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This is in relation for the number of event rings that can be potentially used by other processors within the system. Acked-by: Thinh Nguyen Tested-by: Puma Hsu Tested-by: Daehwan Jung Signed-off-by: Wesley Cheng --- drivers/usb/dwc3/core.c | 12 ++++++++++++ drivers/usb/dwc3/core.h | 2 ++ drivers/usb/dwc3/host.c | 3 +++ 3 files changed, 17 insertions(+) diff --git a/drivers/usb/dwc3/core.c b/drivers/usb/dwc3/core.c index 66a08b527165..17ae5c13fe36 100644 --- a/drivers/usb/dwc3/core.c +++ b/drivers/usb/dwc3/core.c @@ -1699,6 +1699,7 @@ static void dwc3_get_properties(struct dwc3 *dwc) u8 tx_thr_num_pkt_prd = 0; u8 tx_max_burst_prd = 0; u8 tx_fifo_resize_max_num; + u16 num_hc_interrupters; /* default to highest possible threshold */ lpm_nyet_threshold = 0xf; @@ -1719,6 +1720,9 @@ static void dwc3_get_properties(struct dwc3 *dwc) */ tx_fifo_resize_max_num = 6; + /* default to a single XHCI interrupter */ + num_hc_interrupters = 1; + dwc->maximum_speed = usb_get_maximum_speed(dev); dwc->max_ssp_rate = usb_get_maximum_ssp_rate(dev); dwc->dr_mode = usb_get_dr_mode(dev); @@ -1765,6 +1769,12 @@ static void dwc3_get_properties(struct dwc3 *dwc) &tx_thr_num_pkt_prd); device_property_read_u8(dev, "snps,tx-max-burst-prd", &tx_max_burst_prd); + device_property_read_u16(dev, "num-hc-interrupters", + &num_hc_interrupters); + /* DWC3 core allowed to have a max of 8 interrupters */ + if (num_hc_interrupters > 8) + num_hc_interrupters = 8; + dwc->do_fifo_resize = device_property_read_bool(dev, "tx-fifo-resize"); if (dwc->do_fifo_resize) @@ -1851,6 +1861,8 @@ static void dwc3_get_properties(struct dwc3 *dwc) dwc->tx_max_burst_prd = tx_max_burst_prd; dwc->tx_fifo_resize_max_num = tx_fifo_resize_max_num; + + dwc->num_hc_interrupters = num_hc_interrupters; } /* check whether the core supports IMOD */ diff --git a/drivers/usb/dwc3/core.h b/drivers/usb/dwc3/core.h index aaa39e663f60..fbe83914d9f9 100644 --- a/drivers/usb/dwc3/core.h +++ b/drivers/usb/dwc3/core.h @@ -1083,6 +1083,7 @@ struct dwc3_scratchpad_array { * @tx_max_burst_prd: max periodic ESS transmit burst size * @tx_fifo_resize_max_num: max number of fifos allocated during txfifo resize * @clear_stall_protocol: endpoint number that requires a delayed status phase + * @num_hc_interrupters: number of host controller interrupters * @hsphy_interface: "utmi" or "ulpi" * @connected: true when we're connected to a host, false otherwise * @softconnect: true when gadget connect is called, false when disconnect runs @@ -1330,6 +1331,7 @@ struct dwc3 { u8 tx_max_burst_prd; 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Fri, 04 Apr 2025 00:27:42 +0000 (GMT) Received: from nalasex01b.na.qualcomm.com (nalasex01b.na.qualcomm.com [10.47.209.197]) by NALASPPMTA01.qualcomm.com (8.18.1.2/8.18.1.2) with ESMTPS id 5340Rfbg006287 (version=TLSv1.2 cipher=ECDHE-RSA-AES256-GCM-SHA384 bits=256 verify=NOT); Fri, 4 Apr 2025 00:27:41 GMT Received: from hu-wcheng-lv.qualcomm.com (10.49.16.6) by nalasex01b.na.qualcomm.com (10.47.209.197) with Microsoft SMTP Server (version=TLS1_2, cipher=TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384) id 15.2.1544.9; Thu, 3 Apr 2025 17:27:41 -0700 From: Wesley Cheng To: , , , , , , , , , , , , , CC: , , , , , , , Wesley Cheng , Puma Hsu , Daehwan Jung Subject: [PATCH v37 08/31] ALSA: usb-audio: Export USB SND APIs for modules Date: Thu, 3 Apr 2025 17:27:05 -0700 Message-ID: <20250404002728.3590501-9-quic_wcheng@quicinc.com> X-Mailer: git-send-email 2.34.1 In-Reply-To: <20250404002728.3590501-1-quic_wcheng@quicinc.com> References: <20250404002728.3590501-1-quic_wcheng@quicinc.com> Precedence: bulk X-Mailing-List: linux-arm-msm@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 X-ClientProxiedBy: nalasex01c.na.qualcomm.com (10.47.97.35) To nalasex01b.na.qualcomm.com (10.47.209.197) X-QCInternal: smtphost X-Proofpoint-Virus-Version: vendor=nai engine=6200 definitions=5800 signatures=585085 X-Authority-Analysis: v=2.4 cv=CfwI5Krl c=1 sm=1 tr=0 ts=67ef277e cx=c_pps a=ouPCqIW2jiPt+lZRy3xVPw==:117 a=ouPCqIW2jiPt+lZRy3xVPw==:17 a=3H110R4YSZwA:10 a=XR8D0OoHHMoA:10 a=QyXUC8HyAAAA:8 a=1XWaLZrsAAAA:8 a=hD80L64hAAAA:8 a=COk6AnOGAAAA:8 a=4IsYxUhFtTOy7luWsxIA:9 a=TjNXssC_j7lpFel5tvFf:22 X-Proofpoint-ORIG-GUID: bpb2tB3Z3jayu263PL41KeFuMQHwrH-a X-Proofpoint-GUID: bpb2tB3Z3jayu263PL41KeFuMQHwrH-a X-Proofpoint-Virus-Version: vendor=baseguard engine=ICAP:2.0.293,Aquarius:18.0.1095,Hydra:6.0.680,FMLib:17.12.68.34 definitions=2025-04-03_11,2025-04-03_03,2024-11-22_01 X-Proofpoint-Spam-Details: rule=outbound_notspam policy=outbound score=0 priorityscore=1501 phishscore=0 impostorscore=0 lowpriorityscore=0 bulkscore=0 adultscore=0 mlxscore=0 mlxlogscore=999 suspectscore=0 clxscore=1015 spamscore=0 malwarescore=0 classifier=spam authscore=0 authtc=n/a authcc= route=outbound adjust=0 reason=mlx scancount=1 engine=8.19.0-2502280000 definitions=main-2504040001 Some vendor modules will utilize useful parsing and endpoint management APIs to start audio playback/capture. Reviewed-by: Pierre-Louis Bossart Tested-by: Puma Hsu Tested-by: Daehwan Jung Signed-off-by: Wesley Cheng --- sound/usb/card.c | 4 +++ sound/usb/endpoint.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 75 +++++++++++++++++++++++++++++++------------- sound/usb/pcm.h | 11 +++++++ 5 files changed, 71 insertions(+), 21 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9c411b82a218..a0087bde684c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1030,6 +1030,7 @@ int snd_usb_lock_shutdown(struct snd_usb_audio *chip) wake_up(&chip->shutdown_wait); return err; } +EXPORT_SYMBOL_GPL(snd_usb_lock_shutdown); /* autosuspend and unlock the shutdown */ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) @@ -1038,6 +1039,7 @@ void snd_usb_unlock_shutdown(struct snd_usb_audio *chip) if (atomic_dec_and_test(&chip->usage_count)) wake_up(&chip->shutdown_wait); } +EXPORT_SYMBOL_GPL(snd_usb_unlock_shutdown); int snd_usb_autoresume(struct snd_usb_audio *chip) { @@ -1060,6 +1062,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) } return 0; } +EXPORT_SYMBOL_GPL(snd_usb_autoresume); void snd_usb_autosuspend(struct snd_usb_audio *chip) { @@ -1073,6 +1076,7 @@ void snd_usb_autosuspend(struct snd_usb_audio *chip) for (i = 0; i < chip->num_interfaces; i++) usb_autopm_put_interface(chip->intf[i]); } +EXPORT_SYMBOL_GPL(snd_usb_autosuspend); static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) { diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a29f28eb7d0c..1fed039b10ed 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1524,6 +1524,7 @@ int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, mutex_unlock(&chip->mutex); return err; } +EXPORT_SYMBOL_GPL(snd_usb_endpoint_prepare); /* get the current rate set to the given clock by any endpoint */ int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 72b671fb2c84..497d2b27fb59 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -62,6 +62,7 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype } return NULL; } +EXPORT_SYMBOL_GPL(snd_usb_find_csint_desc); /* * Wrapper for usb_control_msg(). diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 08bf535ed163..18467da6fd9e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -148,6 +148,16 @@ find_format(struct list_head *fmt_list_head, snd_pcm_format_t format, return found; } +const struct audioformat * +snd_usb_find_format(struct list_head *fmt_list_head, snd_pcm_format_t format, + unsigned int rate, unsigned int channels, bool strict_match, + struct snd_usb_substream *subs) +{ + return find_format(fmt_list_head, format, rate, channels, strict_match, + subs); +} +EXPORT_SYMBOL_GPL(snd_usb_find_format); + static const struct audioformat * find_substream_format(struct snd_usb_substream *subs, const struct snd_pcm_hw_params *params) @@ -157,6 +167,14 @@ find_substream_format(struct snd_usb_substream *subs, true, subs); } +const struct audioformat * +snd_usb_find_substream_format(struct snd_usb_substream *subs, + const struct snd_pcm_hw_params *params) +{ + return find_substream_format(subs, params); +} +EXPORT_SYMBOL_GPL(snd_usb_find_substream_format); + bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs) { const struct audioformat *fp; @@ -461,20 +479,9 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -/* - * hw_params callback - * - * allocate a buffer and set the given audio format. - * - * so far we use a physically linear buffer although packetize transfer - * doesn't need a continuous area. - * if sg buffer is supported on the later version of alsa, we'll follow - * that. - */ -static int snd_usb_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) +int snd_usb_hw_params(struct snd_usb_substream *subs, + struct snd_pcm_hw_params *hw_params) { - struct snd_usb_substream *subs = substream->runtime->private_data; struct snd_usb_audio *chip = subs->stream->chip; const struct audioformat *fmt; const struct audioformat *sync_fmt; @@ -499,7 +506,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, if (fmt->implicit_fb) { sync_fmt = snd_usb_find_implicit_fb_sync_format(chip, fmt, hw_params, - !substream->stream, + !subs->direction, &sync_fixed_rate); if (!sync_fmt) { usb_audio_dbg(chip, @@ -579,15 +586,28 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, return ret; } +EXPORT_SYMBOL_GPL(snd_usb_hw_params); /* - * hw_free callback + * hw_params callback * - * reset the audio format and release the buffer + * allocate a buffer and set the given audio format. + * + * so far we use a physically linear buffer although packetize transfer + * doesn't need a continuous area. + * if sg buffer is supported on the later version of alsa, we'll follow + * that. */ -static int snd_usb_hw_free(struct snd_pcm_substream *substream) +static int snd_usb_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { struct snd_usb_substream *subs = substream->runtime->private_data; + + return snd_usb_hw_params(subs, hw_params); +} + +int snd_usb_hw_free(struct snd_usb_substream *subs) +{ struct snd_usb_audio *chip = subs->stream->chip; snd_media_stop_pipeline(subs); @@ -603,6 +623,19 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) return 0; } +EXPORT_SYMBOL_GPL(snd_usb_hw_free); + +/* + * hw_free callback + * + * reset the audio format and release the buffer + */ +static int snd_usb_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_usb_substream *subs = substream->runtime->private_data; + + return snd_usb_hw_free(subs); +} /* free-wheeling mode? (e.g. dmix) */ static int in_free_wheeling_mode(struct snd_pcm_runtime *runtime) @@ -1746,8 +1779,8 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream static const struct snd_pcm_ops snd_usb_playback_ops = { .open = snd_usb_pcm_open, .close = snd_usb_pcm_close, - .hw_params = snd_usb_hw_params, - .hw_free = snd_usb_hw_free, + .hw_params = snd_usb_pcm_hw_params, + .hw_free = snd_usb_pcm_hw_free, .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_substream_playback_trigger, .sync_stop = snd_usb_pcm_sync_stop, @@ -1758,8 +1791,8 @@ static const struct snd_pcm_ops snd_usb_playback_ops = { static const struct snd_pcm_ops snd_usb_capture_ops = { .open = snd_usb_pcm_open, .close = snd_usb_pcm_close, - .hw_params = snd_usb_hw_params, - .hw_free = snd_usb_hw_free, + .hw_params = snd_usb_pcm_hw_params, + .hw_free = snd_usb_pcm_hw_free, .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_substream_capture_trigger, .sync_stop = snd_usb_pcm_sync_stop, diff --git a/sound/usb/pcm.h b/sound/usb/pcm.h index 388fe2ba346d..c096021adb2b 100644 --- a/sound/usb/pcm.h +++ b/sound/usb/pcm.h @@ -15,4 +15,15 @@ void snd_usb_preallocate_buffer(struct snd_usb_substream *subs); int snd_usb_audioformat_set_sync_ep(struct snd_usb_audio *chip, struct audioformat *fmt); +const struct audioformat * +snd_usb_find_format(struct list_head *fmt_list_head, snd_pcm_format_t format, + unsigned int rate, unsigned int channels, bool strict_match, + struct snd_usb_substream *subs); +const struct audioformat * +snd_usb_find_substream_format(struct snd_usb_substream *subs, + const struct snd_pcm_hw_params *params); + +int snd_usb_hw_params(struct snd_usb_substream *subs, + struct snd_pcm_hw_params *hw_params); +int snd_usb_hw_free(struct snd_usb_substream *subs); #endif /* __USBAUDIO_PCM_H */ From patchwork Fri Apr 4 00:27:16 2025 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Wesley Cheng X-Patchwork-Id: 14037914 Received: from mx0a-0031df01.pphosted.com (mx0a-0031df01.pphosted.com [205.220.168.131]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by smtp.subspace.kernel.org (Postfix) with ESMTPS id 157D21F55FB; 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It provides examples on how to implement the drivers that will need to be introduced in order to enable USB audio offloading. Signed-off-by: Wesley Cheng --- Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 482 ++++++++++++++++++++++++++++++ 2 files changed, 483 insertions(+) create mode 100644 Documentation/sound/soc/usb.rst diff --git a/Documentation/sound/soc/index.rst b/Documentation/sound/soc/index.rst index e57df2dab2fd..8bed8f8f48da 100644 --- a/Documentation/sound/soc/index.rst +++ b/Documentation/sound/soc/index.rst @@ -18,3 +18,4 @@ The documentation is spilt into the following sections:- jack dpcm codec-to-codec + usb diff --git a/Documentation/sound/soc/usb.rst b/Documentation/sound/soc/usb.rst new file mode 100644 index 000000000000..94c12f9d9dd1 --- /dev/null +++ b/Documentation/sound/soc/usb.rst @@ -0,0 +1,482 @@ +================ +ASoC USB support +================ + +Overview +======== +In order to leverage the existing USB sound device support in ALSA, the +ASoC USB APIs are introduced to allow the subsystems to exchange +configuration information. + +One potential use case would be to support USB audio offloading, which is +an implementation that allows for an alternate power-optimized path in the audio +subsystem to handle the transfer of audio data over the USB bus. This would +let the main processor to stay in lower power modes for longer duration. The +following is an example design of how the ASoC and ALSA pieces can be connected +together to achieve this: + +:: + + USB | ASoC + | _________________________ + | | ASoC Platform card | + | |_________________________| + | | | + | ___V____ ____V____ + | |ASoC BE | |ASoC FE | + | |DAI LNK | |DAI LNK | + | |________| |_________| + | ^ ^ ^ + | | |________| + | ___V____ | + | |SoC-USB | | + ________ ________ | | | + |USB SND |<--->|USBSND |<------------>|________| | + |(card.c)| |offld |<---------- | + |________| |________|___ | | | + ^ ^ | | | ____________V_________ + | | | | | |IPC | + __ V_______________V_____ | | | |______________________| + |USB SND (endpoint.c) | | | | ^ + |_________________________| | | | | + ^ | | | ___________V___________ + | | | |->|audio DSP | + ___________V_____________ | | |_______________________| + |XHCI HCD |<- | + |_________________________| | + + +SoC USB driver +============== +Structures +---------- +``struct snd_soc_usb`` + + - ``list``: list head for SND SoC struct list + - ``component``: reference to ASoC component + - ``connection_status_cb``: callback to notify connection events + - ``update_offload_route_info``: callback to fetch selected USB sound card/PCM + device + - ``priv_data``: driver data + +The snd_soc_usb structure can be referenced using the ASoC platform card +device, or a USB device (udev->dev). This is created by the ASoC BE DAI +link, and the USB sound entity will be able to pass information to the +ASoC BE DAI link using this structure. + +``struct snd_soc_usb_device`` + + - ``card_idx``: sound card index associated with USB sound device + - ``chip_idx``: USB sound chip array index + - ``cpcm_idx``: capture pcm device indexes associated with the USB sound device + - ``ppcm_idx``: playback pcm device indexes associated with the USB sound device + - ``num_playback``: number of playback streams + - ``num_capture``: number of capture streams + - ``list``: list head for the USB sound device list + +The struct snd_soc_usb_device is created by the USB sound offload driver. +This will carry basic parameters/limitations that will be used to +determine the possible offloading paths for this USB audio device. + +Functions +--------- +.. code-block:: rst + + int snd_soc_usb_find_supported_format(int card_idx, + struct snd_pcm_hw_params *params, int direction) +.. + + - ``card_idx``: the index into the USB sound chip array. + - ``params``: Requested PCM parameters from the USB DPCM BE DAI link + - ``direction``: capture or playback + +**snd_soc_usb_find_supported_format()** ensures that the requested audio profile +being requested by the external DSP is supported by the USB device. + +Returns 0 on success, and -EOPNOTSUPP on failure. + +.. code-block:: rst + + int snd_soc_usb_connect(struct device *usbdev, struct snd_soc_usb_device *sdev) +.. + + - ``usbdev``: the usb device that was discovered + - ``sdev``: capabilities of the device + +**snd_soc_usb_connect()** notifies the ASoC USB DCPM BE DAI link of a USB +audio device detection. This can be utilized in the BE DAI +driver to keep track of available USB audio devices. This is intended +to be called by the USB offload driver residing in USB SND. + +Returns 0 on success, negative error code on failure. + +.. code-block:: rst + + int snd_soc_usb_disconnect(struct device *usbdev, struct snd_soc_usb_device *sdev) +.. + + - ``usbdev``: the usb device that was removed + - ``sdev``: capabilities to free + +**snd_soc_usb_disconnect()** notifies the ASoC USB DCPM BE DAI link of a USB +audio device removal. This is intended to be called by the USB offload +driver that resides in USB SND. + +.. code-block:: rst + + void *snd_soc_usb_find_priv_data(struct device *usbdev) +.. + + - ``usbdev``: the usb device to reference to find private data + +**snd_soc_usb_find_priv_data()** fetches the private data saved to the SoC USB +device. + +Returns pointer to priv_data on success, NULL on failure. + +.. code-block:: rst + + int snd_soc_usb_setup_offload_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack) +.. + + - ``component``: ASoC component to add the jack + - ``jack``: jack component to populate + +**snd_soc_usb_setup_offload_jack()** is a helper to add a sound jack control to +the platform sound card. This will allow for consistent naming to be used on +designs that support USB audio offloading. Additionally, this will enable the +jack to notify of changes. + +Returns 0 on success, negative otherwise. + +.. code-block:: rst + + int snd_soc_usb_update_offload_route(struct device *dev, int card, int pcm, + int direction, enum snd_soc_usb_kctl path, + long *route) +.. + + - ``dev``: USB device to look up offload path mapping + - ``card``: USB sound card index + - ``pcm``: USB sound PCM device index + - ``direction``: direction to fetch offload routing information + - ``path``: kcontrol selector - pcm device or card index + - ``route``: mapping of sound card and pcm indexes for the offload path. This is + an array of two integers that will carry the card and pcm device indexes + in that specific order. This can be used as the array for the kcontrol + output. + +**snd_soc_usb_update_offload_route()** calls a registered callback to the USB BE DAI +link to fetch the information about the mapped ASoC devices for executing USB audio +offload for the device. ``route`` may be a pointer to a kcontrol value output array, +which carries values when the kcontrol is read. + +Returns 0 on success, negative otherwise. + +.. code-block:: rst + + struct snd_soc_usb *snd_soc_usb_allocate_port(struct snd_soc_component *component, + void *data); +.. + + - ``component``: DPCM BE DAI link component + - ``data``: private data + +**snd_soc_usb_allocate_port()** allocates a SoC USB device and populates standard +parameters that is used for further operations. + +Returns a pointer to struct soc_usb on success, negative on error. + +.. code-block:: rst + + void snd_soc_usb_free_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to free + +**snd_soc_usb_free_port()** frees a SoC USB device. + +.. code-block:: rst + + void snd_soc_usb_add_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to add + +**snd_soc_usb_add_port()** add an allocated SoC USB device to the SOC USB framework. +Once added, this device can be referenced by further operations. + +.. code-block:: rst + + void snd_soc_usb_remove_port(struct snd_soc_usb *usb); +.. + + - ``usb``: SoC USB device to remove + +**snd_soc_usb_remove_port()** removes a SoC USB device from the SoC USB framework. +After removing a device, any SOC USB operations would not be able to reference the +device removed. + +How to Register to SoC USB +-------------------------- +The ASoC DPCM USB BE DAI link is the entity responsible for allocating and +registering the SoC USB device on the component bind. Likewise, it will +also be responsible for freeing the allocated resources. An example can +be shown below: + +.. code-block:: rst + + static int q6usb_component_probe(struct snd_soc_component *component) + { + ... + data->usb = snd_soc_usb_allocate_port(component, 1, &data->priv); + if (!data->usb) + return -ENOMEM; + + usb->connection_status_cb = q6usb_alsa_connection_cb; + + ret = snd_soc_usb_add_port(usb); + if (ret < 0) { + dev_err(component->dev, "failed to add usb port\n"); + goto free_usb; + } + ... + } + + static void q6usb_component_remove(struct snd_soc_component *component) + { + ... + snd_soc_usb_remove_port(data->usb); + snd_soc_usb_free_port(data->usb); + } + + static const struct snd_soc_component_driver q6usb_dai_component = { + .probe = q6usb_component_probe, + .remove = q6usb_component_remove, + .name = "q6usb-dai-component", + ... + }; +.. + +BE DAI links can pass along vendor specific information as part of the +call to allocate the SoC USB device. This will allow any BE DAI link +parameters or settings to be accessed by the USB offload driver that +resides in USB SND. + +USB Audio Device Connection Flow +-------------------------------- +USB devices can be hotplugged into the USB ports at any point in time. +The BE DAI link should be aware of the current state of the physical USB +port, i.e. if there are any USB devices with audio interface(s) connected. +connection_status_cb() can be used to notify the BE DAI link of any change. + +This is called whenever there is a USB SND interface bind or remove event, +using snd_soc_usb_connect() or snd_soc_usb_disconnect(): + +.. code-block:: rst + + static void qc_usb_audio_offload_probe(struct snd_usb_audio *chip) + { + ... + snd_soc_usb_connect(usb_get_usb_backend(udev), sdev); + ... + } + + static void qc_usb_audio_offload_disconnect(struct snd_usb_audio *chip) + { + ... + snd_soc_usb_disconnect(usb_get_usb_backend(chip->dev), dev->sdev); + ... + } +.. + +In order to account for conditions where driver or device existence is +not guaranteed, USB SND exposes snd_usb_rediscover_devices() to resend the +connect events for any identified USB audio interfaces. Consider the +the following situation: + + **usb_audio_probe()** + | --> USB audio streams allocated and saved to usb_chip[] + | --> Propagate connect event to USB offload driver in USB SND + | --> **snd_soc_usb_connect()** exits as USB BE DAI link is not ready + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> The USB audio device connect event is missed + +To ensure connection events are not missed, **snd_usb_rediscover_devices()** +is executed when the SoC USB device is registered. Now, when the BE DAI +link component probe occurs, the following highlights the sequence: + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> SoC USB device added, and **snd_usb_rediscover_devices()** runs + + **snd_usb_rediscover_devices()** + | --> Traverses through usb_chip[] and for non-NULL entries issue + | **connection_status_cb()** + +In the case where the USB offload driver is unbound, while USB SND is ready, +the **snd_usb_rediscover_devices()** is called during module init. This allows +for the offloading path to also be enabled with the following flow: + + **usb_audio_probe()** + | --> USB audio streams allocated and saved to usb_chip[] + | --> Propagate connect event to USB offload driver in USB SND + | --> USB offload driver **NOT** ready! + + BE DAI link component probe + | --> DAI link is probed and SoC USB port is allocated + | --> No USB connect event due to missing USB offload driver + + USB offload driver probe + | --> **qc_usb_audio_offload_init()** + | --> Calls **snd_usb_rediscover_devices()** to notify of devices + +USB Offload Related Kcontrols +============================= +Details +------- +A set of kcontrols can be utilized by applications to help select the proper sound +devices to enable USB audio offloading. SoC USB exposes the get_offload_dev() +callback that designs can use to ensure that the proper indices are returned to the +application. + +Implementation +-------------- + +**Example:** + + **Sound Cards**: + + :: + + 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D + SM8250-MTP-WCD9380-WSA8810-VA-DMIC + 1 [Seri ]: USB-Audio - Plantronics Blackwire 3225 Seri + Plantronics Plantronics Blackwire + 3225 Seri at usb-xhci-hcd.1.auto-1.1, + full sp + 2 [C320M ]: USB-Audio - Plantronics C320-M + Plantronics Plantronics C320-M at usb-xhci-hcd.1.auto-1.2, full speed + + **PCM Devices**: + + :: + + card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 0: MultiMedia1 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 0: SM8250MTPWCD938 [SM8250-MTP-WCD9380-WSA8810-VA-D], device 1: MultiMedia2 (*) [] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 1: Seri [Plantronics Blackwire 3225 Seri], device 0: USB Audio [USB Audio] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + card 2: C320M [Plantronics C320-M], device 0: USB Audio [USB Audio] + Subdevices: 1/1 + Subdevice #0: subdevice #0 + + **USB Sound Card** - card#1: + + :: + + USB Offload Playback Card Route PCM#0 -1 (range -1->32) + USB Offload Playback PCM Route PCM#0 -1 (range -1->255) + + **USB Sound Card** - card#2: + + :: + + USB Offload Playback Card Route PCM#0 0 (range -1->32) + USB Offload Playback PCM Route PCM#0 1 (range -1->255) + +The above example shows a scenario where the system has one ASoC platform card +(card#0) and two USB sound devices connected (card#1 and card#2). When reading +the available kcontrols for each USB audio device, the following kcontrols lists +the mapped offload card and pcm device indexes for the specific USB device: + + ``USB Offload Playback Card Route PCM#*`` + + ``USB Offload Playback PCM Route PCM#*`` + +The kcontrol is indexed, because a USB audio device could potentially have +several PCM devices. The above kcontrols are defined as: + + - ``USB Offload Playback Card Route PCM#`` **(R)**: Returns the ASoC platform sound + card index for a mapped offload path. The output **"0"** (card index) signifies + that there is an available offload path for the USB SND device through card#0. + If **"-1"** is seen, then no offload path is available for the USB SND device. + This kcontrol exists for each USB audio device that exists in the system, and + its expected to derive the current status of offload based on the output value + for the kcontrol along with the PCM route kcontrol. + + - ``USB Offload Playback PCM Route PCM#`` **(R)**: Returns the ASoC platform sound + PCM device index for a mapped offload path. The output **"1"** (PCM device index) + signifies that there is an available offload path for the USB SND device through + PCM device#0. If **"-1"** is seen, then no offload path is available for the USB\ + SND device. This kcontrol exists for each USB audio device that exists in the + system, and its expected to derive the current status of offload based on the + output value for this kcontrol, in addition to the card route kcontrol. + +USB Offload Playback Route Kcontrol +----------------------------------- +In order to allow for vendor specific implementations on audio offloading device +selection, the SoC USB layer exposes the following: + +.. code-block:: rst + + int (*update_offload_route_info)(struct snd_soc_component *component, + int card, int pcm, int direction, + enum snd_soc_usb_kctl path, + long *route) +.. + +These are specific for the **USB Offload Playback Card Route PCM#** and **USB +Offload PCM Route PCM#** kcontrols. + +When users issue get calls to the kcontrol, the registered SoC USB callbacks will +execute the registered function calls to the DPCM BE DAI link. + +**Callback Registration:** + +.. code-block:: rst + + static int q6usb_component_probe(struct snd_soc_component *component) + { + ... + usb = snd_soc_usb_allocate_port(component, 1, &data->priv); + if (IS_ERR(usb)) + return -ENOMEM; + + usb->connection_status_cb = q6usb_alsa_connection_cb; + usb->update_offload_route_info = q6usb_get_offload_dev; + + ret = snd_soc_usb_add_port(usb); +.. + +Existing USB Sound Kcontrol +--------------------------- +With the introduction of USB offload support, the above USB offload kcontrol +will be added to the pre existing list of kcontrols identified by the USB sound +framework. These kcontrols are still the main controls that are used to +modify characteristics pertaining to the USB audio device. + + :: + + Number of controls: 9 + ctl type num name value + 0 INT 2 Capture Channel Map 0, 0 (range 0->36) + 1 INT 2 Playback Channel Map 0, 0 (range 0->36) + 2 BOOL 1 Headset Capture Switch On + 3 INT 1 Headset Capture Volume 10 (range 0->13) + 4 BOOL 1 Sidetone Playback Switch On + 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) + 6 BOOL 1 Headset Playback Switch On + 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) + 8 INT 1 USB Offload Playback Card Route PCM#0 0 (range -1->32) + 9 INT 1 USB Offload Playback PCM Route PCM#0 1 (range -1->255) + +Since USB audio device controls are handled over the USB control endpoint, use the +existing mechanisms present in the USB mixer to set parameters, such as volume. 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Fri, 4 Apr 2025 00:27:45 GMT Received: from hu-wcheng-lv.qualcomm.com (10.49.16.6) by nalasex01b.na.qualcomm.com (10.47.209.197) with Microsoft SMTP Server (version=TLS1_2, cipher=TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384) id 15.2.1544.9; Thu, 3 Apr 2025 17:27:45 -0700 From: Wesley Cheng To: , , , , , , , , , , , , , CC: , , , , , , , Wesley Cheng Subject: [PATCH v37 22/31] ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp Date: Thu, 3 Apr 2025 17:27:19 -0700 Message-ID: <20250404002728.3590501-23-quic_wcheng@quicinc.com> X-Mailer: git-send-email 2.34.1 In-Reply-To: <20250404002728.3590501-1-quic_wcheng@quicinc.com> References: <20250404002728.3590501-1-quic_wcheng@quicinc.com> Precedence: bulk X-Mailing-List: linux-arm-msm@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 X-ClientProxiedBy: nalasex01c.na.qualcomm.com (10.47.97.35) To nalasex01b.na.qualcomm.com (10.47.209.197) X-QCInternal: smtphost X-Proofpoint-Virus-Version: vendor=nai engine=6200 definitions=5800 signatures=585085 X-Proofpoint-ORIG-GUID: JtsUeQtnRDqLsNjzgrxBHo_Lz98sAHe6 X-Proofpoint-GUID: JtsUeQtnRDqLsNjzgrxBHo_Lz98sAHe6 X-Authority-Analysis: v=2.4 cv=Cvu/cm4D c=1 sm=1 tr=0 ts=67ef2782 cx=c_pps a=ouPCqIW2jiPt+lZRy3xVPw==:117 a=ouPCqIW2jiPt+lZRy3xVPw==:17 a=3H110R4YSZwA:10 a=XR8D0OoHHMoA:10 a=COk6AnOGAAAA:8 a=9B8dPbXtggJm2meAtL4A:9 a=TjNXssC_j7lpFel5tvFf:22 X-Proofpoint-Virus-Version: vendor=baseguard engine=ICAP:2.0.293,Aquarius:18.0.1095,Hydra:6.0.680,FMLib:17.12.68.34 definitions=2025-04-03_11,2025-04-03_03,2024-11-22_01 X-Proofpoint-Spam-Details: rule=outbound_notspam policy=outbound score=0 phishscore=0 adultscore=0 bulkscore=0 suspectscore=0 mlxlogscore=999 malwarescore=0 spamscore=0 impostorscore=0 lowpriorityscore=0 priorityscore=1501 mlxscore=0 clxscore=1015 classifier=spam authscore=0 authtc=n/a authcc= route=outbound adjust=0 reason=mlx scancount=1 engine=8.19.0-2502280000 definitions=main-2504040001 The QC ADSP is able to support USB playback endpoints, so that the main application processor can be placed into lower CPU power modes. This adds the required AFE port configurations and port start command to start an audio session. Specifically, the QC ADSP can support all potential endpoints that are exposed by the audio data interface. This includes isochronous data endpoints, in either synchronous mode or asynchronous mode. In the latter case both implicit or explicit feedback endpoints are supported. The size of audio samples sent per USB frame (microframe) will be adjusted based on information received on the feedback endpoint. Some pre-requisites are needed before issuing the AFE port start command, such as setting the USB AFE dev_token. This carries information about the available USB SND cards and PCM devices that have been discovered on the USB bus. The dev_token field is used by the audio DSP to notify the USB offload driver of which card and PCM index to enable playback on. Signed-off-by: Wesley Cheng --- sound/soc/qcom/qdsp6/q6afe-dai.c | 60 +++++++ sound/soc/qcom/qdsp6/q6afe.c | 192 ++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6afe.h | 36 ++++- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 10 +- 6 files changed, 319 insertions(+), 3 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 7d9628cda875..0f47aadaabe1 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -92,6 +92,39 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } +static int q6afe_usb_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); + int channels = params_channels(params); + int rate = params_rate(params); + struct q6afe_usb_cfg *usb = &dai_data->port_config[dai->id].usb_audio; + + usb->sample_rate = rate; + usb->num_channels = channels; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + usb->bit_width = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_3LE: + usb->bit_width = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + usb->bit_width = 32; + break; + default: + dev_err(dai->dev, "%s: invalid format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + return 0; +} + static int q6i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -394,6 +427,10 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream, q6afe_cdc_dma_port_prepare(dai_data->port[dai->id], &dai_data->port_config[dai->id].dma_cfg); break; + case USB_RX: + q6afe_usb_port_prepare(dai_data->port[dai->id], + &dai_data->port_config[dai->id].usb_audio); + break; default: return -EINVAL; } @@ -622,6 +659,9 @@ static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"TX_CODEC_DMA_TX_5", NULL, "TX_CODEC_DMA_TX_5 Capture"}, {"RX_CODEC_DMA_RX_6 Playback", NULL, "RX_CODEC_DMA_RX_6"}, {"RX_CODEC_DMA_RX_7 Playback", NULL, "RX_CODEC_DMA_RX_7"}, + + /* USB playback AFE port receives data for playback, hence use the RX port */ + {"USB Playback", NULL, "USB_RX"}, }; static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai) @@ -649,6 +689,23 @@ static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai) return 0; } +static const struct snd_soc_dai_ops q6afe_usb_ops = { + .probe = msm_dai_q6_dai_probe, + .prepare = q6afe_dai_prepare, + .hw_params = q6afe_usb_hw_params, + /* + * Shutdown callback required to stop the USB AFE port, which is enabled + * by the prepare() stage. This stops the audio traffic on the USB AFE + * port on the Q6DSP. + */ + .shutdown = q6afe_dai_shutdown, + /* + * Startup callback not needed, as AFE port start command passes the PCM + * parameters within the AFE command, which is provided by the PCM core + * during the prepare() stage. + */ +}; + static const struct snd_soc_dai_ops q6hdmi_ops = { .probe = msm_dai_q6_dai_probe, .remove = msm_dai_q6_dai_remove, @@ -947,6 +1004,8 @@ static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("RX_CODEC_DMA_RX_7", "NULL", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_IN("USB_RX", NULL, 0, SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_component_driver q6afe_dai_component = { @@ -1061,6 +1120,7 @@ static int q6afe_dai_dev_probe(struct platform_device *pdev) cfg.q6i2s_ops = &q6i2s_ops; cfg.q6tdm_ops = &q6tdm_ops; cfg.q6dma_ops = &q6dma_ops; + cfg.q6usb_ops = &q6afe_usb_ops; dais = q6dsp_audio_ports_set_config(dev, &cfg, &num_dais); return devm_snd_soc_register_component(dev, &q6afe_dai_component, dais, num_dais); diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index ef7557be5d66..73ce204e5e14 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -35,6 +35,8 @@ #define AFE_MODULE_TDM 0x0001028A #define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235 +#define AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS 0x000102A5 +#define AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT 0x000102AA #define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238 #define AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG 0x00010239 @@ -44,6 +46,7 @@ #define AFE_PARAM_ID_TDM_CONFIG 0x0001029D #define AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG 0x00010297 #define AFE_PARAM_ID_CODEC_DMA_CONFIG 0x000102B8 +#define AFE_PARAM_ID_USB_AUDIO_CONFIG 0x000102A4 #define AFE_CMD_REMOTE_LPASS_CORE_HW_VOTE_REQUEST 0x000100f4 #define AFE_CMD_RSP_REMOTE_LPASS_CORE_HW_VOTE_REQUEST 0x000100f5 #define AFE_CMD_REMOTE_LPASS_CORE_HW_DEVOTE_REQUEST 0x000100f6 @@ -72,12 +75,16 @@ #define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1 #define AFE_LINEAR_PCM_DATA 0x0 +#define AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG 0x1 /* Port IDs */ #define AFE_API_VERSION_HDMI_CONFIG 0x1 #define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E #define AFE_PORT_ID_HDMI_OVER_DP_RX 0x6020 +/* USB AFE port */ +#define AFE_PORT_ID_USB_RX 0x7000 + #define AFE_API_VERSION_SLIMBUS_CONFIG 0x1 /* Clock set API version */ #define AFE_API_VERSION_CLOCK_SET 1 @@ -513,12 +520,96 @@ struct afe_param_id_cdc_dma_cfg { u16 active_channels_mask; } __packed; +struct afe_param_id_usb_cfg { +/* Minor version used for tracking USB audio device configuration. + * Supported values: AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + */ + u32 cfg_minor_version; +/* Sampling rate of the port. + * Supported values: + * - AFE_PORT_SAMPLE_RATE_8K + * - AFE_PORT_SAMPLE_RATE_11025 + * - AFE_PORT_SAMPLE_RATE_12K + * - AFE_PORT_SAMPLE_RATE_16K + * - AFE_PORT_SAMPLE_RATE_22050 + * - AFE_PORT_SAMPLE_RATE_24K + * - AFE_PORT_SAMPLE_RATE_32K + * - AFE_PORT_SAMPLE_RATE_44P1K + * - AFE_PORT_SAMPLE_RATE_48K + * - AFE_PORT_SAMPLE_RATE_96K + * - AFE_PORT_SAMPLE_RATE_192K + */ + u32 sample_rate; +/* Bit width of the sample. + * Supported values: 16, 24 + */ + u16 bit_width; +/* Number of channels. + * Supported values: 1 and 2 + */ + u16 num_channels; +/* Data format supported by the USB. The supported value is + * 0 (#AFE_USB_AUDIO_DATA_FORMAT_LINEAR_PCM). + */ + u16 data_format; +/* this field must be 0 */ + u16 reserved; +/* device token of actual end USB audio device */ + u32 dev_token; +/* endianness of this interface */ + u32 endian; +/* service interval */ + u32 service_interval; +} __packed; + +/** + * struct afe_param_id_usb_audio_dev_params + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @dev_token: device token of actual end USB audio device + **/ +struct afe_param_id_usb_audio_dev_params { + u32 cfg_minor_version; + u32 dev_token; +} __packed; + +/** + * struct afe_param_id_usb_audio_dev_lpcm_fmt + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @endian: endianness of this interface + **/ +struct afe_param_id_usb_audio_dev_lpcm_fmt { + u32 cfg_minor_version; + u32 endian; +} __packed; + +#define AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL 0x000102B7 + +/** + * struct afe_param_id_usb_audio_svc_interval + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @svc_interval: service interval + **/ +struct afe_param_id_usb_audio_svc_interval { + u32 cfg_minor_version; + u32 svc_interval; +} __packed; + union afe_port_config { struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch; struct afe_param_id_slimbus_cfg slim_cfg; struct afe_param_id_i2s_cfg i2s_cfg; struct afe_param_id_tdm_cfg tdm_cfg; struct afe_param_id_cdc_dma_cfg dma_cfg; + struct afe_param_id_usb_cfg usb_cfg; } __packed; @@ -833,6 +924,7 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { RX_CODEC_DMA_RX_6, 1, 1}, [RX_CODEC_DMA_RX_7] = { AFE_PORT_ID_RX_CODEC_DMA_RX_7, RX_CODEC_DMA_RX_7, 1, 1}, + [USB_RX] = { AFE_PORT_ID_USB_RX, USB_RX, 1, 1}, }; static void q6afe_port_free(struct kref *ref) @@ -1290,6 +1382,101 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port, } EXPORT_SYMBOL_GPL(q6afe_tdm_port_prepare); +/** + * afe_port_send_usb_dev_param() - Send USB dev token + * + * @port: Instance of afe port + * @cardidx: USB SND card index to reference + * @pcmidx: USB SND PCM device index to reference + * + * The USB dev token carries information about which USB SND card instance and + * PCM device to execute the offload on. This information is carried through + * to the stream enable QMI request, which is handled by the offload class + * driver. The information is parsed to determine which USB device to query + * the required resources for. + */ +int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx) +{ + struct afe_param_id_usb_audio_dev_params usb_dev; + int ret; + + memset(&usb_dev, 0, sizeof(usb_dev)); + + usb_dev.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; + usb_dev.dev_token = (cardidx << 16) | (pcmidx << 8); + ret = q6afe_port_set_param_v2(port, &usb_dev, + AFE_PARAM_ID_USB_AUDIO_DEV_PARAMS, + AFE_MODULE_AUDIO_DEV_INTERFACE, + sizeof(usb_dev)); + if (ret) + dev_err(port->afe->dev, "%s: AFE device param cmd failed %d\n", + __func__, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(afe_port_send_usb_dev_param); + +static int afe_port_send_usb_params(struct q6afe_port *port, struct q6afe_usb_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + struct afe_param_id_usb_audio_dev_lpcm_fmt lpcm_fmt; + struct afe_param_id_usb_audio_svc_interval svc_int; + int ret; + + if (!pcfg) { + dev_err(port->afe->dev, "%s: Error, no configuration data\n", __func__); + ret = -EINVAL; + goto exit; + } + + memset(&lpcm_fmt, 0, sizeof(lpcm_fmt)); + memset(&svc_int, 0, sizeof(svc_int)); + + lpcm_fmt.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; + lpcm_fmt.endian = pcfg->usb_cfg.endian; + ret = q6afe_port_set_param_v2(port, &lpcm_fmt, + AFE_PARAM_ID_USB_AUDIO_DEV_LPCM_FMT, + AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(lpcm_fmt)); + if (ret) { + dev_err(port->afe->dev, "%s: AFE device param cmd LPCM_FMT failed %d\n", + __func__, ret); + goto exit; + } + + svc_int.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; + svc_int.svc_interval = pcfg->usb_cfg.service_interval; + ret = q6afe_port_set_param_v2(port, &svc_int, + AFE_PARAM_ID_USB_AUDIO_SVC_INTERVAL, + AFE_MODULE_AUDIO_DEV_INTERFACE, sizeof(svc_int)); + if (ret) + dev_err(port->afe->dev, "%s: AFE device param cmd svc_interval failed %d\n", + __func__, ret); + +exit: + return ret; +} + +/** + * q6afe_usb_port_prepare() - Prepare usb afe port. + * + * @port: Instance of afe port + * @cfg: USB configuration for the afe port + * + */ +void q6afe_usb_port_prepare(struct q6afe_port *port, + struct q6afe_usb_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + + pcfg->usb_cfg.cfg_minor_version = AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG; + pcfg->usb_cfg.sample_rate = cfg->sample_rate; + pcfg->usb_cfg.num_channels = cfg->num_channels; + pcfg->usb_cfg.bit_width = cfg->bit_width; + + afe_port_send_usb_params(port, cfg); +} +EXPORT_SYMBOL_GPL(q6afe_usb_port_prepare); + /** * q6afe_hdmi_port_prepare() - Prepare hdmi afe port. * @@ -1612,7 +1799,10 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) break; case AFE_PORT_ID_WSA_CODEC_DMA_RX_0 ... AFE_PORT_ID_RX_CODEC_DMA_RX_7: cfg_type = AFE_PARAM_ID_CODEC_DMA_CONFIG; - break; + break; + case AFE_PORT_ID_USB_RX: + cfg_type = AFE_PARAM_ID_USB_AUDIO_CONFIG; + break; default: dev_err(dev, "Invalid port id 0x%x\n", port_id); return ERR_PTR(-EINVAL); diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 65d0676075e1..a29abe4ce436 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -3,7 +3,7 @@ #ifndef __Q6AFE_H__ #define __Q6AFE_H__ -#define AFE_PORT_MAX 129 +#define AFE_PORT_MAX 137 #define MSM_AFE_PORT_TYPE_RX 0 #define MSM_AFE_PORT_TYPE_TX 1 @@ -203,6 +203,36 @@ struct q6afe_cdc_dma_cfg { u16 active_channels_mask; }; +/** + * struct q6afe_usb_cfg + * @cfg_minor_version: Minor version used for tracking USB audio device + * configuration. + * Supported values: + * AFE_API_MINOR_VERSION_USB_AUDIO_CONFIG + * @sample_rate: Sampling rate of the port + * Supported values: + * AFE_PORT_SAMPLE_RATE_8K + * AFE_PORT_SAMPLE_RATE_11025 + * AFE_PORT_SAMPLE_RATE_12K + * AFE_PORT_SAMPLE_RATE_16K + * AFE_PORT_SAMPLE_RATE_22050 + * AFE_PORT_SAMPLE_RATE_24K + * AFE_PORT_SAMPLE_RATE_32K + * AFE_PORT_SAMPLE_RATE_44P1K + * AFE_PORT_SAMPLE_RATE_48K + * AFE_PORT_SAMPLE_RATE_96K + * AFE_PORT_SAMPLE_RATE_192K + * @bit_width: Bit width of the sample. + * Supported values: 16, 24 + * @num_channels: Number of channels + * Supported values: 1, 2 + **/ +struct q6afe_usb_cfg { + u32 cfg_minor_version; + u32 sample_rate; + u16 bit_width; + u16 num_channels; +}; struct q6afe_port_config { struct q6afe_hdmi_cfg hdmi; @@ -210,6 +240,7 @@ struct q6afe_port_config { struct q6afe_i2s_cfg i2s_cfg; struct q6afe_tdm_cfg tdm; struct q6afe_cdc_dma_cfg dma_cfg; + struct q6afe_usb_cfg usb_audio; }; struct q6afe_port; @@ -219,6 +250,8 @@ int q6afe_port_start(struct q6afe_port *port); int q6afe_port_stop(struct q6afe_port *port); void q6afe_port_put(struct q6afe_port *port); int q6afe_get_port_id(int index); +void q6afe_usb_port_prepare(struct q6afe_port *port, + struct q6afe_usb_cfg *cfg); void q6afe_hdmi_port_prepare(struct q6afe_port *port, struct q6afe_hdmi_cfg *cfg); void q6afe_slim_port_prepare(struct q6afe_port *port, @@ -228,6 +261,7 @@ void q6afe_tdm_port_prepare(struct q6afe_port *port, struct q6afe_tdm_cfg *cfg); void q6afe_cdc_dma_port_prepare(struct q6afe_port *port, struct q6afe_cdc_dma_cfg *cfg); +int afe_port_send_usb_dev_param(struct q6afe_port *port, int cardidx, int pcmidx); int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id, int clk_src, int clk_root, unsigned int freq, int dir); diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c index 4919001de08b..4eed54b071a5 100644 --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c @@ -97,6 +97,26 @@ } static struct snd_soc_dai_driver q6dsp_audio_fe_dais[] = { + { + .playback = { + .stream_name = "USB Playback", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE, + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 192000, + }, + .id = USB_RX, + .name = "USB_RX", + }, { .playback = { .stream_name = "HDMI Playback", @@ -624,6 +644,9 @@ struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev, case WSA_CODEC_DMA_RX_0 ... RX_CODEC_DMA_RX_7: q6dsp_audio_fe_dais[i].ops = cfg->q6dma_ops; break; + case USB_RX: + q6dsp_audio_fe_dais[i].ops = cfg->q6usb_ops; + break; default: break; } diff --git a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h index 7f052c8a1257..d8dde6dd0aca 100644 --- a/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h +++ b/sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h @@ -11,6 +11,7 @@ struct q6dsp_audio_port_dai_driver_config { const struct snd_soc_dai_ops *q6i2s_ops; const struct snd_soc_dai_ops *q6tdm_ops; const struct snd_soc_dai_ops *q6dma_ops; + const struct snd_soc_dai_ops *q6usb_ops; }; struct snd_soc_dai_driver *q6dsp_audio_ports_set_config(struct device *dev, diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 90228699ba7d..0def036ed3c9 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -435,6 +435,7 @@ static struct session_data *get_session_from_id(struct msm_routing_data *data, return NULL; } + /** * q6routing_stream_close() - Deregister a stream * @@ -515,6 +516,9 @@ static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, return 1; } +static const struct snd_kcontrol_new usb_rx_mixer_controls[] = { + Q6ROUTING_RX_MIXERS(USB_RX) }; + static const struct snd_kcontrol_new hdmi_mixer_controls[] = { Q6ROUTING_RX_MIXERS(HDMI_RX) }; @@ -933,6 +937,9 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("RX_CODEC_DMA_RX_7 Audio Mixer", SND_SOC_NOPM, 0, 0, rx_codec_dma_rx_7_mixer_controls, ARRAY_SIZE(rx_codec_dma_rx_7_mixer_controls)), + SND_SOC_DAPM_MIXER("USB Mixer", SND_SOC_NOPM, 0, 0, + usb_rx_mixer_controls, + ARRAY_SIZE(usb_rx_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0, mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0, @@ -949,7 +956,6 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { mmul7_mixer_controls, ARRAY_SIZE(mmul7_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0, mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)), - }; 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Fri, 04 Apr 2025 00:27:47 +0000 (GMT) Received: from nalasex01b.na.qualcomm.com (nalasex01b.na.qualcomm.com [10.47.209.197]) by NALASPPMTA04.qualcomm.com (8.18.1.2/8.18.1.2) with ESMTPS id 5340RkFU012124 (version=TLSv1.2 cipher=ECDHE-RSA-AES256-GCM-SHA384 bits=256 verify=NOT); Fri, 4 Apr 2025 00:27:46 GMT Received: from hu-wcheng-lv.qualcomm.com (10.49.16.6) by nalasex01b.na.qualcomm.com (10.47.209.197) with Microsoft SMTP Server (version=TLS1_2, cipher=TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384) id 15.2.1544.9; Thu, 3 Apr 2025 17:27:45 -0700 From: Wesley Cheng To: , , , , , , , , , , , , , CC: , , , , , , , Wesley Cheng Subject: [PATCH v37 24/31] ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 Date: Thu, 3 Apr 2025 17:27:21 -0700 Message-ID: <20250404002728.3590501-25-quic_wcheng@quicinc.com> X-Mailer: git-send-email 2.34.1 In-Reply-To: <20250404002728.3590501-1-quic_wcheng@quicinc.com> References: <20250404002728.3590501-1-quic_wcheng@quicinc.com> Precedence: bulk X-Mailing-List: linux-arm-msm@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 X-ClientProxiedBy: nalasex01c.na.qualcomm.com (10.47.97.35) To nalasex01b.na.qualcomm.com (10.47.209.197) X-QCInternal: smtphost X-Proofpoint-Virus-Version: vendor=nai engine=6200 definitions=5800 signatures=585085 X-Proofpoint-ORIG-GUID: RKLDnP-AUjfudNtlYjgEb6PiK4jzZNfr X-Proofpoint-GUID: RKLDnP-AUjfudNtlYjgEb6PiK4jzZNfr X-Authority-Analysis: v=2.4 cv=CPUqXQrD c=1 sm=1 tr=0 ts=67ef2783 cx=c_pps a=ouPCqIW2jiPt+lZRy3xVPw==:117 a=ouPCqIW2jiPt+lZRy3xVPw==:17 a=3H110R4YSZwA:10 a=XR8D0OoHHMoA:10 a=COk6AnOGAAAA:8 a=dc6KhjMOD5a6MgxdBuQA:9 a=RVmHIydaz68A:10 a=TjNXssC_j7lpFel5tvFf:22 X-Proofpoint-Virus-Version: vendor=baseguard engine=ICAP:2.0.293,Aquarius:18.0.1095,Hydra:6.0.680,FMLib:17.12.68.34 definitions=2025-04-03_11,2025-04-03_03,2024-11-22_01 X-Proofpoint-Spam-Details: rule=outbound_notspam policy=outbound score=0 lowpriorityscore=0 impostorscore=0 clxscore=1015 malwarescore=0 suspectscore=0 spamscore=0 phishscore=0 adultscore=0 mlxlogscore=999 mlxscore=0 bulkscore=0 priorityscore=1501 classifier=spam authscore=0 authtc=n/a authcc= route=outbound adjust=0 reason=mlx scancount=1 engine=8.19.0-2502280000 definitions=main-2504040001 Create a USB BE component that will register a new USB port to the ASoC USB framework. This will handle determination on if the requested audio profile is supported by the USB device currently selected. Check for if the PCM format is supported during the hw_params callback. If the profile is not supported then the userspace ALSA entity will receive an error, and can take further action. Signed-off-by: Wesley Cheng --- include/sound/q6usboffload.h | 20 +++ sound/soc/qcom/Kconfig | 12 ++ sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6usb.c | 278 ++++++++++++++++++++++++++++++++++ 4 files changed, 311 insertions(+) create mode 100644 include/sound/q6usboffload.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c diff --git a/include/sound/q6usboffload.h b/include/sound/q6usboffload.h new file mode 100644 index 000000000000..35ae26ba6509 --- /dev/null +++ b/include/sound/q6usboffload.h @@ -0,0 +1,20 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * sound/q6usboffload.h -- QDSP6 USB offload + * + * Copyright (c) 2022-2025 Qualcomm Innovation Center, Inc. All rights reserved. + */ + +/** + * struct q6usb_offload - USB backend DAI link offload parameters + * @dev: dev handle to usb be + * @domain: allocated iommu domain + * @sid: streamID for iommu + * @intr_num: usb interrupter number + **/ +struct q6usb_offload { + struct device *dev; + struct iommu_domain *domain; + long long sid; + u16 intr_num; +}; diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index ca7a30ebd26a..1b4f3faadbc7 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -118,6 +118,18 @@ config SND_SOC_QDSP6_PRM tristate select SND_SOC_QDSP6_PRM_LPASS_CLOCKS +config SND_SOC_QDSP6_USB + tristate "SoC ALSA USB offloading backing for QDSP6" + depends on SND_SOC_USB + select AUXILIARY_BUS + + help + Adds support for USB offloading for QDSP6 ASoC + based platform sound cards. This will enable the + Q6USB DPCM backend DAI link, which will interact + with the SoC USB framework to initialize a session + with active USB SND devices. + config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" depends on QCOM_APR diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile index 26b7c55c9c11..67267304e7e9 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -17,3 +17,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_APM_DAI) += q6apm-dai.o obj-$(CONFIG_SND_SOC_QDSP6_APM_LPASS_DAI) += q6apm-lpass-dais.o obj-$(CONFIG_SND_SOC_QDSP6_PRM) += q6prm.o obj-$(CONFIG_SND_SOC_QDSP6_PRM_LPASS_CLOCKS) += q6prm-clocks.o +obj-$(CONFIG_SND_SOC_QDSP6_USB) += q6usb.o diff --git a/sound/soc/qcom/qdsp6/q6usb.c b/sound/soc/qcom/qdsp6/q6usb.c new file mode 100644 index 000000000000..cb8c4a62a816 --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6usb.c @@ -0,0 +1,278 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2022-2025 Qualcomm Innovation Center, Inc. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include + +#include "q6afe.h" +#include "q6dsp-lpass-ports.h" + +#define Q6_USB_SID_MASK 0xF + +struct q6usb_port_data { + struct auxiliary_device uauxdev; + struct q6afe_usb_cfg usb_cfg; + struct snd_soc_usb *usb; + struct q6usb_offload priv; + + /* Protects against operations between SOC USB and ASoC */ + struct mutex mutex; + struct list_head devices; +}; + +static const struct snd_soc_dapm_widget q6usb_dai_widgets[] = { + SND_SOC_DAPM_HP("USB_RX_BE", NULL), +}; + +static const struct snd_soc_dapm_route q6usb_dapm_routes[] = { + {"USB Playback", NULL, "USB_RX_BE"}, +}; + +static int q6usb_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct q6usb_port_data *data = dev_get_drvdata(dai->dev); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); + int direction = substream->stream; + struct q6afe_port *q6usb_afe; + struct snd_soc_usb_device *sdev; + int ret = -EINVAL; + + mutex_lock(&data->mutex); + + /* No active chip index */ + if (list_empty(&data->devices)) + goto out; + + sdev = list_last_entry(&data->devices, struct snd_soc_usb_device, list); + + ret = snd_soc_usb_find_supported_format(sdev->chip_idx, params, direction); + if (ret < 0) + goto out; + + q6usb_afe = q6afe_port_get_from_id(cpu_dai->dev, USB_RX); + if (IS_ERR(q6usb_afe)) + goto out; + + /* Notify audio DSP about the devices being offloaded */ + ret = afe_port_send_usb_dev_param(q6usb_afe, sdev->card_idx, + sdev->ppcm_idx[sdev->num_playback - 1]); + +out: + mutex_unlock(&data->mutex); + + return ret; +} + +static const struct snd_soc_dai_ops q6usb_ops = { + .hw_params = q6usb_hw_params, +}; + +static struct snd_soc_dai_driver q6usb_be_dais[] = { + { + .playback = { + .stream_name = "USB BE RX", + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | + SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE, + .channels_min = 1, + .channels_max = 2, + .rate_max = 192000, + .rate_min = 8000, + }, + .id = USB_RX, + .name = "USB_RX_BE", + .ops = &q6usb_ops, + }, +}; + +static int q6usb_audio_ports_of_xlate_dai_name(struct snd_soc_component *component, + const struct of_phandle_args *args, + const char **dai_name) +{ + int id = args->args[0]; + int ret = -EINVAL; + int i; + + for (i = 0; i < ARRAY_SIZE(q6usb_be_dais); i++) { + if (q6usb_be_dais[i].id == id) { + *dai_name = q6usb_be_dais[i].name; + ret = 0; + break; + } + } + + return ret; +} + +static int q6usb_alsa_connection_cb(struct snd_soc_usb *usb, + struct snd_soc_usb_device *sdev, bool connected) +{ + struct q6usb_port_data *data; + + if (!usb->component) + return -ENODEV; + + data = dev_get_drvdata(usb->component->dev); + + mutex_lock(&data->mutex); + if (connected) { + /* Selects the latest USB headset plugged in for offloading */ + list_add_tail(&sdev->list, &data->devices); + } else { + list_del(&sdev->list); + } + mutex_unlock(&data->mutex); + + return 0; +} + +static void q6usb_dai_aux_release(struct device *dev) {} + +static int q6usb_dai_add_aux_device(struct q6usb_port_data *data, + struct auxiliary_device *auxdev) +{ + int ret; + + auxdev->dev.parent = data->priv.dev; + auxdev->dev.release = q6usb_dai_aux_release; + auxdev->name = "qc-usb-audio-offload"; + + ret = auxiliary_device_init(auxdev); + if (ret) + return ret; + + ret = auxiliary_device_add(auxdev); + if (ret) + auxiliary_device_uninit(auxdev); + + return ret; +} + +static int q6usb_component_probe(struct snd_soc_component *component) +{ + struct q6usb_port_data *data = dev_get_drvdata(component->dev); + struct snd_soc_usb *usb; + int ret; + + /* Add the QC USB SND aux device */ + ret = q6usb_dai_add_aux_device(data, &data->uauxdev); + if (ret < 0) + return ret; + + usb = snd_soc_usb_allocate_port(component, &data->priv); + if (IS_ERR(usb)) + return -ENOMEM; + + usb->connection_status_cb = q6usb_alsa_connection_cb; + + snd_soc_usb_add_port(usb); + data->usb = usb; + + return 0; +} + +static void q6usb_component_remove(struct snd_soc_component *component) +{ + struct q6usb_port_data *data = dev_get_drvdata(component->dev); + + snd_soc_usb_remove_port(data->usb); + auxiliary_device_delete(&data->uauxdev); + auxiliary_device_uninit(&data->uauxdev); + snd_soc_usb_free_port(data->usb); +} + +static const struct snd_soc_component_driver q6usb_dai_component = { + .probe = q6usb_component_probe, + .remove = q6usb_component_remove, + .name = "q6usb-dai-component", + .dapm_widgets = q6usb_dai_widgets, + .num_dapm_widgets = ARRAY_SIZE(q6usb_dai_widgets), + .dapm_routes = q6usb_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(q6usb_dapm_routes), + .of_xlate_dai_name = q6usb_audio_ports_of_xlate_dai_name, +}; + +static int q6usb_dai_dev_probe(struct platform_device *pdev) +{ + struct device_node *node = pdev->dev.of_node; + struct q6usb_port_data *data; + struct device *dev = &pdev->dev; + struct of_phandle_args args; + int ret; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + ret = of_property_read_u16(node, "qcom,usb-audio-intr-idx", + &data->priv.intr_num); + if (ret) { + dev_err(&pdev->dev, "failed to read intr idx.\n"); + return ret; + } + + ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args); + if (ret < 0) + data->priv.sid = -1; + else + data->priv.sid = args.args[0] & Q6_USB_SID_MASK; + + data->priv.domain = iommu_get_domain_for_dev(&pdev->dev); + + data->priv.dev = dev; + INIT_LIST_HEAD(&data->devices); + dev_set_drvdata(dev, data); + + return devm_snd_soc_register_component(dev, &q6usb_dai_component, + q6usb_be_dais, ARRAY_SIZE(q6usb_be_dais)); +} + +static const struct of_device_id q6usb_dai_device_id[] = { + { .compatible = "qcom,q6usb" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6usb_dai_device_id); + +static struct platform_driver q6usb_dai_platform_driver = { + .driver = { + .name = "q6usb-dai", + .of_match_table = of_match_ptr(q6usb_dai_device_id), + }, + .probe = q6usb_dai_dev_probe, + /* + * Remove not required as resources are cleaned up as part of + * component removal. Others are device managed resources. + */ +}; +module_platform_driver(q6usb_dai_platform_driver); + +MODULE_DESCRIPTION("Q6 USB backend dai driver"); +MODULE_LICENSE("GPL"); From patchwork Fri Apr 4 00:27:25 2025 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Wesley Cheng X-Patchwork-Id: 14037918 Received: from mx0b-0031df01.pphosted.com (mx0b-0031df01.pphosted.com [205.220.180.131]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by smtp.subspace.kernel.org (Postfix) with ESMTPS id 8FA781FCCE7; Fri, 4 Apr 2025 00:28:10 +0000 (UTC) Authentication-Results: smtp.subspace.kernel.org; arc=none smtp.client-ip=205.220.180.131 ARC-Seal: i=1; a=rsa-sha256; d=subspace.kernel.org; s=arc-20240116; t=1743726496; cv=none; b=Gcc2kP/0ciyz3koiC43TKEXCfaRLU+ZpmHXcWYNJvK9nti+eaoPAvow/GBFdbAwzol63heDrUg0uKRqenjohgs+YwErFKt21WmflkuHYO40/rMUVtcL0gvvvvXbTWIP3w0nXPkI/WXMCVxBXOVDml75mYnE8xa2tEzlTnofpc6k= ARC-Message-Signature: i=1; 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Fri, 4 Apr 2025 00:27:47 GMT Received: from hu-wcheng-lv.qualcomm.com (10.49.16.6) by nalasex01b.na.qualcomm.com (10.47.209.197) with Microsoft SMTP Server (version=TLS1_2, cipher=TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384) id 15.2.1544.9; Thu, 3 Apr 2025 17:27:46 -0700 From: Wesley Cheng To: , , , , , , , , , , , , , CC: , , , , , , , Wesley Cheng Subject: [PATCH v37 28/31] ALSA: usb-audio: qcom: Introduce QC USB SND offloading support Date: Thu, 3 Apr 2025 17:27:25 -0700 Message-ID: <20250404002728.3590501-29-quic_wcheng@quicinc.com> X-Mailer: git-send-email 2.34.1 In-Reply-To: <20250404002728.3590501-1-quic_wcheng@quicinc.com> References: <20250404002728.3590501-1-quic_wcheng@quicinc.com> Precedence: bulk X-Mailing-List: linux-arm-msm@vger.kernel.org List-Id: List-Subscribe: List-Unsubscribe: MIME-Version: 1.0 X-ClientProxiedBy: nalasex01c.na.qualcomm.com (10.47.97.35) To nalasex01b.na.qualcomm.com (10.47.209.197) X-QCInternal: smtphost X-Proofpoint-Virus-Version: vendor=nai engine=6200 definitions=5800 signatures=585085 X-Proofpoint-ORIG-GUID: rwslUyzjogjeMZqtv3_twm3qhE5cEvu7 X-Proofpoint-GUID: rwslUyzjogjeMZqtv3_twm3qhE5cEvu7 X-Authority-Analysis: v=2.4 cv=CPUqXQrD c=1 sm=1 tr=0 ts=67ef2784 cx=c_pps a=ouPCqIW2jiPt+lZRy3xVPw==:117 a=ouPCqIW2jiPt+lZRy3xVPw==:17 a=3H110R4YSZwA:10 a=XR8D0OoHHMoA:10 a=COk6AnOGAAAA:8 a=YR8s8YPlvNioTnS7kv0A:9 a=RVmHIydaz68A:10 a=TjNXssC_j7lpFel5tvFf:22 X-Proofpoint-Virus-Version: vendor=baseguard engine=ICAP:2.0.293,Aquarius:18.0.1095,Hydra:6.0.680,FMLib:17.12.68.34 definitions=2025-04-03_11,2025-04-03_03,2024-11-22_01 X-Proofpoint-Spam-Details: rule=outbound_notspam policy=outbound score=0 lowpriorityscore=0 impostorscore=0 clxscore=1015 malwarescore=0 suspectscore=0 spamscore=0 phishscore=0 adultscore=0 mlxlogscore=999 mlxscore=0 bulkscore=0 priorityscore=1501 classifier=spam authscore=0 authtc=n/a authcc= route=outbound adjust=0 reason=mlx scancount=1 engine=8.19.0-2502280000 definitions=main-2504040001 Several Qualcomm SoCs have a dedicated audio DSP, which has the ability to support USB sound devices. This vendor driver will implement the required handshaking with the DSP, in order to pass along required resources that will be utilized by the DSP's USB SW. The communication channel used for this handshaking will be using the QMI protocol. Required resources include: - Allocated secondary event ring address - EP transfer ring address - Interrupter number The above information will allow for the audio DSP to execute USB transfers over the USB bus. It will also be able to support devices that have an implicit feedback and sync endpoint as well. Offloading these data transfers will allow the main/applications processor to enter lower CPU power modes, and sustain a longer duration in those modes. Audio offloading is initiated with the following sequence: 1. Userspace configures to route audio playback to USB backend and starts playback on the platform soundcard. 2. The Q6DSP AFE will communicate to the audio DSP to start the USB AFE port. 3. This results in a QMI packet with a STREAM enable command. 4. The QC audio offload driver will fetch the required resources, and pass this information as part of the QMI response to the STREAM enable command. 5. Once the QMI response is received the audio DSP will start queuing data on the USB bus. As part of step#2, the audio DSP is aware of the USB SND card and pcm device index that is being selected, and is communicated as part of the QMI request received by QC audio offload. These indices will be used to handle the stream enable QMI request. Signed-off-by: Wesley Cheng --- sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1987 +++++++++++++++++++++++++++++ 4 files changed, 2004 insertions(+), 1 deletion(-) create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 4a9569a3a39a..6daa551738da 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -176,6 +176,20 @@ config SND_BCD2000 To compile this driver as a module, choose M here: the module will be called snd-bcd2000. +config SND_USB_AUDIO_QMI + tristate "Qualcomm Audio Offload driver" + depends on QCOM_QMI_HELPERS && SND_USB_AUDIO && USB_XHCI_SIDEBAND && SND_SOC_USB + help + Say Y here to enable the Qualcomm USB audio offloading feature. + + This module sets up the required QMI stream enable/disable + responses to requests generated by the audio DSP. It passes the + USB transfer resource references, so that the audio DSP can issue + USB transfers to the host controller. + + To compile this driver as a module, choose M here: the module + will be called snd-usb-audio-qmi. + source "sound/usb/line6/Kconfig" endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 30bd5348477b..e62794a87e73 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -35,5 +35,5 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ qcom/ obj-$(CONFIG_SND_USB_LINE6) += line6/ diff --git a/sound/usb/qcom/Makefile b/sound/usb/qcom/Makefile new file mode 100644 index 000000000000..1eb51160e2e5 --- /dev/null +++ b/sound/usb/qcom/Makefile @@ -0,0 +1,2 @@ +snd-usb-audio-qmi-y := usb_audio_qmi_v01.o qc_audio_offload.o +obj-$(CONFIG_SND_USB_AUDIO_QMI) += snd-usb-audio-qmi.o diff --git a/sound/usb/qcom/qc_audio_offload.c b/sound/usb/qcom/qc_audio_offload.c new file mode 100644 index 000000000000..87e74c949370 --- /dev/null +++ b/sound/usb/qcom/qc_audio_offload.c @@ -0,0 +1,1987 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2022-2025 Qualcomm Innovation Center, Inc. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../usbaudio.h" +#include "../card.h" +#include "../endpoint.h" +#include "../format.h" +#include "../helper.h" +#include "../pcm.h" +#include "../power.h" + +#include "usb_audio_qmi_v01.h" + +/* Stream disable request timeout during USB device disconnect */ +#define DEV_RELEASE_WAIT_TIMEOUT 10000 /* in ms */ + +/* Data interval calculation parameters */ +#define BUS_INTERVAL_FULL_SPEED 1000 /* in us */ +#define BUS_INTERVAL_HIGHSPEED_AND_ABOVE 125 /* in us */ +#define MAX_BINTERVAL_ISOC_EP 16 + +#define QMI_STREAM_REQ_CARD_NUM_MASK 0xffff0000 +#define QMI_STREAM_REQ_DEV_NUM_MASK 0xff00 +#define QMI_STREAM_REQ_DIRECTION 0xff + +/* iommu resource parameters and management */ +#define PREPEND_SID_TO_IOVA(iova, sid) ((u64)(((u64)(iova)) | \ + (((u64)sid) << 32))) +#define IOVA_MASK(iova) (((u64)(iova)) & 0xFFFFFFFF) +#define IOVA_BASE 0x1000 +#define IOVA_XFER_RING_BASE (IOVA_BASE + PAGE_SIZE * (SNDRV_CARDS + 1)) +#define IOVA_XFER_BUF_BASE (IOVA_XFER_RING_BASE + PAGE_SIZE * SNDRV_CARDS * 32) +#define IOVA_XFER_RING_MAX (IOVA_XFER_BUF_BASE - PAGE_SIZE) +#define IOVA_XFER_BUF_MAX (0xfffff000 - PAGE_SIZE) + +#define MAX_XFER_BUFF_LEN (24 * PAGE_SIZE) + +struct iova_info { + struct list_head list; + unsigned long start_iova; + size_t size; + bool in_use; +}; + +struct intf_info { + /* IOMMU ring/buffer mapping information */ + unsigned long data_xfer_ring_va; + size_t data_xfer_ring_size; + unsigned long sync_xfer_ring_va; + size_t sync_xfer_ring_size; + unsigned long xfer_buf_va; + size_t xfer_buf_size; + phys_addr_t xfer_buf_pa; + u8 *xfer_buf; + + /* USB endpoint information */ + unsigned int data_ep_pipe; + unsigned int sync_ep_pipe; + unsigned int data_ep_idx; + unsigned int sync_ep_idx; + + u8 intf_num; + u8 pcm_card_num; + u8 pcm_dev_num; + u8 direction; + bool in_use; +}; + +struct uaudio_qmi_dev { + struct device *dev; + struct q6usb_offload *data; + struct auxiliary_device *auxdev; + + /* list to keep track of available iova */ + struct list_head xfer_ring_list; + size_t xfer_ring_iova_size; + unsigned long curr_xfer_ring_iova; + struct list_head xfer_buf_list; + size_t xfer_buf_iova_size; + unsigned long curr_xfer_buf_iova; + + /* bit fields representing pcm card enabled */ + unsigned long card_slot; + /* indicate event ring mapped or not */ + bool er_mapped; +}; + +struct uaudio_dev { + struct usb_device *udev; + /* audio control interface */ + struct usb_host_interface *ctrl_intf; + unsigned int usb_core_id; + atomic_t in_use; + struct kref kref; + wait_queue_head_t disconnect_wq; + + /* interface specific */ + int num_intf; + struct intf_info *info; + struct snd_usb_audio *chip; + + /* xhci sideband */ + struct xhci_sideband *sb; + + /* SoC USB device */ + struct snd_soc_usb_device *sdev; +}; + +static struct uaudio_dev uadev[SNDRV_CARDS]; +static struct uaudio_qmi_dev *uaudio_qdev; +static struct uaudio_qmi_svc *uaudio_svc; +static DEFINE_MUTEX(qdev_mutex); + +struct uaudio_qmi_svc { + struct qmi_handle *uaudio_svc_hdl; + struct sockaddr_qrtr client_sq; + bool client_connected; +}; + +enum mem_type { + MEM_EVENT_RING, + MEM_XFER_RING, + MEM_XFER_BUF, +}; + +/* Supported audio formats */ +enum usb_qmi_audio_format { + USB_QMI_PCM_FORMAT_S8 = 0, + USB_QMI_PCM_FORMAT_U8, + USB_QMI_PCM_FORMAT_S16_LE, + USB_QMI_PCM_FORMAT_S16_BE, + USB_QMI_PCM_FORMAT_U16_LE, + USB_QMI_PCM_FORMAT_U16_BE, + USB_QMI_PCM_FORMAT_S24_LE, + USB_QMI_PCM_FORMAT_S24_BE, + USB_QMI_PCM_FORMAT_U24_LE, + USB_QMI_PCM_FORMAT_U24_BE, + USB_QMI_PCM_FORMAT_S24_3LE, + USB_QMI_PCM_FORMAT_S24_3BE, + USB_QMI_PCM_FORMAT_U24_3LE, + USB_QMI_PCM_FORMAT_U24_3BE, + USB_QMI_PCM_FORMAT_S32_LE, + USB_QMI_PCM_FORMAT_S32_BE, + USB_QMI_PCM_FORMAT_U32_LE, + USB_QMI_PCM_FORMAT_U32_BE, +}; + +static int usb_qmi_get_pcm_num(struct snd_usb_audio *chip, int direction) +{ + struct snd_usb_substream *subs = NULL; + struct snd_usb_stream *as; + int count = 0; + + list_for_each_entry(as, &chip->pcm_list, list) { + subs = &as->substream[direction]; + if (subs->ep_num) + count++; + } + + return count; +} + +static enum usb_qmi_audio_device_speed_enum_v01 +get_speed_info(enum usb_device_speed udev_speed) +{ + switch (udev_speed) { + case USB_SPEED_LOW: + return USB_QMI_DEVICE_SPEED_LOW_V01; + case USB_SPEED_FULL: + return USB_QMI_DEVICE_SPEED_FULL_V01; + case USB_SPEED_HIGH: + return USB_QMI_DEVICE_SPEED_HIGH_V01; + case USB_SPEED_SUPER: + return USB_QMI_DEVICE_SPEED_SUPER_V01; + case USB_SPEED_SUPER_PLUS: + return USB_QMI_DEVICE_SPEED_SUPER_PLUS_V01; + default: + return USB_QMI_DEVICE_SPEED_INVALID_V01; + } +} + +static struct snd_usb_substream *find_substream(unsigned int card_num, + unsigned int pcm_idx, + unsigned int direction) +{ + struct snd_usb_substream *subs = NULL; + struct snd_usb_audio *chip; + struct snd_usb_stream *as; + + chip = uadev[card_num].chip; + if (!chip || atomic_read(&chip->shutdown)) + goto done; + + if (pcm_idx >= chip->pcm_devs) + goto done; + + if (direction > SNDRV_PCM_STREAM_CAPTURE) + goto done; + + list_for_each_entry(as, &chip->pcm_list, list) { + if (as->pcm_index == pcm_idx) { + subs = &as->substream[direction]; + goto done; + } + } + +done: + return subs; +} + +static int info_idx_from_ifnum(int card_num, int intf_num, bool enable) +{ + int i; + + /* + * default index 0 is used when info is allocated upon + * first enable audio stream req for a pcm device + */ + if (enable && !uadev[card_num].info) + return 0; + + for (i = 0; i < uadev[card_num].num_intf; i++) { + if (enable && !uadev[card_num].info[i].in_use) + return i; + else if (!enable && + uadev[card_num].info[i].intf_num == intf_num) + return i; + } + + return -EINVAL; +} + +static int get_data_interval_from_si(struct snd_usb_substream *subs, + u32 service_interval) +{ + unsigned int bus_intval_mult; + unsigned int bus_intval; + unsigned int binterval; + + if (subs->dev->speed >= USB_SPEED_HIGH) + bus_intval = BUS_INTERVAL_HIGHSPEED_AND_ABOVE; + else + bus_intval = BUS_INTERVAL_FULL_SPEED; + + if (service_interval % bus_intval) + return -EINVAL; + + bus_intval_mult = service_interval / bus_intval; + binterval = ffs(bus_intval_mult); + if (!binterval || binterval > MAX_BINTERVAL_ISOC_EP) + return -EINVAL; + + /* check if another bit is set then bail out */ + bus_intval_mult = bus_intval_mult >> binterval; + if (bus_intval_mult) + return -EINVAL; + + return (binterval - 1); +} + +/* maps audio format received over QMI to asound.h based pcm format */ +static snd_pcm_format_t map_pcm_format(enum usb_qmi_audio_format fmt_received) +{ + switch (fmt_received) { + case USB_QMI_PCM_FORMAT_S8: + return SNDRV_PCM_FORMAT_S8; + case USB_QMI_PCM_FORMAT_U8: + return SNDRV_PCM_FORMAT_U8; + case USB_QMI_PCM_FORMAT_S16_LE: + return SNDRV_PCM_FORMAT_S16_LE; + case USB_QMI_PCM_FORMAT_S16_BE: + return SNDRV_PCM_FORMAT_S16_BE; + case USB_QMI_PCM_FORMAT_U16_LE: + return SNDRV_PCM_FORMAT_U16_LE; + case USB_QMI_PCM_FORMAT_U16_BE: + return SNDRV_PCM_FORMAT_U16_BE; + case USB_QMI_PCM_FORMAT_S24_LE: + return SNDRV_PCM_FORMAT_S24_LE; + case USB_QMI_PCM_FORMAT_S24_BE: + return SNDRV_PCM_FORMAT_S24_BE; + case USB_QMI_PCM_FORMAT_U24_LE: + return SNDRV_PCM_FORMAT_U24_LE; + case USB_QMI_PCM_FORMAT_U24_BE: + return SNDRV_PCM_FORMAT_U24_BE; + case USB_QMI_PCM_FORMAT_S24_3LE: + return SNDRV_PCM_FORMAT_S24_3LE; + case USB_QMI_PCM_FORMAT_S24_3BE: + return SNDRV_PCM_FORMAT_S24_3BE; + case USB_QMI_PCM_FORMAT_U24_3LE: + return SNDRV_PCM_FORMAT_U24_3LE; + case USB_QMI_PCM_FORMAT_U24_3BE: + return SNDRV_PCM_FORMAT_U24_3BE; + case USB_QMI_PCM_FORMAT_S32_LE: + return SNDRV_PCM_FORMAT_S32_LE; + case USB_QMI_PCM_FORMAT_S32_BE: + return SNDRV_PCM_FORMAT_S32_BE; + case USB_QMI_PCM_FORMAT_U32_LE: + return SNDRV_PCM_FORMAT_U32_LE; + case USB_QMI_PCM_FORMAT_U32_BE: + return SNDRV_PCM_FORMAT_U32_BE; + default: + /* + * We expect the caller to do input validation so we should + * never hit this. But we do have to return a proper + * snd_pcm_format_t value due to the __bitwise attribute; so + * just return the equivalent of 0 in case of bad input. + */ + return SNDRV_PCM_FORMAT_S8; + } +} + +/* + * Sends QMI disconnect indication message, assumes chip->mutex and qdev_mutex + * lock held by caller. + */ +static int uaudio_send_disconnect_ind(struct snd_usb_audio *chip) +{ + struct qmi_uaudio_stream_ind_msg_v01 disconnect_ind = {0}; + struct uaudio_qmi_svc *svc = uaudio_svc; + struct uaudio_dev *dev; + int ret = 0; + + dev = &uadev[chip->card->number]; + + if (atomic_read(&dev->in_use)) { + mutex_unlock(&chip->mutex); + mutex_unlock(&qdev_mutex); + dev_dbg(uaudio_qdev->data->dev, "sending qmi indication suspend\n"); + disconnect_ind.dev_event = USB_QMI_DEV_DISCONNECT_V01; + disconnect_ind.slot_id = dev->udev->slot_id; + disconnect_ind.controller_num = dev->usb_core_id; + disconnect_ind.controller_num_valid = 1; + ret = qmi_send_indication(svc->uaudio_svc_hdl, &svc->client_sq, + QMI_UAUDIO_STREAM_IND_V01, + QMI_UAUDIO_STREAM_IND_MSG_V01_MAX_MSG_LEN, + qmi_uaudio_stream_ind_msg_v01_ei, + &disconnect_ind); + if (ret < 0) + dev_err(uaudio_qdev->data->dev, + "qmi send failed with err: %d\n", ret); + + ret = wait_event_interruptible_timeout(dev->disconnect_wq, + !atomic_read(&dev->in_use), + msecs_to_jiffies(DEV_RELEASE_WAIT_TIMEOUT)); + if (!ret) { + dev_err(uaudio_qdev->data->dev, + "timeout while waiting for dev_release\n"); + atomic_set(&dev->in_use, 0); + } else if (ret < 0) { + dev_err(uaudio_qdev->data->dev, + "failed with ret %d\n", ret); + atomic_set(&dev->in_use, 0); + } + mutex_lock(&qdev_mutex); + mutex_lock(&chip->mutex); + } + + return ret; +} + +/* Offloading IOMMU management */ +static unsigned long uaudio_get_iova(unsigned long *curr_iova, + size_t *curr_iova_size, + struct list_head *head, size_t size) +{ + struct iova_info *info, *new_info = NULL; + struct list_head *curr_head; + size_t tmp_size = size; + unsigned long va = 0; + + if (size % PAGE_SIZE) + goto done; + + if (size > *curr_iova_size) + goto done; + + if (*curr_iova_size == 0) + goto done; + + list_for_each_entry(info, head, list) { + /* exact size iova_info */ + if (!info->in_use && info->size == size) { + info->in_use = true; + va = info->start_iova; + *curr_iova_size -= size; + goto done; + } else if (!info->in_use && tmp_size >= info->size) { + if (!new_info) + new_info = info; + tmp_size -= info->size; + if (tmp_size) + continue; + + va = new_info->start_iova; + for (curr_head = &new_info->list; curr_head != + &info->list; curr_head = curr_head->next) { + new_info = list_entry(curr_head, struct + iova_info, list); + new_info->in_use = true; + } + info->in_use = true; + *curr_iova_size -= size; + goto done; + } else { + /* iova region in use */ + new_info = NULL; + tmp_size = size; + } + } + + info = kzalloc(sizeof(*info), GFP_KERNEL); + if (!info) { + va = 0; + goto done; + } + + va = *curr_iova; + info->start_iova = *curr_iova; + info->size = size; + info->in_use = true; + *curr_iova += size; + *curr_iova_size -= size; + list_add_tail(&info->list, head); + +done: + return va; +} + +static void uaudio_put_iova(unsigned long va, size_t size, struct list_head + *head, size_t *curr_iova_size) +{ + struct iova_info *info; + size_t tmp_size = size; + bool found = false; + + list_for_each_entry(info, head, list) { + if (info->start_iova == va) { + if (!info->in_use) + return; + + found = true; + info->in_use = false; + if (info->size == size) + goto done; + } + + if (found && tmp_size >= info->size) { + info->in_use = false; + tmp_size -= info->size; + if (!tmp_size) + goto done; + } + } + + if (!found) + return; + +done: + *curr_iova_size += size; +} + +/** + * uaudio_iommu_unmap() - unmaps iommu memory for adsp + * @mtype: ring type + * @va: virtual address to unmap + * @iova_size: region size + * @mapped_iova_size: mapped region size + * + * Unmaps the memory region that was previously assigned to the adsp. + * + */ +static void uaudio_iommu_unmap(enum mem_type mtype, unsigned long va, + size_t iova_size, size_t mapped_iova_size) +{ + size_t umap_size; + bool unmap = true; + + if (!va || !iova_size) + return; + + switch (mtype) { + case MEM_EVENT_RING: + if (uaudio_qdev->er_mapped) + uaudio_qdev->er_mapped = false; + else + unmap = false; + break; + + case MEM_XFER_RING: + uaudio_put_iova(va, iova_size, &uaudio_qdev->xfer_ring_list, + &uaudio_qdev->xfer_ring_iova_size); + break; + case MEM_XFER_BUF: + uaudio_put_iova(va, iova_size, &uaudio_qdev->xfer_buf_list, + &uaudio_qdev->xfer_buf_iova_size); + break; + default: + unmap = false; + } + + if (!unmap || !mapped_iova_size) + return; + + umap_size = iommu_unmap(uaudio_qdev->data->domain, va, mapped_iova_size); + if (umap_size != mapped_iova_size) + dev_err(uaudio_qdev->data->dev, + "unmapped size %zu for iova 0x%08lx of mapped size %zu\n", + umap_size, va, mapped_iova_size); +} + +/** + * uaudio_iommu_map() - maps iommu memory for adsp + * @mtype: ring type + * @dma_coherent: dma coherent + * @pa: physical address for ring/buffer + * @size: size of memory region + * @sgt: sg table for memory region + * + * Maps the XHCI related resources to a memory region that is assigned to be + * used by the adsp. This will be mapped to the domain, which is created by + * the ASoC USB backend driver. + * + */ +static unsigned long uaudio_iommu_map(enum mem_type mtype, bool dma_coherent, + phys_addr_t pa, size_t size, + struct sg_table *sgt) +{ + struct scatterlist *sg; + unsigned long va = 0; + size_t total_len = 0; + unsigned long va_sg; + phys_addr_t pa_sg; + bool map = true; + size_t sg_len; + int prot; + int ret; + int i; + + prot = IOMMU_READ | IOMMU_WRITE; + + if (dma_coherent) + prot |= IOMMU_CACHE; + + switch (mtype) { + case MEM_EVENT_RING: + va = IOVA_BASE; + /* er already mapped */ + if (uaudio_qdev->er_mapped) + map = false; + break; + case MEM_XFER_RING: + va = uaudio_get_iova(&uaudio_qdev->curr_xfer_ring_iova, + &uaudio_qdev->xfer_ring_iova_size, + &uaudio_qdev->xfer_ring_list, size); + break; + case MEM_XFER_BUF: + va = uaudio_get_iova(&uaudio_qdev->curr_xfer_buf_iova, + &uaudio_qdev->xfer_buf_iova_size, + &uaudio_qdev->xfer_buf_list, size); + break; + default: + dev_err(uaudio_qdev->data->dev, "unknown mem type %d\n", mtype); + } + + if (!va || !map) + goto done; + + if (!sgt) + goto skip_sgt_map; + + va_sg = va; + for_each_sg(sgt->sgl, sg, sgt->nents, i) { + sg_len = PAGE_ALIGN(sg->offset + sg->length); + pa_sg = page_to_phys(sg_page(sg)); + ret = iommu_map(uaudio_qdev->data->domain, va_sg, pa_sg, sg_len, + prot, GFP_KERNEL); + if (ret) { + uaudio_iommu_unmap(MEM_XFER_BUF, va, size, total_len); + va = 0; + goto done; + } + + va_sg += sg_len; + total_len += sg_len; + } + + if (size != total_len) { + uaudio_iommu_unmap(MEM_XFER_BUF, va, size, total_len); + va = 0; + } + return va; + +skip_sgt_map: + iommu_map(uaudio_qdev->data->domain, va, pa, size, prot, GFP_KERNEL); + +done: + return va; +} + +/* looks up alias, if any, for controller DT node and returns the index */ +static int usb_get_controller_id(struct usb_device *udev) +{ + if (udev->bus->sysdev && udev->bus->sysdev->of_node) + return of_alias_get_id(udev->bus->sysdev->of_node, "usb"); + + return -ENODEV; +} + +/** + * uaudio_dev_intf_cleanup() - cleanup transfer resources + * @udev: usb device + * @info: usb offloading interface + * + * Cleans up the transfer ring related resources which are assigned per + * endpoint from XHCI. This is invoked when the USB endpoints are no + * longer in use by the adsp. + * + */ +static void uaudio_dev_intf_cleanup(struct usb_device *udev, struct intf_info *info) +{ + uaudio_iommu_unmap(MEM_XFER_RING, info->data_xfer_ring_va, + info->data_xfer_ring_size, info->data_xfer_ring_size); + info->data_xfer_ring_va = 0; + info->data_xfer_ring_size = 0; + + uaudio_iommu_unmap(MEM_XFER_RING, info->sync_xfer_ring_va, + info->sync_xfer_ring_size, info->sync_xfer_ring_size); + info->sync_xfer_ring_va = 0; + info->sync_xfer_ring_size = 0; + + uaudio_iommu_unmap(MEM_XFER_BUF, info->xfer_buf_va, info->xfer_buf_size, + info->xfer_buf_size); + info->xfer_buf_va = 0; + + usb_free_coherent(udev, info->xfer_buf_size, info->xfer_buf, + info->xfer_buf_pa); + info->xfer_buf_size = 0; + info->xfer_buf = NULL; + info->xfer_buf_pa = 0; + + info->in_use = false; +} + +/** + * uaudio_event_ring_cleanup_free() - cleanup secondary event ring + * @dev: usb offload device + * + * Cleans up the secondary event ring that was requested. This will + * occur when the adsp is no longer transferring data on the USB bus + * across all endpoints. + * + */ +static void uaudio_event_ring_cleanup_free(struct uaudio_dev *dev) +{ + clear_bit(dev->chip->card->number, &uaudio_qdev->card_slot); + /* all audio devices are disconnected */ + if (!uaudio_qdev->card_slot) { + uaudio_iommu_unmap(MEM_EVENT_RING, IOVA_BASE, PAGE_SIZE, + PAGE_SIZE); + xhci_sideband_remove_interrupter(uadev[dev->chip->card->number].sb); + } +} + +static void uaudio_dev_cleanup(struct uaudio_dev *dev) +{ + int if_idx; + + if (!dev->udev) + return; + + /* free xfer buffer and unmap xfer ring and buf per interface */ + for (if_idx = 0; if_idx < dev->num_intf; if_idx++) { + if (!dev->info[if_idx].in_use) + continue; + uaudio_dev_intf_cleanup(dev->udev, &dev->info[if_idx]); + dev_dbg(uaudio_qdev->data->dev, + "release resources: intf# %d card# %d\n", + dev->info[if_idx].intf_num, dev->chip->card->number); + } + + dev->num_intf = 0; + + /* free interface info */ + kfree(dev->info); + dev->info = NULL; + uaudio_event_ring_cleanup_free(dev); + dev->udev = NULL; +} + +/** + * disable_audio_stream() - disable usb snd endpoints + * @subs: usb substream + * + * Closes the USB SND endpoints associated with the current audio stream + * used. This will decrement the USB SND endpoint opened reference count. + * + */ +static void disable_audio_stream(struct snd_usb_substream *subs) +{ + struct snd_usb_audio *chip = subs->stream->chip; + + snd_usb_hw_free(subs); + snd_usb_autosuspend(chip); +} + +/* QMI service disconnect handlers */ +static void qmi_stop_session(void) +{ + struct snd_usb_substream *subs; + struct usb_host_endpoint *ep; + struct snd_usb_audio *chip; + struct intf_info *info; + int pcm_card_num; + int if_idx; + int idx; + + mutex_lock(&qdev_mutex); + /* find all active intf for set alt 0 and cleanup usb audio dev */ + for (idx = 0; idx < SNDRV_CARDS; idx++) { + if (!atomic_read(&uadev[idx].in_use)) + continue; + + chip = uadev[idx].chip; + for (if_idx = 0; if_idx < uadev[idx].num_intf; if_idx++) { + if (!uadev[idx].info || !uadev[idx].info[if_idx].in_use) + continue; + info = &uadev[idx].info[if_idx]; + pcm_card_num = info->pcm_card_num; + subs = find_substream(pcm_card_num, info->pcm_dev_num, + info->direction); + if (!subs || !chip || atomic_read(&chip->shutdown)) { + dev_err(&subs->dev->dev, + "no sub for c#%u dev#%u dir%u\n", + info->pcm_card_num, + info->pcm_dev_num, + info->direction); + continue; + } + /* Release XHCI endpoints */ + if (info->data_ep_pipe) + ep = usb_pipe_endpoint(uadev[pcm_card_num].udev, + info->data_ep_pipe); + xhci_sideband_remove_endpoint(uadev[pcm_card_num].sb, ep); + + if (info->sync_ep_pipe) + ep = usb_pipe_endpoint(uadev[pcm_card_num].udev, + info->sync_ep_pipe); + xhci_sideband_remove_endpoint(uadev[pcm_card_num].sb, ep); + + disable_audio_stream(subs); + } + atomic_set(&uadev[idx].in_use, 0); + mutex_lock(&chip->mutex); + uaudio_dev_cleanup(&uadev[idx]); + mutex_unlock(&chip->mutex); + } + mutex_unlock(&qdev_mutex); +} + +/** + * uaudio_sideband_notifier() - xHCI sideband event handler + * @intf: USB interface handle + * @evt: xHCI sideband event type + * + * This callback is executed when the xHCI sideband encounters a sequence + * that requires the sideband clients to take action. An example, is when + * xHCI frees the transfer ring, so the client has to ensure that the + * offload path is halted. + * + */ +static int uaudio_sideband_notifier(struct usb_interface *intf, + struct xhci_sideband_event *evt) +{ + struct snd_usb_audio *chip; + struct uaudio_dev *dev; + int if_idx; + + if (!intf || !evt) + return 0; + + chip = usb_get_intfdata(intf); + + mutex_lock(&qdev_mutex); + mutex_lock(&chip->mutex); + + dev = &uadev[chip->card->number]; + + if (evt->type == XHCI_SIDEBAND_XFER_RING_FREE) { + unsigned int *ep = (unsigned int *) evt->evt_data; + + for (if_idx = 0; if_idx < dev->num_intf; if_idx++) { + if (dev->info[if_idx].data_ep_idx == *ep || + dev->info[if_idx].sync_ep_idx == *ep) + uaudio_send_disconnect_ind(chip); + } + } + + mutex_unlock(&qdev_mutex); + mutex_unlock(&chip->mutex); + + return 0; +} + +/** + * qmi_bye_cb() - qmi bye message callback + * @handle: QMI handle + * @node: id of the dying node + * + * This callback is invoked when the QMI bye control message is received + * from the QMI client. Handle the message accordingly by ensuring that + * the USB offload path is disabled and cleaned up. At this point, ADSP + * is not utilizing the USB bus. + * + */ +static void qmi_bye_cb(struct qmi_handle *handle, unsigned int node) +{ + struct uaudio_qmi_svc *svc = uaudio_svc; + + if (svc->uaudio_svc_hdl != handle) + return; + + if (svc->client_connected && svc->client_sq.sq_node == node) { + qmi_stop_session(); + + /* clear QMI client parameters to block further QMI messages */ + svc->client_sq.sq_node = 0; + svc->client_sq.sq_port = 0; + svc->client_sq.sq_family = 0; + svc->client_connected = false; + } +} + +/** + * qmi_svc_disconnect_cb() - qmi client disconnected + * @handle: QMI handle + * @node: id of the dying node + * @port: port of the dying client + * + * Invoked when the remote QMI client is disconnected. Handle this event + * the same way as when the QMI bye message is received. This will ensure + * the USB offloading path is disabled and cleaned up. + * + */ +static void qmi_svc_disconnect_cb(struct qmi_handle *handle, + unsigned int node, unsigned int port) +{ + struct uaudio_qmi_svc *svc; + + if (!uaudio_svc) + return; + + svc = uaudio_svc; + if (svc->uaudio_svc_hdl != handle) + return; + + if (svc->client_connected && svc->client_sq.sq_node == node && + svc->client_sq.sq_port == port) { + qmi_stop_session(); + + /* clear QMI client parameters to block further QMI messages */ + svc->client_sq.sq_node = 0; + svc->client_sq.sq_port = 0; + svc->client_sq.sq_family = 0; + svc->client_connected = false; + } +} + +/* QMI client callback handlers from QMI interface */ +static struct qmi_ops uaudio_svc_ops_options = { + .bye = qmi_bye_cb, + .del_client = qmi_svc_disconnect_cb, +}; + +/* kref release callback when all streams are disabled */ +static void uaudio_dev_release(struct kref *kref) +{ + struct uaudio_dev *dev = container_of(kref, struct uaudio_dev, kref); + + uaudio_event_ring_cleanup_free(dev); + atomic_set(&dev->in_use, 0); + wake_up(&dev->disconnect_wq); +} + +/** + * enable_audio_stream() - enable usb snd endpoints + * @subs: usb substream + * @pcm_format: pcm format requested + * @channels: number of channels + * @cur_rate: sample rate + * @datainterval: interval + * + * Opens all USB SND endpoints used for the data interface. This will increment + * the USB SND endpoint's opened count. Requests to keep the interface resumed + * until the audio stream is stopped. Will issue the USB set interface control + * message to enable the data interface. + * + */ +static int enable_audio_stream(struct snd_usb_substream *subs, + snd_pcm_format_t pcm_format, + unsigned int channels, unsigned int cur_rate, + int datainterval) +{ + struct snd_pcm_hw_params params; + struct snd_usb_audio *chip; + struct snd_interval *i; + struct snd_mask *m; + int ret; + + chip = subs->stream->chip; + + _snd_pcm_hw_params_any(¶ms); + + m = hw_param_mask(¶ms, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_leave(m, pcm_format); + + i = hw_param_interval(¶ms, SNDRV_PCM_HW_PARAM_CHANNELS); + snd_interval_setinteger(i); + i->min = channels; + i->max = channels; + + i = hw_param_interval(¶ms, SNDRV_PCM_HW_PARAM_RATE); + snd_interval_setinteger(i); + i->min = cur_rate; + i->max = cur_rate; + + pm_runtime_barrier(&chip->intf[0]->dev); + snd_usb_autoresume(chip); + + ret = snd_usb_hw_params(subs, ¶ms); + if (ret < 0) + goto put_suspend; + + if (!atomic_read(&chip->shutdown)) { + ret = snd_usb_lock_shutdown(chip); + if (ret < 0) + goto detach_ep; + + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); + if (ret < 0) + goto unlock; + + snd_usb_unlock_shutdown(chip); + + dev_dbg(uaudio_qdev->data->dev, + "selected %s iface:%d altsetting:%d datainterval:%dus\n", + subs->direction ? "capture" : "playback", + subs->cur_audiofmt->iface, subs->cur_audiofmt->altsetting, + (1 << subs->cur_audiofmt->datainterval) * + (subs->dev->speed >= USB_SPEED_HIGH ? + BUS_INTERVAL_HIGHSPEED_AND_ABOVE : + BUS_INTERVAL_FULL_SPEED)); + } + + return 0; + +unlock: + snd_usb_unlock_shutdown(chip); + +detach_ep: + snd_usb_hw_free(subs); + +put_suspend: + snd_usb_autosuspend(chip); + + return ret; +} + +/** + * uaudio_transfer_buffer_setup() - fetch and populate xfer buffer params + * @subs: usb substream + * @xfer_buf: xfer buf to be allocated + * @xfer_buf_len: size of allocation + * @mem_info: QMI response info + * + * Allocates and maps the transfer buffers that will be utilized by the + * audio DSP. Will populate the information in the QMI response that is + * sent back to the stream enable request. + * + */ +static int uaudio_transfer_buffer_setup(struct snd_usb_substream *subs, + u8 *xfer_buf, u32 xfer_buf_len, + struct mem_info_v01 *mem_info) +{ + struct sg_table xfer_buf_sgt; + phys_addr_t xfer_buf_pa; + u32 len = xfer_buf_len; + bool dma_coherent; + unsigned long va; + u32 remainder; + u32 mult; + int ret; + + dma_coherent = dev_is_dma_coherent(subs->dev->bus->sysdev); + + /* xfer buffer, multiple of 4K only */ + if (!len) + len = PAGE_SIZE; + + mult = len / PAGE_SIZE; + remainder = len % PAGE_SIZE; + len = mult * PAGE_SIZE; + len += remainder ? PAGE_SIZE : 0; + + if (len > MAX_XFER_BUFF_LEN) { + dev_err(uaudio_qdev->data->dev, + "req buf len %d > max buf len %lu, setting %lu\n", + len, MAX_XFER_BUFF_LEN, MAX_XFER_BUFF_LEN); + len = MAX_XFER_BUFF_LEN; + } + + xfer_buf = usb_alloc_coherent(subs->dev, len, GFP_KERNEL, &xfer_buf_pa); + if (!xfer_buf) + return -ENOMEM; + + dma_get_sgtable(subs->dev->bus->sysdev, &xfer_buf_sgt, xfer_buf, + xfer_buf_pa, len); + va = uaudio_iommu_map(MEM_XFER_BUF, dma_coherent, xfer_buf_pa, len, + &xfer_buf_sgt); + if (!va) { + ret = -ENOMEM; + goto unmap_sync; + } + + mem_info->pa = xfer_buf_pa; + mem_info->size = len; + mem_info->va = PREPEND_SID_TO_IOVA(va, uaudio_qdev->data->sid); + sg_free_table(&xfer_buf_sgt); + + return 0; + +unmap_sync: + usb_free_coherent(subs->dev, len, xfer_buf, xfer_buf_pa); + + return ret; +} + +/** + * uaudio_endpoint_setup() - fetch and populate endpoint params + * @subs: usb substream + * @endpoint: usb endpoint to add + * @card_num: uadev index + * @mem_info: QMI response info + * @ep_desc: QMI ep desc response field + * + * Initialize the USB endpoint being used for a particular USB + * stream. Will request XHCI sec intr to reserve the EP for + * offloading as well as populating the QMI response with the + * transfer ring parameters. + * + */ +static phys_addr_t +uaudio_endpoint_setup(struct snd_usb_substream *subs, + struct snd_usb_endpoint *endpoint, int card_num, + struct mem_info_v01 *mem_info, + struct usb_endpoint_descriptor_v01 *ep_desc) +{ + struct usb_host_endpoint *ep; + phys_addr_t tr_pa = 0; + struct sg_table *sgt; + bool dma_coherent; + unsigned long va; + struct page *pg; + int ret = -ENODEV; + + dma_coherent = dev_is_dma_coherent(subs->dev->bus->sysdev); + + ep = usb_pipe_endpoint(subs->dev, endpoint->pipe); + if (!ep) { + dev_err(uaudio_qdev->data->dev, "data ep # %d context is null\n", + subs->data_endpoint->ep_num); + goto exit; + } + + memcpy(ep_desc, &ep->desc, sizeof(ep->desc)); + + ret = xhci_sideband_add_endpoint(uadev[card_num].sb, ep); + if (ret < 0) { + dev_err(&subs->dev->dev, + "failed to add data ep to sec intr\n"); + ret = -ENODEV; + goto exit; + } + + sgt = xhci_sideband_get_endpoint_buffer(uadev[card_num].sb, ep); + if (!sgt) { + dev_err(&subs->dev->dev, + "failed to get data ep ring address\n"); + ret = -ENODEV; + goto remove_ep; + } + + pg = sg_page(sgt->sgl); + tr_pa = page_to_phys(pg); + mem_info->pa = sg_dma_address(sgt->sgl); + sg_free_table(sgt); + + /* data transfer ring */ + va = uaudio_iommu_map(MEM_XFER_RING, dma_coherent, tr_pa, + PAGE_SIZE, NULL); + if (!va) { + ret = -ENOMEM; + goto clear_pa; + } + + mem_info->va = PREPEND_SID_TO_IOVA(va, uaudio_qdev->data->sid); + mem_info->size = PAGE_SIZE; + + return 0; + +clear_pa: + mem_info->pa = 0; +remove_ep: + xhci_sideband_remove_endpoint(uadev[card_num].sb, ep); +exit: + return ret; +} + +/** + * uaudio_event_ring_setup() - fetch and populate event ring params + * @subs: usb substream + * @card_num: uadev index + * @mem_info: QMI response info + * + * Register secondary interrupter to XHCI and fetch the event buffer info + * and populate the information into the QMI response. + * + */ +static int uaudio_event_ring_setup(struct snd_usb_substream *subs, + int card_num, struct mem_info_v01 *mem_info) +{ + struct sg_table *sgt; + phys_addr_t er_pa; + bool dma_coherent; + unsigned long va; + struct page *pg; + int ret; + + dma_coherent = dev_is_dma_coherent(subs->dev->bus->sysdev); + er_pa = 0; + + /* event ring */ + ret = xhci_sideband_create_interrupter(uadev[card_num].sb, 1, false, + 0, uaudio_qdev->data->intr_num); + if (ret < 0) { + dev_err(&subs->dev->dev, "failed to fetch interrupter\n"); + goto exit; + } + + sgt = xhci_sideband_get_event_buffer(uadev[card_num].sb); + if (!sgt) { + dev_err(&subs->dev->dev, + "failed to get event ring address\n"); + ret = -ENODEV; + goto remove_interrupter; + } + + pg = sg_page(sgt->sgl); + er_pa = page_to_phys(pg); + mem_info->pa = sg_dma_address(sgt->sgl); + sg_free_table(sgt); + + va = uaudio_iommu_map(MEM_EVENT_RING, dma_coherent, er_pa, + PAGE_SIZE, NULL); + if (!va) { + ret = -ENOMEM; + goto clear_pa; + } + + mem_info->va = PREPEND_SID_TO_IOVA(va, uaudio_qdev->data->sid); + mem_info->size = PAGE_SIZE; + + return 0; + +clear_pa: + mem_info->pa = 0; +remove_interrupter: + xhci_sideband_remove_interrupter(uadev[card_num].sb); +exit: + return ret; +} + +/** + * uaudio_populate_uac_desc() - parse UAC parameters and populate QMI resp + * @subs: usb substream + * @resp: QMI response buffer + * + * Parses information specified within UAC descriptors which explain the + * sample parameters that the device expects. This information is populated + * to the QMI response sent back to the audio DSP. + * + */ +static int uaudio_populate_uac_desc(struct snd_usb_substream *subs, + struct qmi_uaudio_stream_resp_msg_v01 *resp) +{ + struct usb_interface_descriptor *altsd; + struct usb_host_interface *alts; + struct usb_interface *iface; + int protocol; + + iface = usb_ifnum_to_if(subs->dev, subs->cur_audiofmt->iface); + if (!iface) { + dev_err(&subs->dev->dev, "interface # %d does not exist\n", + subs->cur_audiofmt->iface); + return -ENODEV; + } + + alts = &iface->altsetting[subs->cur_audiofmt->altset_idx]; + altsd = get_iface_desc(alts); + protocol = altsd->bInterfaceProtocol; + + if (protocol == UAC_VERSION_1) { + struct uac1_as_header_descriptor *as; + + as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, + UAC_AS_GENERAL); + if (!as) { + dev_err(&subs->dev->dev, + "%u:%d : no UAC_AS_GENERAL desc\n", + subs->cur_audiofmt->iface, + subs->cur_audiofmt->altset_idx); + return -ENODEV; + } + + resp->data_path_delay = as->bDelay; + resp->data_path_delay_valid = 1; + + resp->usb_audio_subslot_size = subs->cur_audiofmt->fmt_sz; + resp->usb_audio_subslot_size_valid = 1; + + resp->usb_audio_spec_revision = le16_to_cpu((__force __le16)0x0100); + resp->usb_audio_spec_revision_valid = 1; + } else if (protocol == UAC_VERSION_2) { + resp->usb_audio_subslot_size = subs->cur_audiofmt->fmt_sz; + resp->usb_audio_subslot_size_valid = 1; + + resp->usb_audio_spec_revision = le16_to_cpu((__force __le16)0x0200); + resp->usb_audio_spec_revision_valid = 1; + } else if (protocol == UAC_VERSION_3) { + if (iface->intf_assoc->bFunctionSubClass == + UAC3_FUNCTION_SUBCLASS_FULL_ADC_3_0) { + dev_err(&subs->dev->dev, + "full adc is not supported\n"); + return -EINVAL; + } + + switch (le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize)) { + case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_16: + case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_16: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_16: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_16: { + resp->usb_audio_subslot_size = 0x2; + break; + } + + case UAC3_BADD_EP_MAXPSIZE_SYNC_MONO_24: + case UAC3_BADD_EP_MAXPSIZE_SYNC_STEREO_24: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_MONO_24: + case UAC3_BADD_EP_MAXPSIZE_ASYNC_STEREO_24: { + resp->usb_audio_subslot_size = 0x3; + break; + } + + default: + dev_err(&subs->dev->dev, + "%d: %u: Invalid wMaxPacketSize\n", + subs->cur_audiofmt->iface, + subs->cur_audiofmt->altset_idx); + return -EINVAL; + } + resp->usb_audio_subslot_size_valid = 1; + } else { + dev_err(&subs->dev->dev, "unknown protocol version %x\n", + protocol); + return -ENODEV; + } + + memcpy(&resp->std_as_opr_intf_desc, &alts->desc, sizeof(alts->desc)); + + return 0; +} + +/** + * prepare_qmi_response() - prepare stream enable response + * @subs: usb substream + * @req_msg: QMI request message + * @resp: QMI response buffer + * @info_idx: usb interface array index + * + * Prepares the QMI response for a USB QMI stream enable request. Will parse + * out the parameters within the stream enable request, in order to match + * requested audio profile to the ones exposed by the USB device connected. + * + * In addition, will fetch the XHCI transfer resources needed for the handoff to + * happen. This includes, transfer ring and buffer addresses and secondary event + * ring address. These parameters will be communicated as part of the USB QMI + * stream enable response. + * + */ +static int prepare_qmi_response(struct snd_usb_substream *subs, + struct qmi_uaudio_stream_req_msg_v01 *req_msg, + struct qmi_uaudio_stream_resp_msg_v01 *resp, + int info_idx) +{ + struct q6usb_offload *data; + int pcm_dev_num; + int card_num; + u8 *xfer_buf = NULL; + int ret; + + pcm_dev_num = (req_msg->usb_token & QMI_STREAM_REQ_DEV_NUM_MASK) >> 8; + card_num = (req_msg->usb_token & QMI_STREAM_REQ_CARD_NUM_MASK) >> 16; + + if (!uadev[card_num].ctrl_intf) { + dev_err(&subs->dev->dev, "audio ctrl intf info not cached\n"); + ret = -ENODEV; + goto err; + } + + ret = uaudio_populate_uac_desc(subs, resp); + if (ret < 0) + goto err; + + resp->slot_id = subs->dev->slot_id; + resp->slot_id_valid = 1; + + data = snd_soc_usb_find_priv_data(uaudio_qdev->auxdev->dev.parent); + if (!data) + goto err; + + uaudio_qdev->data = data; + + resp->std_as_opr_intf_desc_valid = 1; + ret = uaudio_endpoint_setup(subs, subs->data_endpoint, card_num, + &resp->xhci_mem_info.tr_data, + &resp->std_as_data_ep_desc); + if (ret < 0) + goto err; + + resp->std_as_data_ep_desc_valid = 1; + + if (subs->sync_endpoint) { + ret = uaudio_endpoint_setup(subs, subs->sync_endpoint, card_num, + &resp->xhci_mem_info.tr_sync, + &resp->std_as_sync_ep_desc); + if (ret < 0) + goto drop_data_ep; + + resp->std_as_sync_ep_desc_valid = 1; + } + + resp->interrupter_num_valid = 1; + resp->controller_num_valid = 0; + ret = usb_get_controller_id(subs->dev); + if (ret >= 0) { + resp->controller_num = ret; + resp->controller_num_valid = 1; + } + + /* event ring */ + ret = uaudio_event_ring_setup(subs, card_num, + &resp->xhci_mem_info.evt_ring); + if (ret < 0) + goto drop_sync_ep; + + uaudio_qdev->er_mapped = true; + resp->interrupter_num = xhci_sideband_interrupter_id(uadev[card_num].sb); + + resp->speed_info = get_speed_info(subs->dev->speed); + if (resp->speed_info == USB_QMI_DEVICE_SPEED_INVALID_V01) { + ret = -ENODEV; + goto free_sec_ring; + } + + resp->speed_info_valid = 1; + + ret = uaudio_transfer_buffer_setup(subs, xfer_buf, req_msg->xfer_buff_size, + &resp->xhci_mem_info.xfer_buff); + if (ret < 0) { + ret = -ENOMEM; + goto free_sec_ring; + } + + resp->xhci_mem_info_valid = 1; + + if (!atomic_read(&uadev[card_num].in_use)) { + kref_init(&uadev[card_num].kref); + init_waitqueue_head(&uadev[card_num].disconnect_wq); + uadev[card_num].num_intf = + subs->dev->config->desc.bNumInterfaces; + uadev[card_num].info = kcalloc(uadev[card_num].num_intf, + sizeof(struct intf_info), + GFP_KERNEL); + if (!uadev[card_num].info) { + ret = -ENOMEM; + goto unmap_er; + } + uadev[card_num].udev = subs->dev; + atomic_set(&uadev[card_num].in_use, 1); + } else { + kref_get(&uadev[card_num].kref); + } + + uadev[card_num].usb_core_id = resp->controller_num; + + /* cache intf specific info to use it for unmap and free xfer buf */ + uadev[card_num].info[info_idx].data_xfer_ring_va = + IOVA_MASK(resp->xhci_mem_info.tr_data.va); + uadev[card_num].info[info_idx].data_xfer_ring_size = PAGE_SIZE; + uadev[card_num].info[info_idx].sync_xfer_ring_va = + IOVA_MASK(resp->xhci_mem_info.tr_sync.va); + uadev[card_num].info[info_idx].sync_xfer_ring_size = PAGE_SIZE; + uadev[card_num].info[info_idx].xfer_buf_va = + IOVA_MASK(resp->xhci_mem_info.xfer_buff.va); + uadev[card_num].info[info_idx].xfer_buf_pa = + resp->xhci_mem_info.xfer_buff.pa; + uadev[card_num].info[info_idx].xfer_buf_size = + resp->xhci_mem_info.xfer_buff.size; + uadev[card_num].info[info_idx].data_ep_pipe = subs->data_endpoint ? + subs->data_endpoint->pipe : 0; + uadev[card_num].info[info_idx].sync_ep_pipe = subs->sync_endpoint ? + subs->sync_endpoint->pipe : 0; + uadev[card_num].info[info_idx].data_ep_idx = subs->data_endpoint ? + subs->data_endpoint->ep_num : 0; + uadev[card_num].info[info_idx].sync_ep_idx = subs->sync_endpoint ? + subs->sync_endpoint->ep_num : 0; + uadev[card_num].info[info_idx].xfer_buf = xfer_buf; + uadev[card_num].info[info_idx].pcm_card_num = card_num; + uadev[card_num].info[info_idx].pcm_dev_num = pcm_dev_num; + uadev[card_num].info[info_idx].direction = subs->direction; + uadev[card_num].info[info_idx].intf_num = subs->cur_audiofmt->iface; + uadev[card_num].info[info_idx].in_use = true; + + set_bit(card_num, &uaudio_qdev->card_slot); + + return 0; + +unmap_er: + uaudio_iommu_unmap(MEM_EVENT_RING, IOVA_BASE, PAGE_SIZE, PAGE_SIZE); +free_sec_ring: + xhci_sideband_remove_interrupter(uadev[card_num].sb); +drop_sync_ep: + if (subs->sync_endpoint) { + uaudio_iommu_unmap(MEM_XFER_RING, + IOVA_MASK(resp->xhci_mem_info.tr_sync.va), + PAGE_SIZE, PAGE_SIZE); + xhci_sideband_remove_endpoint(uadev[card_num].sb, + usb_pipe_endpoint(subs->dev, subs->sync_endpoint->pipe)); + } +drop_data_ep: + uaudio_iommu_unmap(MEM_XFER_RING, IOVA_MASK(resp->xhci_mem_info.tr_data.va), + PAGE_SIZE, PAGE_SIZE); + xhci_sideband_remove_endpoint(uadev[card_num].sb, + usb_pipe_endpoint(subs->dev, subs->data_endpoint->pipe)); + +err: + return ret; +} + +/** + * handle_uaudio_stream_req() - handle stream enable/disable request + * @handle: QMI client handle + * @sq: qrtr socket + * @txn: QMI transaction context + * @decoded_msg: decoded QMI message + * + * Main handler for the QMI stream enable/disable requests. This executes the + * corresponding enable/disable stream apis, respectively. + * + */ +static void handle_uaudio_stream_req(struct qmi_handle *handle, + struct sockaddr_qrtr *sq, + struct qmi_txn *txn, + const void *decoded_msg) +{ + struct qmi_uaudio_stream_req_msg_v01 *req_msg; + struct qmi_uaudio_stream_resp_msg_v01 resp = {{0}, 0}; + struct uaudio_qmi_svc *svc = uaudio_svc; + struct snd_usb_audio *chip = NULL; + struct snd_usb_substream *subs; + struct usb_host_endpoint *ep; + int datainterval = -EINVAL; + int info_idx = -EINVAL; + struct intf_info *info; + u8 pcm_card_num; + u8 pcm_dev_num; + u8 direction; + int ret = 0; + + if (!svc->client_connected) { + svc->client_sq = *sq; + svc->client_connected = true; + } + + mutex_lock(&qdev_mutex); + req_msg = (struct qmi_uaudio_stream_req_msg_v01 *)decoded_msg; + if (!req_msg->audio_format_valid || !req_msg->bit_rate_valid || + !req_msg->number_of_ch_valid || !req_msg->xfer_buff_size_valid) { + ret = -EINVAL; + goto response; + } + + if (!uaudio_qdev) { + ret = -EINVAL; + goto response; + } + + direction = (req_msg->usb_token & QMI_STREAM_REQ_DIRECTION); + pcm_dev_num = (req_msg->usb_token & QMI_STREAM_REQ_DEV_NUM_MASK) >> 8; + pcm_card_num = (req_msg->usb_token & QMI_STREAM_REQ_CARD_NUM_MASK) >> 16; + if (pcm_card_num >= SNDRV_CARDS) { + ret = -EINVAL; + goto response; + } + + if (req_msg->audio_format > USB_QMI_PCM_FORMAT_U32_BE) { + ret = -EINVAL; + goto response; + } + + subs = find_substream(pcm_card_num, pcm_dev_num, direction); + chip = uadev[pcm_card_num].chip; + if (!subs || !chip || atomic_read(&chip->shutdown)) { + ret = -ENODEV; + goto response; + } + + info_idx = info_idx_from_ifnum(pcm_card_num, subs->cur_audiofmt ? + subs->cur_audiofmt->iface : -1, req_msg->enable); + if (atomic_read(&chip->shutdown) || !subs->stream || !subs->stream->pcm || + !subs->stream->chip) { + ret = -ENODEV; + goto response; + } + + if (req_msg->enable) { + if (info_idx < 0 || chip->system_suspend) { + ret = -EBUSY; + goto response; + } + } + + if (req_msg->service_interval_valid) { + ret = get_data_interval_from_si(subs, + req_msg->service_interval); + if (ret == -EINVAL) + goto response; + + datainterval = ret; + } + + uadev[pcm_card_num].ctrl_intf = chip->ctrl_intf; + + if (req_msg->enable) { + ret = enable_audio_stream(subs, + map_pcm_format(req_msg->audio_format), + req_msg->number_of_ch, req_msg->bit_rate, + datainterval); + + if (!ret) + ret = prepare_qmi_response(subs, req_msg, &resp, + info_idx); + } else { + info = &uadev[pcm_card_num].info[info_idx]; + if (info->data_ep_pipe) { + ep = usb_pipe_endpoint(uadev[pcm_card_num].udev, + info->data_ep_pipe); + if (ep) { + xhci_sideband_stop_endpoint(uadev[pcm_card_num].sb, + ep); + xhci_sideband_remove_endpoint(uadev[pcm_card_num].sb, + ep); + } + + info->data_ep_pipe = 0; + } + + if (info->sync_ep_pipe) { + ep = usb_pipe_endpoint(uadev[pcm_card_num].udev, + info->sync_ep_pipe); + if (ep) { + xhci_sideband_stop_endpoint(uadev[pcm_card_num].sb, + ep); + xhci_sideband_remove_endpoint(uadev[pcm_card_num].sb, + ep); + } + + info->sync_ep_pipe = 0; + } + + disable_audio_stream(subs); + } + +response: + if (!req_msg->enable && ret != -EINVAL && ret != -ENODEV) { + mutex_lock(&chip->mutex); + if (info_idx >= 0) { + info = &uadev[pcm_card_num].info[info_idx]; + uaudio_dev_intf_cleanup(uadev[pcm_card_num].udev, + info); + } + if (atomic_read(&uadev[pcm_card_num].in_use)) + kref_put(&uadev[pcm_card_num].kref, + uaudio_dev_release); + mutex_unlock(&chip->mutex); + } + mutex_unlock(&qdev_mutex); + + resp.usb_token = req_msg->usb_token; + resp.usb_token_valid = 1; + resp.internal_status = ret; + resp.internal_status_valid = 1; + resp.status = ret ? USB_QMI_STREAM_REQ_FAILURE_V01 : ret; + resp.status_valid = 1; + ret = qmi_send_response(svc->uaudio_svc_hdl, sq, txn, + QMI_UAUDIO_STREAM_RESP_V01, + QMI_UAUDIO_STREAM_RESP_MSG_V01_MAX_MSG_LEN, + qmi_uaudio_stream_resp_msg_v01_ei, &resp); +} + +static struct qmi_msg_handler uaudio_stream_req_handlers = { + .type = QMI_REQUEST, + .msg_id = QMI_UAUDIO_STREAM_REQ_V01, + .ei = qmi_uaudio_stream_req_msg_v01_ei, + .decoded_size = QMI_UAUDIO_STREAM_REQ_MSG_V01_MAX_MSG_LEN, + .fn = handle_uaudio_stream_req, +}; + +/** + * qc_usb_audio_offload_init_qmi_dev() - initializes qmi dev + * + * Initializes the USB qdev, which is used to carry information pertaining to + * the offloading resources. This device is freed only when there are no longer + * any offloading candidates. (i.e, when all audio devices are disconnected) + * + */ +static int qc_usb_audio_offload_init_qmi_dev(void) +{ + uaudio_qdev = kzalloc(sizeof(*uaudio_qdev), GFP_KERNEL); + if (!uaudio_qdev) + return -ENOMEM; + + /* initialize xfer ring and xfer buf iova list */ + INIT_LIST_HEAD(&uaudio_qdev->xfer_ring_list); + uaudio_qdev->curr_xfer_ring_iova = IOVA_XFER_RING_BASE; + uaudio_qdev->xfer_ring_iova_size = + IOVA_XFER_RING_MAX - IOVA_XFER_RING_BASE; + + INIT_LIST_HEAD(&uaudio_qdev->xfer_buf_list); + uaudio_qdev->curr_xfer_buf_iova = IOVA_XFER_BUF_BASE; + uaudio_qdev->xfer_buf_iova_size = + IOVA_XFER_BUF_MAX - IOVA_XFER_BUF_BASE; + + return 0; +} + +/* Populates ppcm_idx array with supported PCM indexes */ +static int qc_usb_audio_offload_fill_avail_pcms(struct snd_usb_audio *chip, + struct snd_soc_usb_device *sdev) +{ + struct snd_usb_stream *as; + struct snd_usb_substream *subs; + int idx = 0; + + list_for_each_entry(as, &chip->pcm_list, list) { + subs = &as->substream[SNDRV_PCM_STREAM_PLAYBACK]; + if (subs->ep_num) { + sdev->ppcm_idx[idx] = as->pcm->device; + idx++; + } + /* + * Break if the current index exceeds the number of possible + * playback streams counted from the UAC descriptors. + */ + if (idx >= sdev->num_playback) + break; + } + + return -1; +} + +/** + * qc_usb_audio_offload_probe() - platform op connect handler + * @chip: USB SND device + * + * Platform connect handler when a USB SND device is detected. Will + * notify SOC USB about the connection to enable the USB ASoC backend + * and populate internal USB chip array. + * + */ +static void qc_usb_audio_offload_probe(struct snd_usb_audio *chip) +{ + struct usb_interface *intf = chip->intf[chip->num_interfaces - 1]; + struct usb_interface_descriptor *altsd; + struct usb_host_interface *alts; + struct snd_soc_usb_device *sdev; + struct xhci_sideband *sb; + + /* + * If there is no priv_data, or no playback paths, the connected + * device doesn't support offloading. Avoid populating entries for + * this device. + */ + if (!snd_soc_usb_find_priv_data(uaudio_qdev->auxdev->dev.parent) || + !usb_qmi_get_pcm_num(chip, 0)) + return; + + mutex_lock(&qdev_mutex); + mutex_lock(&chip->mutex); + if (!uadev[chip->card->number].chip) { + sdev = kzalloc(sizeof(*sdev), GFP_KERNEL); + if (!sdev) + goto exit; + + sb = xhci_sideband_register(intf, XHCI_SIDEBAND_VENDOR, + uaudio_sideband_notifier); + if (!sb) + goto free_sdev; + } else { + sb = uadev[chip->card->number].sb; + sdev = uadev[chip->card->number].sdev; + } + + uadev[chip->card->number].sb = sb; + uadev[chip->card->number].chip = chip; + uadev[chip->card->number].sdev = sdev; + + alts = &intf->altsetting[0]; + altsd = get_iface_desc(alts); + + /* Wait until all PCM devices are populated before notifying soc-usb */ + if (altsd->bInterfaceNumber == chip->last_iface) { + sdev->num_playback = usb_qmi_get_pcm_num(chip, 0); + + /* + * Allocate playback pcm index array based on number of possible + * playback paths within the UAC descriptors. + */ + sdev->ppcm_idx = kcalloc(sdev->num_playback, sizeof(unsigned int), + GFP_KERNEL); + if (!sdev->ppcm_idx) + goto unreg_xhci; + + qc_usb_audio_offload_fill_avail_pcms(chip, sdev); + sdev->card_idx = chip->card->number; + sdev->chip_idx = chip->index; + + snd_soc_usb_connect(uaudio_qdev->auxdev->dev.parent, sdev); + } + + mutex_unlock(&chip->mutex); + mutex_unlock(&qdev_mutex); + + return; + +unreg_xhci: + xhci_sideband_unregister(sb); + uadev[chip->card->number].sb = NULL; +free_sdev: + kfree(sdev); + uadev[chip->card->number].sdev = NULL; + uadev[chip->card->number].chip = NULL; +exit: + mutex_unlock(&chip->mutex); + mutex_unlock(&qdev_mutex); +} + +/** + * qc_usb_audio_cleanup_qmi_dev() - release qmi device + * + * Frees the USB qdev. Only occurs when there are no longer any potential + * devices that can utilize USB audio offloading. + * + */ +static void qc_usb_audio_cleanup_qmi_dev(void) +{ + kfree(uaudio_qdev); + uaudio_qdev = NULL; +} + +/** + * qc_usb_audio_offload_disconnect() - platform op disconnect handler + * @chip: USB SND device + * + * Platform disconnect handler. Will ensure that any pending stream is + * halted by issuing a QMI disconnect indication packet to the adsp. + * + */ +static void qc_usb_audio_offload_disconnect(struct snd_usb_audio *chip) +{ + struct uaudio_dev *dev; + int card_num; + + if (!chip) + return; + + card_num = chip->card->number; + if (card_num >= SNDRV_CARDS) + return; + + mutex_lock(&qdev_mutex); + mutex_lock(&chip->mutex); + dev = &uadev[card_num]; + + /* Device has already been cleaned up, or never populated */ + if (!dev->chip) { + mutex_unlock(&qdev_mutex); + mutex_unlock(&chip->mutex); + return; + } + + /* cleaned up already */ + if (!dev->udev) + goto done; + + uaudio_send_disconnect_ind(chip); + uaudio_dev_cleanup(dev); +done: + /* + * If num_interfaces == 1, the last USB SND interface is being removed. + * This is to accommodate for devices w/ multiple UAC functions. + */ + if (chip->num_interfaces == 1) { + snd_soc_usb_disconnect(uaudio_qdev->auxdev->dev.parent, dev->sdev); + xhci_sideband_unregister(dev->sb); + dev->chip = NULL; + kfree(dev->sdev->ppcm_idx); + kfree(dev->sdev); + dev->sdev = NULL; + } + mutex_unlock(&chip->mutex); + + mutex_unlock(&qdev_mutex); +} + +/** + * qc_usb_audio_offload_suspend() - USB offload PM suspend handler + * @intf: USB interface + * @message: suspend type + * + * PM suspend handler to ensure that the USB offloading driver is able to stop + * any pending traffic, so that the bus can be suspended. + * + */ +static void qc_usb_audio_offload_suspend(struct usb_interface *intf, + pm_message_t message) +{ + struct snd_usb_audio *chip = usb_get_intfdata(intf); + int card_num; + + if (!chip) + return; + + card_num = chip->card->number; + if (card_num >= SNDRV_CARDS) + return; + + mutex_lock(&qdev_mutex); + mutex_lock(&chip->mutex); + + uaudio_send_disconnect_ind(chip); + + mutex_unlock(&qdev_mutex); + mutex_unlock(&chip->mutex); +} + +static struct snd_usb_platform_ops offload_ops = { + .connect_cb = qc_usb_audio_offload_probe, + .disconnect_cb = qc_usb_audio_offload_disconnect, + .suspend_cb = qc_usb_audio_offload_suspend, +}; + +static int qc_usb_audio_probe(struct auxiliary_device *auxdev, + const struct auxiliary_device_id *id) + +{ + struct uaudio_qmi_svc *svc; + int ret; + + svc = kzalloc(sizeof(*svc), GFP_KERNEL); + if (!svc) + return -ENOMEM; + + svc->uaudio_svc_hdl = kzalloc(sizeof(*svc->uaudio_svc_hdl), GFP_KERNEL); + if (!svc->uaudio_svc_hdl) { + ret = -ENOMEM; + goto free_svc; + } + + ret = qmi_handle_init(svc->uaudio_svc_hdl, + QMI_UAUDIO_STREAM_REQ_MSG_V01_MAX_MSG_LEN, + &uaudio_svc_ops_options, + &uaudio_stream_req_handlers); + ret = qmi_add_server(svc->uaudio_svc_hdl, UAUDIO_STREAM_SERVICE_ID_V01, + UAUDIO_STREAM_SERVICE_VERS_V01, 0); + + uaudio_svc = svc; + + qc_usb_audio_offload_init_qmi_dev(); + uaudio_qdev->auxdev = auxdev; + + ret = snd_usb_register_platform_ops(&offload_ops); + if (ret < 0) + goto release_qmi; + + return 0; + +release_qmi: + qmi_handle_release(svc->uaudio_svc_hdl); +free_svc: + kfree(svc); + + return ret; +} + +static void qc_usb_audio_remove(struct auxiliary_device *auxdev) +{ + struct uaudio_qmi_svc *svc = uaudio_svc; + int idx; + + /* + * Remove all connected devices after unregistering ops, to ensure + * that no further connect events will occur. The disconnect routine + * will issue the QMI disconnect indication, which results in the + * external DSP to stop issuing transfers. + */ + snd_usb_unregister_platform_ops(); + for (idx = 0; idx < SNDRV_CARDS; idx++) + qc_usb_audio_offload_disconnect(uadev[idx].chip); + + qc_usb_audio_cleanup_qmi_dev(); + + qmi_handle_release(svc->uaudio_svc_hdl); + kfree(svc); + uaudio_svc = NULL; +} + +static const struct auxiliary_device_id qc_usb_audio_table[] = { + { .name = "q6usb.qc-usb-audio-offload" }, + {}, +}; +MODULE_DEVICE_TABLE(auxiliary, qc_usb_audio_table); + +static struct auxiliary_driver qc_usb_audio_offload_drv = { + .name = "qc-usb-audio-offload", + .id_table = qc_usb_audio_table, + .probe = qc_usb_audio_probe, + .remove = qc_usb_audio_remove, +}; +module_auxiliary_driver(qc_usb_audio_offload_drv); + +MODULE_DESCRIPTION("QC USB Audio Offloading"); +MODULE_LICENSE("GPL");