driver for new ASRC in i.MX815/865 Shengjiu Wang (3): ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI and platform drivers .../devicetree/bindings/sound/fsl,easrc.txt | 57 + sound/soc/fsl/fsl_asrc.h | 11 +- sound/soc/fsl/fsl_asrc_common.h | 22 + sound/soc/fsl/fsl_easrc.c | 2265 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 668 +++++ sound/soc/fsl/fsl_easrc_dma.c | 440 ++++ 6 files changed, 3453 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.txt create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h create mode 100644 sound/soc/fsl/fsl_easrc_dma.c -- 2.21.0

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