October 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 1/2] ALSA: docs: Add Marian M2 driver documentation
by Ivan Orlov 05 Nov '23

05 Nov '23

Add documentation for the new MARIAN Seraph M2 sound card. It covers current development status, available controls of the card and information about the integrated loopback. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Remove redundant documentation fix from the next patch in the series Documentation/sound/cards/index.rst | 1 + Documentation/sound/cards/marian-m2.rst | 104 ++++++++++++++++++++++++ 2 files changed, 105 insertions(+) create mode 100644 Documentation/sound/cards/marian-m2.rst diff --git a/Documentation/sound/cards/index.rst b/Documentation/sound/cards/index.rst index e68bbb13c384..e873592d8d00 100644 --- a/Documentation/sound/cards/index.rst +++ b/Documentation/sound/cards/index.rst @@ -19,3 +19,4 @@ Card-Specific Information serial-u16550 img-spdif-in pcmtest + marian-m2 diff --git a/Documentation/sound/cards/marian-m2.rst b/Documentation/sound/cards/marian-m2.rst new file mode 100644 index 000000000000..bf12445e20d7 --- /dev/null +++ b/Documentation/sound/cards/marian-m2.rst @@ -0,0 +1,104 @@ +.. SPDX-License-Identifier: GPL-2.0 + +======================= +MARIAN Seraph M2 Driver +======================= + +Sep 18, 2023 + +Ivan Orlov <ivan.orlov0322(a)gmail.com> + +STATE OF DEVELOPMENT +==================== + +This driver is based on the driver written by Florian Faber in 2012, which seemed to work fine. +However, the initial code contained multiple issues, which had to be solved before sending the +driver upstream. + +The vendor lost the full documentation, so what we have here was recovered from drafts and found +after experiments with the card. + +What seems to be working fine: +- Playback and capture for all supported rates +- Integrated loopback (with some exceptions, see below) + +MEMORY MODEL +============ + +The hardware requires one huge contiguous DMA space to be allocated. After allocation, the bus address of +this buffer should be written to the hardware register. + +We can split this space into two parts: the first one contains samples for capture, another one contains +play samples: + +CAPTURE_CH_0, CAPTURE_CH_1, ..., CAPTURE_CH_127 | PLAY_CH_0, PLAY_CH_1, ..., PLAY_CH_127 + +The card supports the non-interleaved access mode only, so samples for each channel lay together: + +C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 | C0, C0, ..., C0, C1, C1, ..., C1, ..., C127 + +The count of samples per each channel buffer needs to be set explicitly, so the address of the first +byte of the playback data depends on this value. The playback buffer starts where the capture buffer ends. +It makes the arbitrary period count/buffer size feature impossible to implement, and the driver supports only +2 periods per buffer. + +Controls +======== + +Input 1 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal +Input 2 Sync + 0 - No signal, 1 - valid MADI signal found, 2 - Synced with MADI signal + +Input 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Input 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Input 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Input 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Input 1 Frequency + Measured frequency on Input 1 +Input 2 Frequency + Measured frequency on Input 2 + +Output 1 Channel Mode + 0 - 56 channels, 1 - 64 channels +Output 2 Channel Mode + 0 - 56 channels, 1 - 64 channels + +Output 1 Frame Mode + 0 - 48 kHz, 1 - 96 kHz +Output 2 Frame Mode + 0 - 48 kHz, 1 - 96 kHz + +Clock Source + Internal/Sync Bus, Input Port 1, Input Port 2 + +Speed mode + VCO clock range (slow/fast) + +DCO Frequency (Hz) + +DCO Frequency (ms) + +DCO Detune + Range: -200...200 + +Loopback + Enable/Disable integrated loopback + + +Loopback +======== + +The card contains integrated loopback. When it is enabled, it sets the hardware’s DAW-in memory pointer +to the hardware’s DAW-out memory. So, what you play is what you record. + +You can enable the integrated loopback using the corresponding control. + +The loopback seems to work well on lower rates (like 28000). However, when the rate goes higher, I observe +multiple mistakes in recorded byte ordering. -- 2.34.1

5 12

[PATCH AUTOSEL 5.15 13/28] ASoC: codecs: wsa-macro: handle component name prefix
by Sasha Levin 04 Nov '23

04 Nov '23

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit c29e5263d32a6d0ec094d425ae7fef3fa8d4da1c ] When comparing widget names in wsa_macro_spk_boost_event(), consider also the component's name prefix. Otherwise the WSA codec won't have proper mixer setup resulting in no sound playback through speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://lore.kernel.org/r/20231003155710.821315-3-krzysztof.kozlowski@linar… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-wsa-macro.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index dd1a8b7bc794c..643ddf7dd6c5d 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1668,12 +1668,12 @@ static int wsa_macro_spk_boost_event(struct snd_soc_dapm_widget *w, u16 boost_path_ctl, boost_path_cfg1; u16 reg, reg_mix; - if (!strcmp(w->name, "WSA_RX INT0 CHAIN")) { + if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT0 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST0_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX0_RX_PATH_CFG1; reg = CDC_WSA_RX0_RX_PATH_CTL; reg_mix = CDC_WSA_RX0_RX_PATH_MIX_CTL; - } else if (!strcmp(w->name, "WSA_RX INT1 CHAIN")) { + } else if (!snd_soc_dapm_widget_name_cmp(w, "WSA_RX INT1 CHAIN")) { boost_path_ctl = CDC_WSA_BOOST1_BOOST_PATH_CTL; boost_path_cfg1 = CDC_WSA_RX1_RX_PATH_CFG1; reg = CDC_WSA_RX1_RX_PATH_CTL; -- 2.42.0

2 2

[PATCH] ASoC: rt5682s: Add LDO control for dacref
by Jack Yu 01 Nov '23

01 Nov '23

Add LDO control for dacref. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 12 ++++++++++++ sound/soc/codecs/rt5682s.h | 3 +++ 2 files changed, 15 insertions(+) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index c261c33c4be7..c05996e8fc5a 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1017,6 +1017,15 @@ static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); static const DECLARE_TLV_DB_SCALE(cbj_bst_tlv, -1200, 150, 0); +/* LDO output select */ +static const char * const rt5682s_ldo_output_select[] = { + "1.607V", "1.5V", "1.406V", "1.731V" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682s_ldo_output_enum, RT5682S_BIAS_CUR_CTRL_7, + RT5682S_DVO_LDO_DACREF_SFT, rt5682s_ldo_output_select); + + static const struct snd_kcontrol_new rt5682s_snd_controls[] = { /* DAC Digital Volume */ SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5682S_DAC1_DIG_VOL, @@ -1035,6 +1044,9 @@ static const struct snd_kcontrol_new rt5682s_snd_controls[] = { /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5682S_STO1_ADC_BOOST, RT5682S_STO1_ADC_L_BST_SFT, RT5682S_STO1_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + + /* LDO Output Select Control */ + SOC_ENUM("LDO Output Select", rt5682s_ldo_output_enum), }; /** diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 1d79d432d0d8..0ae5a35825f4 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1263,6 +1263,9 @@ #define RT5682S_JDH_NO_PLUG (0x1 << 4) #define RT5682S_JDH_PLUG (0x0 << 4) +/* Bias current control 7 (0x0110) */ +#define RT5682S_DVO_LDO_DACREF_SFT 4 + /* Charge Pump Internal Register1 (0x0125) */ #define RT5682S_CP_CLK_HP_MASK (0x3 << 4) #define RT5682S_CP_CLK_HP_100KHZ (0x0 << 4) -- 2.34.1

2 5

[PATCH 0/3]
by Zhu Ning 01 Nov '23

01 Nov '23

*** BLURB HERE *** Hi , 3 patches here for the es8326 driver... We developed a new version of the chip. 3 Three patches are used for compatibility with the old and new versions of the chip. The test results from the test department met our expectations Thanks, Zhuning. Zhu Ning (3): ASoC: codecs: ES8326: Add chip version flag ASoC: codecs: ES8326: Changing initialisation and broadcasting ASoC: codecs: ES8326: Changing the headset detection time sound/soc/codecs/es8326.c | 219 ++++++++++++++++++++++---------------- 1 file changed, 127 insertions(+), 92 deletions(-) mode change 100644 => 100755 sound/soc/codecs/es8326.c -- 2.17.1

3 7

[PATCH next] ALSA: hda: cs35l41: Fix missing error code in cs35l41_smart_amp()
by Harshit Mogalapalli 31 Oct '23

31 Oct '23

When firmware status is invalid, assign -EINVAL to ret as ret is '0' at that point and returning success is incorrect when firmware status is invalid. Fixes: a51d8ba03a4f ("ALSA: hda: cs35l41: Check CSPL state after loading firmware") Signed-off-by: Harshit Mogalapalli <harshit.m.mogalapalli(a)oracle.com> --- This is found using smatch and only compile tested. --- sound/pci/hda/cs35l41_hda.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 496ff6a9d300..09f8d8fa4c71 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1042,6 +1042,7 @@ static int cs35l41_smart_amp(struct cs35l41_hda *cs35l41) default: dev_err(cs35l41->dev, "Firmware status is invalid: %u\n", fw_status); + ret = -EINVAL; goto clean_dsp; } -- 2.39.3

1 0

[PATCH v4] ALSA: virtio: use ack callback
by Matias Ezequiel Vara Larsen 31 Oct '23

31 Oct '23

This commit uses the ack() callback to determine when a buffer has been updated, then exposes it to guest. The current mechanism splits a dma buffer into descriptors that are exposed to the device. This dma buffer is shared with the user application. When the device consumes a buffer, the driver moves the request from the used ring to available ring. The driver exposes the buffer to the device without knowing if the content has been updated from the user. The section 2.8.21.1 of the virtio spec states that: "The device MAY access the descriptor chains the driver created and the memory they refer to immediately". If the device picks up buffers from the available ring just after it is notified, it happens that the content may be old. When the ack() callback is invoked, the driver exposes only the buffers that have already been updated, i.e., enqueued in the available ring. Thus, the device always picks up a buffer that is updated. For capturing, the driver starts by exposing all the available buffers to device. After device updates the content of a buffer, it enqueues it in the used ring. It is only after the ack() for capturing is issued that the driver re-enqueues the buffer in the available ring. Co-developed-by: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Signed-off-by: Anton Yakovlev <anton.yakovlev(a)opensynergy.com> Signed-off-by: Matias Ezequiel Vara Larsen <mvaralar(a)redhat.com> --- Changelog: v3 -> v4: * Disable rewinds * Add SNDRV_PCM_INFO_SYNC_APPLPTR flag * Use pcm indirect API for pointer() and ack() callbacks v2 -> v3: * Use ack() callback instead of copy()/fill_silence() * Change commit name * Rewrite commit message * Remove virtsnd_pcm_msg_send_locked() * Use single callback for both capture and playback * Fix kernel test robot warnings regarding documentation * v2 patch at: https://lore.kernel.org/lkml/87y1fzkq8c.wl-tiwai@suse.de/T/ v1 -> v2: * Use snd_pcm_set_managed_buffer_all()for buffer allocation/freeing. * Make virtsnd_pcm_msg_send() generic by specifying the offset and size for the modified part of the buffer; this way no assumptions need to be made. * Disable SNDRV_PCM_INFO_NO_REWINDS since now only sequential reading/writing of frames is supported. * Correct comment at virtsnd_pcm_msg_send(). * v1 patch at: https://lore.kernel.org/lkml/20231016151000.GE119987@fedora/t/ sound/virtio/virtio_pcm.c | 6 +- sound/virtio/virtio_pcm.h | 9 ++- sound/virtio/virtio_pcm_msg.c | 79 ++++++++++++--------- sound/virtio/virtio_pcm_ops.c | 125 +++++++++++++++++++++++++++------- 4 files changed, 158 insertions(+), 61 deletions(-) diff --git a/sound/virtio/virtio_pcm.c b/sound/virtio/virtio_pcm.c index c10d91fff2fb..967e4c45be9b 100644 --- a/sound/virtio/virtio_pcm.c +++ b/sound/virtio/virtio_pcm.c @@ -109,7 +109,9 @@ static int virtsnd_pcm_build_hw(struct virtio_pcm_substream *vss, SNDRV_PCM_INFO_BATCH | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_PAUSE; + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_REWINDS | + SNDRV_PCM_INFO_SYNC_APPLPTR; if (!info->channels_min || info->channels_min > info->channels_max) { dev_err(&vdev->dev, @@ -471,7 +473,7 @@ int virtsnd_pcm_build_devs(struct virtio_snd *snd) for (kss = ks->substream; kss; kss = kss->next) vs->substreams[kss->number]->substream = kss; - snd_pcm_set_ops(vpcm->pcm, i, &virtsnd_pcm_ops); + snd_pcm_set_ops(vpcm->pcm, i, &virtsnd_pcm_ops[i]); } snd_pcm_set_managed_buffer_all(vpcm->pcm, diff --git a/sound/virtio/virtio_pcm.h b/sound/virtio/virtio_pcm.h index 062eb8e8f2cf..5dd1b43b9493 100644 --- a/sound/virtio/virtio_pcm.h +++ b/sound/virtio/virtio_pcm.h @@ -9,6 +9,7 @@ #include <linux/atomic.h> #include <linux/virtio_config.h> #include <sound/pcm.h> +#include <sound/pcm-indirect.h> struct virtio_pcm; struct virtio_pcm_msg; @@ -21,6 +22,7 @@ struct virtio_pcm_msg; * @direction: Stream data flow direction (SNDRV_PCM_STREAM_XXX). * @features: Stream VirtIO feature bit map (1 << VIRTIO_SND_PCM_F_XXX). * @substream: Kernel ALSA substream. + * @pcm_indirect: Kernel indirect pcm structure. * @hw: Kernel ALSA substream hardware descriptor. * @elapsed_period: Kernel work to handle the elapsed period state. * @lock: Spinlock that protects fields shared by interrupt handlers and @@ -46,6 +48,7 @@ struct virtio_pcm_substream { u32 direction; u32 features; struct snd_pcm_substream *substream; + struct snd_pcm_indirect pcm_indirect; struct snd_pcm_hardware hw; struct work_struct elapsed_period; spinlock_t lock; @@ -57,7 +60,6 @@ struct virtio_pcm_substream { bool suspended; struct virtio_pcm_msg **msgs; unsigned int nmsgs; - int msg_last_enqueued; unsigned int msg_count; wait_queue_head_t msg_empty; }; @@ -90,7 +92,7 @@ struct virtio_pcm { struct virtio_pcm_stream streams[SNDRV_PCM_STREAM_LAST + 1]; }; -extern const struct snd_pcm_ops virtsnd_pcm_ops; +extern const struct snd_pcm_ops virtsnd_pcm_ops[]; int virtsnd_pcm_validate(struct virtio_device *vdev); @@ -117,7 +119,8 @@ int virtsnd_pcm_msg_alloc(struct virtio_pcm_substream *vss, void virtsnd_pcm_msg_free(struct virtio_pcm_substream *vss); -int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss); +int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss, unsigned long offset, + unsigned long bytes); unsigned int virtsnd_pcm_msg_pending_num(struct virtio_pcm_substream *vss); diff --git a/sound/virtio/virtio_pcm_msg.c b/sound/virtio/virtio_pcm_msg.c index aca2dc1989ba..542446c4c7ba 100644 --- a/sound/virtio/virtio_pcm_msg.c +++ b/sound/virtio/virtio_pcm_msg.c @@ -155,7 +155,6 @@ int virtsnd_pcm_msg_alloc(struct virtio_pcm_substream *vss, sizeof(msg->xfer)); sg_init_one(&msg->sgs[PCM_MSG_SG_STATUS], &msg->status, sizeof(msg->status)); - msg->length = period_bytes; virtsnd_pcm_sg_from(&msg->sgs[PCM_MSG_SG_DATA], sg_num, data, period_bytes); @@ -186,61 +185,75 @@ void virtsnd_pcm_msg_free(struct virtio_pcm_substream *vss) /** * virtsnd_pcm_msg_send() - Send asynchronous I/O messages. * @vss: VirtIO PCM substream. + * @offset: starting position that has been updated + * @bytes: number of bytes that has been updated * * All messages are organized in an ordered circular list. Each time the * function is called, all currently non-enqueued messages are added to the - * virtqueue. For this, the function keeps track of two values: - * - * msg_last_enqueued = index of the last enqueued message, - * msg_count = # of pending messages in the virtqueue. + * virtqueue. For this, the function uses offset and bytes to calculate the + * messages that need to be added. * * Context: Any context. Expects the tx/rx queue and the VirtIO substream * spinlocks to be held by caller. * Return: 0 on success, -errno on failure. */ -int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss) +int virtsnd_pcm_msg_send(struct virtio_pcm_substream *vss, unsigned long offset, + unsigned long bytes) { - struct snd_pcm_runtime *runtime = vss->substream->runtime; struct virtio_snd *snd = vss->snd; struct virtio_device *vdev = snd->vdev; struct virtqueue *vqueue = virtsnd_pcm_queue(vss)->vqueue; - int i; - int n; + unsigned long period_bytes = snd_pcm_lib_period_bytes(vss->substream); + unsigned long start, end, i; + unsigned int msg_count = vss->msg_count; bool notify = false; + int rc; - i = (vss->msg_last_enqueued + 1) % runtime->periods; - n = runtime->periods - vss->msg_count; + start = offset / period_bytes; + end = (offset + bytes - 1) / period_bytes; - for (; n; --n, i = (i + 1) % runtime->periods) { + for (i = start; i <= end; i++) { struct virtio_pcm_msg *msg = vss->msgs[i]; struct scatterlist *psgs[] = { &msg->sgs[PCM_MSG_SG_XFER], &msg->sgs[PCM_MSG_SG_DATA], &msg->sgs[PCM_MSG_SG_STATUS] }; - int rc; - - msg->xfer.stream_id = cpu_to_le32(vss->sid); - memset(&msg->status, 0, sizeof(msg->status)); - - if (vss->direction == SNDRV_PCM_STREAM_PLAYBACK) - rc = virtqueue_add_sgs(vqueue, psgs, 2, 1, msg, - GFP_ATOMIC); - else - rc = virtqueue_add_sgs(vqueue, psgs, 1, 2, msg, - GFP_ATOMIC); - - if (rc) { - dev_err(&vdev->dev, - "SID %u: failed to send I/O message\n", - vss->sid); - return rc; + unsigned long n; + + n = period_bytes - (offset % period_bytes); + if (n > bytes) + n = bytes; + + msg->length += n; + if (msg->length == period_bytes) { + msg->xfer.stream_id = cpu_to_le32(vss->sid); + memset(&msg->status, 0, sizeof(msg->status)); + + if (vss->direction == SNDRV_PCM_STREAM_PLAYBACK) + rc = virtqueue_add_sgs(vqueue, psgs, 2, 1, msg, + GFP_ATOMIC); + else + rc = virtqueue_add_sgs(vqueue, psgs, 1, 2, msg, + GFP_ATOMIC); + + if (rc) { + dev_err(&vdev->dev, + "SID %u: failed to send I/O message\n", + vss->sid); + return rc; + } + + vss->msg_count++; } - vss->msg_last_enqueued = i; - vss->msg_count++; + offset = 0; + bytes -= n; } + if (msg_count == vss->msg_count) + return 0; + if (!(vss->features & (1U << VIRTIO_SND_PCM_F_MSG_POLLING))) notify = virtqueue_kick_prepare(vqueue); @@ -309,6 +322,8 @@ static void virtsnd_pcm_msg_complete(struct virtio_pcm_msg *msg, if (vss->hw_ptr >= vss->buffer_bytes) vss->hw_ptr -= vss->buffer_bytes; + msg->length = 0; + vss->xfer_xrun = false; vss->msg_count--; @@ -320,8 +335,6 @@ static void virtsnd_pcm_msg_complete(struct virtio_pcm_msg *msg, le32_to_cpu(msg->status.latency_bytes)); schedule_work(&vss->elapsed_period); - - virtsnd_pcm_msg_send(vss); } else if (!vss->msg_count) { wake_up_all(&vss->msg_empty); } diff --git a/sound/virtio/virtio_pcm_ops.c b/sound/virtio/virtio_pcm_ops.c index f8bfb87624be..ad12aae52fc3 100644 --- a/sound/virtio/virtio_pcm_ops.c +++ b/sound/virtio/virtio_pcm_ops.c @@ -282,7 +282,6 @@ static int virtsnd_pcm_prepare(struct snd_pcm_substream *substream) vss->buffer_bytes = snd_pcm_lib_buffer_bytes(substream); vss->hw_ptr = 0; - vss->msg_last_enqueued = -1; } else { struct snd_pcm_runtime *runtime = substream->runtime; unsigned int buffer_bytes = snd_pcm_lib_buffer_bytes(substream); @@ -300,6 +299,11 @@ static int virtsnd_pcm_prepare(struct snd_pcm_substream *substream) vss->suspended = false; vss->msg_count = 0; + memset(&vss->pcm_indirect, 0, sizeof(vss->pcm_indirect)); + vss->pcm_indirect.sw_buffer_size = + vss->pcm_indirect.hw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + msg = virtsnd_pcm_ctl_msg_alloc(vss, VIRTIO_SND_R_PCM_PREPARE, GFP_KERNEL); if (!msg) @@ -324,7 +328,7 @@ static int virtsnd_pcm_trigger(struct snd_pcm_substream *substream, int command) struct virtio_snd_queue *queue; struct virtio_snd_msg *msg; unsigned long flags; - int rc; + int rc = 0; switch (command) { case SNDRV_PCM_TRIGGER_START: @@ -333,7 +337,8 @@ static int virtsnd_pcm_trigger(struct snd_pcm_substream *substream, int command) spin_lock_irqsave(&queue->lock, flags); spin_lock(&vss->lock); - rc = virtsnd_pcm_msg_send(vss); + if (vss->direction == SNDRV_PCM_STREAM_CAPTURE) + rc = virtsnd_pcm_msg_send(vss, 0, vss->buffer_bytes); if (!rc) vss->xfer_enabled = true; spin_unlock(&vss->lock); @@ -428,37 +433,111 @@ static int virtsnd_pcm_sync_stop(struct snd_pcm_substream *substream) } /** - * virtsnd_pcm_pointer() - Get the current hardware position for the PCM - * substream. + * virtsnd_pcm_pb_pointer() - Get the current hardware position for the PCM + * substream for playback. * @substream: Kernel ALSA substream. * - * Context: Any context. Takes and releases the VirtIO substream spinlock. + * Context: Any context. * Return: Hardware position in frames inside [0 ... buffer_size) range. */ static snd_pcm_uframes_t -virtsnd_pcm_pointer(struct snd_pcm_substream *substream) +virtsnd_pcm_pb_pointer(struct snd_pcm_substream *substream) +{ + struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); + + return snd_pcm_indirect_playback_pointer(substream, + &vss->pcm_indirect, + vss->hw_ptr); +} + +/** + * virtsnd_pcm_cp_pointer() - Get the current hardware position for the PCM + * substream for capture. + * @substream: Kernel ALSA substream. + * + * Context: Any context. + * Return: Hardware position in frames inside [0 ... buffer_size) range. + */ +static snd_pcm_uframes_t +virtsnd_pcm_cp_pointer(struct snd_pcm_substream *substream) +{ + struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); + + return snd_pcm_indirect_capture_pointer(substream, + &vss->pcm_indirect, + vss->hw_ptr); +} + +static void virtsnd_pcm_trans_copy(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) { struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); - snd_pcm_uframes_t hw_ptr = SNDRV_PCM_POS_XRUN; + + virtsnd_pcm_msg_send(vss, rec->sw_data, bytes); +} + +static int virtsnd_pcm_pb_ack(struct snd_pcm_substream *substream) +{ + struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); + struct virtio_snd_queue *queue = virtsnd_pcm_queue(vss); + unsigned long flags; + int rc; + + spin_lock_irqsave(&queue->lock, flags); + spin_lock(&vss->lock); + + rc = snd_pcm_indirect_playback_transfer(substream, &vss->pcm_indirect, + virtsnd_pcm_trans_copy); + + spin_unlock(&vss->lock); + spin_unlock_irqrestore(&queue->lock, flags); + + return rc; +} + +static int virtsnd_pcm_cp_ack(struct snd_pcm_substream *substream) +{ + struct virtio_pcm_substream *vss = snd_pcm_substream_chip(substream); + struct virtio_snd_queue *queue = virtsnd_pcm_queue(vss); unsigned long flags; + int rc; + + spin_lock_irqsave(&queue->lock, flags); + spin_lock(&vss->lock); + + rc = snd_pcm_indirect_capture_transfer(substream, &vss->pcm_indirect, + virtsnd_pcm_trans_copy); - spin_lock_irqsave(&vss->lock, flags); - if (!vss->xfer_xrun) - hw_ptr = bytes_to_frames(substream->runtime, vss->hw_ptr); - spin_unlock_irqrestore(&vss->lock, flags); + spin_unlock(&vss->lock); + spin_unlock_irqrestore(&queue->lock, flags); - return hw_ptr; + return rc; } /* PCM substream operators map. */ -const struct snd_pcm_ops virtsnd_pcm_ops = { - .open = virtsnd_pcm_open, - .close = virtsnd_pcm_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = virtsnd_pcm_hw_params, - .hw_free = virtsnd_pcm_hw_free, - .prepare = virtsnd_pcm_prepare, - .trigger = virtsnd_pcm_trigger, - .sync_stop = virtsnd_pcm_sync_stop, - .pointer = virtsnd_pcm_pointer, +const struct snd_pcm_ops virtsnd_pcm_ops[] = { + { + .open = virtsnd_pcm_open, + .close = virtsnd_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = virtsnd_pcm_hw_params, + .hw_free = virtsnd_pcm_hw_free, + .prepare = virtsnd_pcm_prepare, + .trigger = virtsnd_pcm_trigger, + .sync_stop = virtsnd_pcm_sync_stop, + .pointer = virtsnd_pcm_pb_pointer, + .ack = virtsnd_pcm_pb_ack, + }, + { + .open = virtsnd_pcm_open, + .close = virtsnd_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = virtsnd_pcm_hw_params, + .hw_free = virtsnd_pcm_hw_free, + .prepare = virtsnd_pcm_prepare, + .trigger = virtsnd_pcm_trigger, + .sync_stop = virtsnd_pcm_sync_stop, + .pointer = virtsnd_pcm_cp_pointer, + .ack = virtsnd_pcm_cp_ack, + }, }; base-commit: 8a749fd1a8720d4619c91c8b6e7528c0a355c0aa -- 2.41.0

4 4

[PATCH alsa-ucm-conf 1/2] bytcr-wm5102: Add support for speakers connected to HPOUT2
by Hans de Goede 31 Oct '23

31 Oct '23

On the Lenovo Yoga Tab 3 YT3-X90 the speaker amplifiers are connected to the HPOUT2 output pins instead of the the SPK output pins. Check for "cfg-spk:hpout2" in the components string which indicates this setup and add a ucm2/codecs/wm5102/HPOut2-Speaker.conf file for this setup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/Intel/bytcr-wm5102/HiFi.conf | 11 ++++++++++- ucm2/codecs/wm5102/EnableSeq.conf | 2 ++ ucm2/codecs/wm5102/HPOut2-Speaker.conf | 22 ++++++++++++++++++++++ 3 files changed, 34 insertions(+), 1 deletion(-) create mode 100644 ucm2/codecs/wm5102/HPOut2-Speaker.conf diff --git a/ucm2/Intel/bytcr-wm5102/HiFi.conf b/ucm2/Intel/bytcr-wm5102/HiFi.conf index 1269868..2afc757 100644 --- a/ucm2/Intel/bytcr-wm5102/HiFi.conf +++ b/ucm2/Intel/bytcr-wm5102/HiFi.conf @@ -21,7 +21,16 @@ SectionVerb { } } -Include.spk.File "/codecs/wm5102/Speaker.conf" +If.spk { + Condition { + Type String + Haystack "${CardComponents}" + Needle "cfg-spk:hpout2" + } + True.Include.hpout2.File "/codecs/wm5102/HPOut2-Speaker.conf" + False.Include.spk.File "/codecs/wm5102/Speaker.conf" +} + Include.hp.File "/codecs/wm5102/HeadPhones.conf" Include.mic.File "/codecs/wm5102/IN3-InternalMic.conf" Include.headset.File "/codecs/wm5102/IN1-HeadsetMic.conf" diff --git a/ucm2/codecs/wm5102/EnableSeq.conf b/ucm2/codecs/wm5102/EnableSeq.conf index 6a857a2..87cb3bb 100644 --- a/ucm2/codecs/wm5102/EnableSeq.conf +++ b/ucm2/codecs/wm5102/EnableSeq.conf @@ -4,6 +4,8 @@ EnableSequence [ cset "name='SPKOUTR Input 1' AIF1RX2" cset "name='HPOUT1L Input 1' AIF1RX1" cset "name='HPOUT1R Input 1' AIF1RX2" + cset "name='HPOUT2L Input 1' AIF1RX1" + cset "name='HPOUT2R Input 1' AIF1RX2" # Both mics are quite soft by default, boost then cset "name='IN1L Volume' 28" diff --git a/ucm2/codecs/wm5102/HPOut2-Speaker.conf b/ucm2/codecs/wm5102/HPOut2-Speaker.conf new file mode 100644 index 0000000..375f9ba --- /dev/null +++ b/ucm2/codecs/wm5102/HPOut2-Speaker.conf @@ -0,0 +1,22 @@ +SectionDevice."Speaker" { + Comment "Speakers" + + Value { + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] + + EnableSequence [ + cset "name='HPOUT2 Digital Switch' on" + cset "name='Speaker Switch' on" + ] + + DisableSequence [ + cset "name='HPOUT2 Digital Switch' off" + cset "name='Speaker Switch' off" + ] +} -- 2.41.0

2 3

[PATCH alsa-ucm-conf] chtnau8824: Mono speaker fixes
by Hans de Goede 31 Oct '23

31 Oct '23

2 mono speaker setup fixes: 1. Use the kernel's components string to check for a mono-speaker device when the board has a components string. 2. So far known nau8824 boards with a mono speaker where using the right speaker channel, which is unusual. Normally mono speaker setups use only the left speaker channel. The Cyberbook T116 tablet is a nau8824 based model, which indeed uses the left speaker channel for its single speaker. Modify ucm2/codecs/nau8824/MonoSpeaker.conf to send a left+right channel mix to both speaker channels, so that things will work independent of which speaker channel is used for a mono setup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/Intel/chtnau8824/HiFi.conf | 15 +++++++++++++++ ucm2/codecs/nau8824/MonoSpeaker.conf | 7 ++++--- 2 files changed, 19 insertions(+), 3 deletions(-) diff --git a/ucm2/Intel/chtnau8824/HiFi.conf b/ucm2/Intel/chtnau8824/HiFi.conf index c3d830d..1a1faf4 100644 --- a/ucm2/Intel/chtnau8824/HiFi.conf +++ b/ucm2/Intel/chtnau8824/HiFi.conf @@ -14,6 +14,21 @@ If.Controls { } } +# Figure out which components are in use on the device, we check both the +# components string (present on newer kernels) as well as checking for DMI +# strings for compatibility with older kernels. Note DMI matches for new +# models should only be added to the kernel, this UCM profile will then +# automatically pick up the info from the components string. + +If.components-mono-spk { + Condition { + Type String + Haystack "${CardComponents}" + Needle "cfg-spk:1" + } + True.Define.Speaker "MonoSpeaker" +} + If.cfg-mspk { Condition { Type RegexMatch diff --git a/ucm2/codecs/nau8824/MonoSpeaker.conf b/ucm2/codecs/nau8824/MonoSpeaker.conf index 6b4ef8a..5428133 100644 --- a/ucm2/codecs/nau8824/MonoSpeaker.conf +++ b/ucm2/codecs/nau8824/MonoSpeaker.conf @@ -11,9 +11,10 @@ SectionDevice."Speaker" { ] EnableSequence [ - # nau8824 mono speaker boards have the speaker on the right chan - cset "name='Speaker Left DACL Volume' 0" - cset "name='Speaker Left DACR Volume' 0" + # Some nau8824 mono speaker boards have the speaker on the right chan + # others on the left, enable output of both channels on both speakers + cset "name='Speaker Left DACL Volume' 1" + cset "name='Speaker Left DACR Volume' 1" cset "name='Speaker Right DACL Volume' 1" cset "name='Speaker Right DACR Volume' 1" cset "name='Ext Spk Switch' on" -- 2.41.0

2 1

[PATCH alsa-ucm-conf] codecs/es8316: Fix mono speaker settings from previous boot getting applied
by Hans de Goede 31 Oct '23

31 Oct '23

When codecs/es8316/MonoSpeaker.conf's EnableSeq was run last before shutdown the 'Playback Polarity' and 'DAC Mono Mix Switch' controls for Mono Speaker output get saved and restored on the next boot. These settings are undone by the MonoSpeaker DisableSeq, but that never runs when an updated kernel switches from the default "cfg-spk:1" in components to "cfg-spk:2" for devices which do actually have 2 speakers, which causes the ucm profile to use codecs/es8316/Speaker.conf instead. Set the 'Playback Polarity' and 'DAC Mono Mix Switch' controls to their defaults in codecs/es8316/EnableSeq.conf to avoid them getting stuck in Mono mix mode in this scenario. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/es8316/EnableSeq.conf | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/ucm2/codecs/es8316/EnableSeq.conf b/ucm2/codecs/es8316/EnableSeq.conf index b6c81d8..b75c4a7 100644 --- a/ucm2/codecs/es8316/EnableSeq.conf +++ b/ucm2/codecs/es8316/EnableSeq.conf @@ -22,4 +22,8 @@ EnableSequence [ # Setup muxes / switches cset "name='Left Headphone Mixer Left DAC Switch' on" cset "name='Right Headphone Mixer Right DAC Switch' on" + + # Undo MonoSpeaker settings which may have been saved on shutdown + cset "name='Playback Polarity' 0" + cset "name='DAC Mono Mix Switch' off" ] -- 2.41.0

2 2

[PATCH v3] ASoC: tas2783: Add source files for tas2783 driver.
by Baojun Xu 31 Oct '23

31 Oct '23

Add source file and header file for tas2783 soundwire driver. Also update Kconfig and Makefile for tas2783 driver. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v3: - change spelling for description in sound/soc/codecs/Kconfig. - add select REGMAP_SOUNDWIRE for TAS2783 in Kconfig. - remove delay.h, device.h, of_gpio.h, interrupt.h - change array tas2783_cali_reg to fixed size. - add data port registers area in tas2783_readable_register. - add comments for volatile register(reset). - change comment for volume get. - change all private struct variable name to tas_priv. - add description for why have calibration data in UEFI. - remove firmware data save. - add left, right channel setting after firmware load. - move pm_runtime functions to driver from component probe. - move firmware request to driver from component probe. - change name to component from codec. - change firmware binary name to tas2783-x.bin. - remove start_addr in struct tas2783_firmware_node. - remove firmware_node, codec in struct tasdevice_priv. - remove TAS2783_MAX_NO_NODES define. --- sound/soc/codecs/Kconfig | 14 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 857 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 100 ++++ 4 files changed, 973 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f1e1dbc509f6..2973fe8975fd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -244,6 +244,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1803,6 +1804,19 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2783 driver implements a flexible and configurable + algorithm cofficient setting, for one, two or multiple TAS2783 + chips. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a87e56938ce5..208f76a653fa 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -280,6 +280,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -656,6 +657,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..d9719f15b17c --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,856 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/of.h> +#include <sound/pcm_params.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, + TAS2783_CALIBRATION_RE_LOW, + TAS2783_CALIBRATION_INV_RE, + TAS2783_CALIBRATION_POW, + TAS2783_CALIBRATION_TLIMIT +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a}, /* AMP inactive. */ + { 0x8097, 0xc8}, + { 0x80b5, 0x74}, + { 0x8099, 0x20}, + { 0xfe8d, 0x0d}, + { 0xfebe, 0x4a}, + { 0x8230, 0x00}, + { 0x8231, 0x00}, + { 0x8232, 0x00}, + { 0x8233, 0x01}, + { 0x8418, 0x00}, /* Set volume to 0 dB. */ + { 0x8419, 0x00}, + { 0x841a, 0x00}, + { 0x841b, 0x00}, + { 0x8428, 0x40}, /* Unmute channel */ + { 0x8429, 0x00}, + { 0x842a, 0x00}, + { 0x842b, 0x00}, + { 0x8548, 0x00}, /* Set volume to 0 dB. */ + { 0x8549, 0x00}, + { 0x854a, 0x00}, + { 0x854b, 0x00}, + { 0x8558, 0x40}, /* Unmute channel */ + { 0x8559, 0x00}, + { 0x855a, 0x00}, + { 0x855b, 0x00}, + { 0x800a, 0x3a}, /* Enable both channel */ + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x805c, 0x99}, + { 0x40400088, 0}, /* FUNC_1, FU21, SEL_1(Mute) */ + { 0x40400090, 0}, /* FUNC_1, FU21, SEL_2(Channel volume) */ + { 0x40400108, 0}, /* FUNC_1, FU23, MUTE */ +}; + +static bool tas2783_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x000 ... 0x080: /* Data port 0. */ + case 0x100 ... 0x140: /* Data port 1. */ + case 0x200 ... 0x240: /* Data port 2. */ + case 0x300 ... 0x340: /* Data port 3. */ + case 0x400 ... 0x440: /* Data port 4. */ + case 0x500 ... 0x540: /* Data port 5. */ + case 0x8000 ... 0xc000: /* Page 0 ~ 127. */ + case 0xfe80 ... 0xfeff: /* Page 253. */ + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_UDMPU21, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, TAS2783_DEVICE_CHANNEL_LEFT): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, TAS2783_DEVICE_CHANNEL_RIGHT): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE22, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x8001: + /* Only reset register was volatiled. + * Write any value into this register, mean RESET device. + */ + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x41008000 + TASDEVICE_REG(0xa1, 0x60, 0x7f), + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + /* Keep in valid area, out of range value don't care. */ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val = 0, ret; + + if (!map || !ucontrol) { + dev_err(tas_dev->dev, "%s, wrong regmap.\n", __func__); + return -EINVAL; + } + /* Read current volume from the device. */ + ret = regmap_read(map, mc->reg, &val); + if (ret) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + return ret; + } + ucontrol->value.integer.value[0] = + tasdevice_clamp(val, mc->max, mc->invert); + + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val, ret; + + if (!map || !ucontrol) { + dev_err(tas_dev->dev, "%s, wrong regmap.\n", __func__); + return -EINVAL; + } + val = tasdevice_clamp(ucontrol->value.integer.value[0], + mc->max, mc->invert); + + ret = regmap_write(map, mc->reg, val); + if (ret != 0) { + dev_dbg(tas_dev->dev, "%s, Put vol %d into %x %x.\n", + __func__, val, mc->reg, ret); + } + + return ret; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask = 0; + int ret = 0, val = 0; + + if (!map || !ucontrol) { + dev_err(tas_dev->dev, "%s, wrong regmap.\n", __func__); + return -EINVAL; + } + /* Read current volume from the device. */ + ret = regmap_read(map, mc->reg, &val); + if (ret != 0) { + dev_err(tas_dev->dev, "%s get AMP vol from %x with %d.\n", + __func__, mc->reg, ret); + return ret; + } + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + ucontrol->value.integer.value[0] = tasdevice_clamp(val, mc->max, + mc->invert); + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask; + int val, ret; + + if (!map || !ucontrol) { + dev_err(tas_dev->dev, "%s, wrong regmap.\n", __func__); + return -EINVAL; + } + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], mc->max, + mc->invert); + ret = regmap_update_bits(map, mc->reg, mask, val << mc->shift); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + } + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib( + struct tasdevice_priv *tas_dev, unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *reg_start; + int ret; + + if (!map) { + dev_err(tas_dev->dev, "%s, wrong regmap.\n", __func__); + return; + } + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s, slaver doesn't exist.\n", + __func__); + return; + } + if ((tas_dev->sdw_peripheral->id.unique_id < TAS2783_ID_MIN) || + (tas_dev->sdw_peripheral->id.unique_id > TAS2783_ID_MAX)) + return; + reg_start = (u8 *)(cali_data+(tas_dev->sdw_peripheral->id.unique_id + - TAS2783_ID_MIN)*sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + reg_start + i, 4); + if (ret != 0) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +static int tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_dev->tm; + unsigned int attr = 0, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_dev->cali_data.total_sz = 128; + /* Sometimes, calibration was performed from Windows, + * and data was saved in UEFI. + * So we can share it from linux, and data size is variable. + * Get real size and read it from UEFI. + */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, tas_dev->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, + tas_dev->cali_data.data); + dev_dbg(tas_dev->dev, "cali get %lx bytes result:%ld\n", + tas_dev->cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return 0; + } + + tmp_val = (unsigned int *)tas_dev->cali_data.data; + + crc = crc32(~0, tas_dev->cali_data.data, 84) ^ ~0; + + if (crc == tmp_val[21]) { + /* Date and time of calibration was done. */ + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_dev->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_dev, tmp_val); + } else { + dev_dbg(tas_dev->dev, "CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[21]); + tas_dev->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret, value_sdw; + + mutex_lock(&tas_dev->codec_lock); + + if (!map) { + dev_err(tas_dev->dev, "%s, wrong regmap.\n", __func__); + ret = -EINVAL; + goto out; + } + if (!fmw || !fmw->data) { + /* No firmware binary, devices will work in ROM mode. */ + dev_err(tas_dev->dev, + "Failed to read %s, no side-effect on driver running\n", + tas_dev->rca_binaryname); + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = le32_to_cpup((__le32 *)&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, "Size not matching, %d %u", + (int)fmw->size, img_sz); + ret = -EINVAL; + goto out; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int)*5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int)*5)); + } + if (ret != 0) { + dev_dbg(tas_dev->dev, "Load FW fail: %d.\n", ret); + goto out; + } + offset += sizeof(unsigned int)*5 + p->length; + } + /* Select left-right channel based on unique id. */ + value_sdw = 0x1a; + value_sdw += ((tas_dev->sdw_peripheral->id.unique_id & 1) << 4); + regmap_write(map, TASDEVICE_REG(0, 0, 0x0a), value_sdw); + + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_SPK("SPK", NULL), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + snd_soc_dai_dma_data_set(dai, direction, stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_data *stream; + int ret; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, stream->sdw_stream); + if (ret) { + dev_err(dai->dev, "Unable to configure port\n"); + return ret; + } + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, + stream->sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + if (!map) { + dev_err(tas_dev->dev, "%s, wrong regmap.\n", __func__); + return -EINVAL; + } + + if (mute == 0) {/* Unmute. */ + /* FU23 Unmute, 0x40400108. */ + ret = regmap_write(map, SDW_SDCA_CTL(1, 2, 1, 0), 0); + ret += regmap_write(map, TASDEVICE_REG(0, 0, 0x02), 0x0); + } else {/* Mute */ + /* FU23 mute */ + ret = regmap_write(map, SDW_SDCA_CTL(1, 2, 1, 0), 1); + ret += regmap_write(map, TASDEVICE_REG(0, 0, 0x02), 0x1a); + } + if (ret) { + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + } + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + struct regmap *map = tas_dev->regmap; + int ret; + + if (!map) { + dev_err(tas_dev->dev, "Failed to load regmap.\n"); + return; + } + ret = regmap_write(map, TAS2873_REG_SWRESET, 1); + if (ret) { + dev_err(tas_dev->dev, "Reset failed.\n"); + return; + } + usleep_range(1000, 1050); +} + +static int tasdevice_component_probe(struct snd_soc_component *component) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + tas_dev->component = component; + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + dev_dbg(tas_dev->dev, "%s was called.\n", __func__); + + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_component_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + } + + tas2783_reset(tas_dev); + /* tas2783-8[9,...,f].bin was copied into /lib/firmware/ */ + scnprintf(tas_dev->rca_binaryname, 64, "tas2783-%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + if (ret) { + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + + pm_runtime_enable(tas_dev->dev); + + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (tas_priv->hw_init) + return 0; + + /* Mark Slave initialization complete */ + tas_priv->hw_init = true; + + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); + + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + tas_dev->hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, + &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed devm_regmap_init: %d\n", ret); + goto out; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + + return ret; +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev) { + pm_runtime_disable(tas_dev->dev); + tasdevice_remove(tas_dev); + } + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..1fe56f05b9d9 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,100 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coff setting for single TAS2783 chips. + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 +\ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x0d, 0x3c) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +#define TAS2783_ID_MIN 0x08 // Unique id start +#define TAS2783_ID_MAX 0x0F // Unique id end + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_PDE22 0x0B +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_UDMPU21 0x10 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 + +#define TAS2783_MAX_CALIDATA_SIZE 252 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool hw_init; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

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