February 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: AMD: Configure master codec on all playback/capture cases
by Agrawal, Akshu 04 Mar '19

04 Mar '19

In the system design da7219 is the master codec and clocks are generated by it. Bclk is to be generated at the required rate for other codecs used when da7219 is acting only as clock master. For this call hw_params of da7219 during playback/capture on non da7219 codecs. Being able to set bclk at lower rate also fixes noise issue observed on some dmics. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 20 ++++++++++++++++++-- 1 file changed, 18 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index a5daad973ce5..ad327415290a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -45,6 +45,7 @@ #define CZ_PLAT_CLK 48000000 #define DUAL_CHANNEL 2 +static struct snd_soc_dai *codec_dai; static struct snd_soc_jack cz_jack; static struct clk *da7219_dai_clk; extern int bt_uart_enable; @@ -53,8 +54,10 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_card *card = rtd->card; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_component *component = codec_dai->component; + struct snd_soc_component *component; + + codec_dai = rtd->codec_dai; + component = codec_dai->component; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -255,6 +258,16 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) da7219_clk_disable(); } +static static int cz_da7219_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + /* da7219 Codec is clock master so setup as per the needs */ + if (codec_dai->driver->ops->hw_params) + return codec_dai->driver->ops->hw_params(substream, params, + codec_dai); + return 0; +} + static const struct snd_soc_ops cz_da7219_play_ops = { .startup = cz_da7219_play_startup, .shutdown = cz_da7219_shutdown, @@ -268,16 +281,19 @@ static const struct snd_soc_ops cz_da7219_cap_ops = { static const struct snd_soc_ops cz_max_play_ops = { .startup = cz_max_startup, .shutdown = cz_max_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic0_cap_ops = { .startup = cz_dmic0_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static const struct snd_soc_ops cz_dmic1_cap_ops = { .startup = cz_dmic1_startup, .shutdown = cz_dmic_shutdown, + .hw_params = cz_da7219_hw_params, }; static struct snd_soc_dai_link cz_dai_7219_98357[] = { -- 2.17.1

4 4

[alsa-devel] [PATCH RFC 0/3] tda998x updates for DAI formats and bclk_ratio
by Russell King - ARM Linux admin 04 Mar '19

04 Mar '19

This series addresses two issues with TDA998x that have been identified: 1) Peter found that the I2S format was not being explicitly set, and retains its value from whatever was previously running on the platform. Work around this by implementing support for setting the I2S format from the DAI format, rather than merely defaulting the register back to its power-on value. 2) Sven found that TDA998x does not work on his Freescale platform, which always uses a 64·fs bitclock. The TDA998x driver was deriving this information from the sample width, which, while it works for Beagle Bone Black, does not allow the driver to be used with other I2S sources that may have different behaviours. To work around that, we implement support for snd_soc_dai_set_bclk_ratio() in hdmi-codec, and propagate its value to TDA998x and other HDMI codecs via a new member. However, since snd_soc_dai_set_bclk_ratio() is never called, we need to avoid breaking any existing users, so we detect the lack of call by an impossible zero value, and subsitute a value corresponding with the TDA998x's old behaviour. It is hoped that snd_soc_dai_set_bclk_ratio() will see more adoption in ASoC, and the TDA998x specific defaulting can be removed. drivers/gpu/drm/i2c/tda998x_drv.c | 75 ++++++++++++++++++++++++++------------- include/drm/i2c/tda998x.h | 12 +++++-- include/sound/hdmi-codec.h | 1 + sound/soc/codecs/hdmi-codec.c | 45 +++++++++++++++++++++-- 4 files changed, 104 insertions(+), 29 deletions(-) -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

5 19

[alsa-devel] [PATCH] ASoC: hdmi-codec: avoid limiting params->msbits in hw_params()
by Russell King 04 Mar '19

04 Mar '19

Limiting the value of the passed in params->msbits in the hw_params() callback is redundant on three counts: 1. We already specify in the DAI driver that we can only handle up to 24 bits. This means msbits will be limited to 24 via the ALSA constraints imposed by the ASoC core, unless we have multiple codecs that can handle more bits. 2. Nothing in our hw_params() implementation uses this value. 3. The copy of the params that we are passed by the ASoC core never reads back the msbits value. Consequently, this code is unnecessary and does nothing useful. Remove it. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index d5f73c837281..35df73e42cbc 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -484,9 +484,6 @@ static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, params_width(params), params_rate(params), params_channels(params)); - if (params_width(params) > 24) - params->msbits = 24; - ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, sizeof(hp.iec.status)); if (ret < 0) { -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: fix reset gpio reference counting
by Philipp Puschmann 04 Mar '19

04 Mar '19

This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> --- Changes in v2: Fix typo in title tlv320aix3x -> tlv320aic3x sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: mediatek: btcvsd add loopback
by KaiChieh Chuang 03 Mar '19

03 Mar '19

add direct loopback path from rx to tx Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- sound/soc/mediatek/common/mtk-btcvsd.c | 69 +++++++++++++++++++++++++++++++++- 1 file changed, 68 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index 1b8bcdaf..9a163d7 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -49,6 +49,7 @@ enum bt_sco_state { BT_SCO_STATE_IDLE, BT_SCO_STATE_RUNNING, BT_SCO_STATE_ENDING, + BT_SCO_STATE_LOOPBACK, }; enum bt_sco_direct { @@ -486,7 +487,8 @@ static irqreturn_t mtk_btcvsd_snd_irq_handler(int irq_id, void *dev) if (bt->rx->state != BT_SCO_STATE_RUNNING && bt->rx->state != BT_SCO_STATE_ENDING && bt->tx->state != BT_SCO_STATE_RUNNING && - bt->tx->state != BT_SCO_STATE_ENDING) { + bt->tx->state != BT_SCO_STATE_ENDING && + bt->tx->state != BT_SCO_STATE_LOOPBACK) { dev_warn(bt->dev, "%s(), in idle state: rx->state: %d, tx->state: %d\n", __func__, bt->rx->state, bt->tx->state); goto irq_handler_exit; @@ -512,6 +514,42 @@ static irqreturn_t mtk_btcvsd_snd_irq_handler(int irq_id, void *dev) buf_cnt_tx = btsco_packet_info[packet_type][2]; buf_cnt_rx = btsco_packet_info[packet_type][3]; + if (bt->tx->state == BT_SCO_STATE_LOOPBACK) { + u8 *src, *dst; + unsigned long connsys_addr_rx, ap_addr_rx; + unsigned long connsys_addr_tx, ap_addr_tx; + + connsys_addr_rx = *bt->bt_reg_pkt_r; + ap_addr_rx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_rx & 0xFFFF); + + connsys_addr_tx = *bt->bt_reg_pkt_w; + ap_addr_tx = (unsigned long)bt->bt_sram_bank2_base + + (connsys_addr_tx & 0xFFFF); + + if (connsys_addr_tx == 0xdeadfeed || + connsys_addr_rx == 0xdeadfeed) { + /* bt return 0xdeadfeed if read reg during bt sleep */ + dev_warn(bt->dev, "%s(), connsys_addr_tx == 0xdeadfeed\n", + __func__); + goto irq_handler_exit; + } + + src = (u8 *)ap_addr_rx; + dst = (u8 *)ap_addr_tx; + + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_BT2ARM, src, + bt->tx->temp_packet_buf, + packet_length, + packet_num); + mtk_btcvsd_snd_data_transfer(BT_SCO_DIRECT_ARM2BT, + bt->tx->temp_packet_buf, dst, + packet_length, + packet_num); + bt->rx->rw_cnt++; + bt->tx->rw_cnt++; + } + if (bt->rx->state == BT_SCO_STATE_RUNNING || bt->rx->state == BT_SCO_STATE_ENDING) { if (bt->rx->xrun) { @@ -1067,6 +1105,33 @@ static int btcvsd_band_set(struct snd_kcontrol *kcontrol, return 0; } +static int btcvsd_loopback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + bool lpbk_en = bt->tx->state == BT_SCO_STATE_LOOPBACK; + + ucontrol->value.integer.value[0] = lpbk_en; + return 0; +} + +static int btcvsd_loopback_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *cmpnt = snd_soc_kcontrol_component(kcontrol); + struct mtk_btcvsd_snd *bt = snd_soc_component_get_drvdata(cmpnt); + + if (ucontrol->value.integer.value[0]) { + mtk_btcvsd_snd_set_state(bt, bt->tx, BT_SCO_STATE_LOOPBACK); + mtk_btcvsd_snd_set_state(bt, bt->rx, BT_SCO_STATE_LOOPBACK); + } else { + mtk_btcvsd_snd_set_state(bt, bt->tx, BT_SCO_STATE_RUNNING); + mtk_btcvsd_snd_set_state(bt, bt->rx, BT_SCO_STATE_RUNNING); + } + return 0; +} + static int btcvsd_tx_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1202,6 +1267,8 @@ static int btcvsd_tx_timestamp_get(struct snd_kcontrol *kcontrol, static const struct snd_kcontrol_new mtk_btcvsd_snd_controls[] = { SOC_ENUM_EXT("BTCVSD Band", btcvsd_enum[0], btcvsd_band_get, btcvsd_band_set), + SOC_SINGLE_BOOL_EXT("BTCVSD Loopback Switch", 0, + btcvsd_loopback_get, btcvsd_loopback_set), SOC_SINGLE_BOOL_EXT("BTCVSD Tx Mute Switch", 0, btcvsd_tx_mute_get, btcvsd_tx_mute_set), SOC_SINGLE_BOOL_EXT("BTCVSD Tx Irq Received Switch", 0, -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: hdmi-codec: fix S/PDIF DAI
by Russell King 03 Mar '19

03 Mar '19

When using the S/PDIF DAI, there is no requirement to call snd_soc_dai_set_fmt() as there is no DAI format definition that defines S/PDIF. In any case, S/PDIF does not have separate clocks, this is embedded into the data stream. Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt to configure TDA998x via the hw_params callback fails as the hdmi_codec_daifmt is left initialised to zero. Since the S/PDIF DAI will only be used by S/PDIF, prepare the hdmi_codec_daifmt structure for this format. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> --- sound/soc/codecs/hdmi-codec.c | 118 +++++++++++++++++++++--------------------- 1 file changed, 59 insertions(+), 59 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index e5b6769b9797..d5f73c837281 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -529,73 +529,71 @@ static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); struct hdmi_codec_daifmt cf = { 0 }; - int ret = 0; dev_dbg(dai->dev, "%s()\n", __func__); - if (dai->id == DAI_ID_SPDIF) { - cf.fmt = HDMI_SPDIF; - } else { - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - cf.bit_clk_master = 1; - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFM: - cf.frame_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBM_CFS: - cf.bit_clk_master = 1; - break; - case SND_SOC_DAIFMT_CBS_CFS: - break; - default: - return -EINVAL; - } + if (dai->id == DAI_ID_SPDIF) + return 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { - case SND_SOC_DAIFMT_NB_NF: - break; - case SND_SOC_DAIFMT_NB_IF: - cf.frame_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_NF: - cf.bit_clk_inv = 1; - break; - case SND_SOC_DAIFMT_IB_IF: - cf.frame_clk_inv = 1; - cf.bit_clk_inv = 1; - break; - } + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - cf.fmt = HDMI_I2S; - break; - case SND_SOC_DAIFMT_DSP_A: - cf.fmt = HDMI_DSP_A; - break; - case SND_SOC_DAIFMT_DSP_B: - cf.fmt = HDMI_DSP_B; - break; - case SND_SOC_DAIFMT_RIGHT_J: - cf.fmt = HDMI_RIGHT_J; - break; - case SND_SOC_DAIFMT_LEFT_J: - cf.fmt = HDMI_LEFT_J; - break; - case SND_SOC_DAIFMT_AC97: - cf.fmt = HDMI_AC97; - break; - default: - dev_err(dai->dev, "Invalid DAI interface format\n"); - return -EINVAL; - } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; } hcp->daifmt[dai->id] = cf; - return ret; + return 0; } static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) @@ -792,8 +790,10 @@ static int hdmi_codec_probe(struct platform_device *pdev) i++; } - if (hcd->spdif) + if (hcd->spdif) { hcp->daidrv[i] = hdmi_spdif_dai; + hcp->daifmt[DAI_ID_SPDIF].fmt = HDMI_SPDIF; + } dev_set_drvdata(dev, hcp); -- 2.7.4

3 2

[alsa-devel] [PATCH 0/5] ASoC: stm32: sai: miscellaneous fixes
by Olivier Moysan 03 Mar '19

03 Mar '19

This patch-set is a collection of fixes for STM32 SAI driver. Olivier Moysan (5): ASoC: stm32: sai: fix iec958 controls indexation ASoC: stm32: sai: fix exposed capabilities in spdif mode ASoC: stm32: sai: fix race condition in irq handler ASoC: stm32: sai: fix oversampling mode ASoC: stm32: sai: fix set_sync service sound/soc/stm/stm32_sai.c | 7 ++++--- sound/soc/stm/stm32_sai_sub.c | 48 ++++++++++++++++++++++++++++++++++--------- 2 files changed, 42 insertions(+), 13 deletions(-) -- 2.7.4

2 10

[alsa-devel] [PATCH] ASoC: cs42xx8: Remove S32_LE in format list
by S.j. Wang 02 Mar '19

02 Mar '19

cs42xx8 is a 24-bit A/D and 24-bit D/A device, so the S32_LE should not be in the supported format list. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..271972337d23 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -32,8 +32,7 @@ #define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_LE) /* codec private data */ struct cs42xx8_priv { -- 1.9.1

4 5

[alsa-devel] [PATCH 3/3] arm64: dts: hi3660-hikey960: Add i2s & sound device
by Pengcheng Li 01 Mar '19

01 Mar '19

From: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Tanglei Han <hantanglei(a)huawei.com> Signed-off-by: Guangke Ji <jiguangke(a)huawei.com> Signed-off-by: Feng Chen <puck.chen(a)hisilicon.com> Signed-off-by: Kaihua Zhong <zhongkaihua(a)huawei.com> Signed-off-by: Jun Chen <chenjun14(a)huawei.com> Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Yiping Xu <xuyiping(a)hisilicon.com> Signed-off-by: Pengcheng Li <lipengcheng8(a)huawei.com> Cc: Wei Xu <xuwei5(a)hisilicon.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> --- arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts | 29 +++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts index 186251f..457cebc 100644 --- a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts +++ b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts @@ -27,6 +27,35 @@ /* rewrite this at bootloader */ reg = <0x0 0x0 0x0 0x0>; }; + + i2s2: hi3660_i2s { + compatible = "hisilicon,hi3660-i2s-1.0"; + reg = <0x0 0xe804f800 0x0 0x400>, + <0x0 0xe804e000 0x0 0x400>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pmx_func &i2s2_cfg_func>; + dmas = <&asp_dmac 18 &asp_dmac 19>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "hikey-hdmi"; + simple-audio-card,format = "i2s"; + + simple-audio-card,bitclock-master = <&sound_master>; + simple-audio-card,frame-master = <&sound_master>; + + sound_master: simple-audio-card,cpu { + sound-dai = <&i2s2>; + }; + + simple-audio-card,codec { + sound-dai = <&adv7533>; + }; + }; + }; &uart5 { -- 2.8.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MOTU MicroBook II
by Manuel Reinhardt 28 Feb '19

28 Feb '19

Add an entry to the quirks-table to for usb-audio to recognize the Microbook II (although it only exposes vendor interfaces). A simple boot quirk is also implemented to set up the sample rate and make sure that no audio urbs are sent before the device is ready. This patch only provides audio playback and capture at 96kHz sample rate. Notice the following shortcomings: - The sample rate is currently hardcoded to 96k although the device also supports 48k and 44.1k. - The various mixer controls of the MicroBook are not made available. - The keep-iface control should be on by default because the device shuts down whenever the altsetting is reset which is usually unwanted. (I don't know the best way to do this) - The communication format used by the MicroBook for sample rate setting and also other setup has been reverse engineered by looking at the usbmon output while running the windows driver through virtualbox. In this patch the first byte of every message is set to \0 while in the observed communications the first byte acts as a "message-counter" increasing its value with every message sent. Leaving it at \0 does not seem to affect the device. Signed-off-by: Manuel Reinhardt <manuel.rhdt(a)gmail.com> --- sound/usb/pcm.c | 4 ++ sound/usb/quirks-table.h | 65 +++++++++++++++++++++++++ sound/usb/quirks.c | 101 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 170 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index db114f3977e0..056af0a57b22 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,6 +354,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + ep = 0x84; + ifnum = 0; + goto add_sync_ep_from_ifnum; } if (attr == USB_ENDPOINT_SYNC_ASYNC && diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b345beb447bd..86e80916a029 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3398,5 +3398,70 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* MOTU Microbook II */ +{ + USB_DEVICE(0x07fd, 0x0004), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MOTU", + .product_name = "MicroBookII", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x84, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x00d8, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x03, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x0120, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f372c624bbf4..e6ce1bbe6ca6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1000,6 +1000,105 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } + +#define MICROBOOK_BUF_SIZE 128 + +static int snd_usb_motu_microbookii_communicate(struct usb_device *dev, u8 *buf, + int buf_size, int *length) +{ + int err, actual_length; + + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x01), buf, *length, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "MicroBookII snd: ", DUMP_PREFIX_NONE, 16, 1, + buf, actual_length, false); + + memset(buf, 0, buf_size); + + err = usb_interrupt_msg(dev, usb_rcvintpipe(dev, 0x82), buf, buf_size, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "MicroBookII rcv: ", DUMP_PREFIX_NONE, 16, 1, + buf, actual_length, false); + + *length = actual_length; + return 0; +} + +static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) +{ + int err, actual_length, poll_attempts = 0; + static const u8 set_samplerate_seq[] = { 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x14, + 0x00, 0x00, 0x00, 0x01 }; + static const u8 poll_ready_seq[] = { 0x00, 0x04, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x18 }; + u8 *buf = kzalloc(MICROBOOK_BUF_SIZE, GFP_KERNEL); + + if (!buf) + return -ENOMEM; + + dev_info(&dev->dev, "Waiting for MOTU Microbook II to boot up...\n"); + + /* First we tell the device which sample rate to use. */ + memcpy(buf, set_samplerate_seq, sizeof(set_samplerate_seq)); + actual_length = sizeof(set_samplerate_seq); + err = snd_usb_motu_microbookii_communicate(dev, buf, MICROBOOK_BUF_SIZE, + &actual_length); + + if (err < 0) { + dev_err(&dev->dev, + "failed setting the sample rate for Motu MicroBook II: %d\n", + err); + goto free_buf; + } + + /* Then we poll every 100 ms until the device informs of its readiness. */ + while (true) { + if (++poll_attempts > 100) { + dev_err(&dev->dev, + "failed booting Motu MicroBook II: timeout\n"); + err = -ENODEV; + goto free_buf; + } + + memset(buf, 0, MICROBOOK_BUF_SIZE); + memcpy(buf, poll_ready_seq, sizeof(poll_ready_seq)); + + actual_length = sizeof(poll_ready_seq); + err = snd_usb_motu_microbookii_communicate( + dev, buf, MICROBOOK_BUF_SIZE, &actual_length); + if (err < 0) { + dev_err(&dev->dev, + "failed booting Motu MicroBook II: communication error %d\n", + err); + goto free_buf; + } + + /* the device signals its readiness through a message of the + * form + * XX 06 00 00 00 00 0b 18 00 00 00 01 + * If the device is not yet ready to accept audio data, the + * last byte of that sequence is 00. + */ + if (actual_length == 12 && buf[actual_length - 1] == 1) + break; + + msleep(100); + } + + dev_info(&dev->dev, "MOTU MicroBook II ready\n"); + +free_buf: + kfree(buf); + return err; +} + /* * Setup quirks */ @@ -1177,6 +1276,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_gamecon780_boot_quirk(dev); case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + return snd_usb_motu_microbookii_boot_quirk(dev); } return 0; -- 2.20.1

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