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

[PATCH v5 00/11] Add ASoC AHUB components for Tegra210 and later
by Sameer Pujar 07 Aug '20

07 Aug '20

Overview ======== Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. * Add YAML DT binding documentation for above mentioned modules. * Helper function for ACIF programming is exposed for Tegra210 and later. * Add ASoC driver components for each of the above modules. * Build ACONNECT and ADMA drivers which are essential to realize audio use case. * Add DT entries for above components for Tegra210, Tegra186 and Tegra194. As per the suggestion in [0] audio graph based sound card support is pushed in a separate series. [0] https://lkml.org/lkml/2020/6/27/4 Changelog ========= v4 -> v5 -------- * Common changes - simple-card driver changes are dropped. Changes are migrated to audio graph card and are moved to a separate series as suggested. - '#sound-dai-cells' property is not needed for planned audio graph card Hence dropped from documentation and related DT binding of component drivers. - CIF and DAP DAIs are added for I/O drivers (DMIC, DSPK, I2S) to represent DAI links using audio graph card. Similary DAIs are added in AHUB driver to describe endpoints in audio crossbar. Routing is updated to reflect the same in drivers. v3 -> v4 -------- * [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210" - Removed multiple examples and retained one example per doc - Fixed as per inputs on the previous series - Tested bindings with 'make dt_binding_check/dtbs_check' * [2/23] "ASoC: tegra: Add support for CIF programming" - No change * Common changes (for patch [3/10] to [7/10]) - Mixer control overrides, for PCM parameters (rate, channel, bits), in each driver are dropped. - Updated routing as per DPCM usage - Minor changes related to formatting * New changes (patch [8/23] to [18/23] and patch [23/23]) - Based on discussions in following threads DPCM is used for Tegra Audio. https://lkml.org/lkml/2020/2/20/91 https://lkml.org/lkml/2020/4/30/519 - The simple-card driver is used for Tegra Audio and accordingly some enhancements are made in simple-card and core drivers. - Patch [8/23] to [18/23] are related to simple-card and core changes. - Patch [23/23] adds sound card support to realize complete audio path. This is based on simple-card driver with proposed enhancements. - Re-ordered patches depending on above v2 -> v3 -------- * [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB - Updated licence - Removed redundancy w.r.t items/const/enum - Added constraints wherever needed with "pattern" property * [2/10] "ASoC: tegra: add support for CIF programming" - Removed tegra_cif.c - Instead added inline helper function in tegra_cif.h * common changes (for patch [3/10] to [7/10]) - Replace LATE system calls with Normal sleep - Remove explicit RPM suspend in driver remove() call - Use devm_kzalloc() instead of devm_kcalloc() for single element - Replace 'ret' with 'err' for better reading - Consistent error printing style across drivers - Minor formating fixes * [8/10] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later" (New patch) - Enables ACONNECT and AHUB components. With this AHUB and components are registered with ASoC core. v1 -> v2 -------- * [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB" - no changes * [2/9] "ASoC: tegra: add support for CIF programming" - removed CIF programming changes for legacy chips. - this patch now exposes helper function for CIF programming, which can be used on Tegra210 later. - later tegra_cif.c can be extended for legacy chips as well. - updated commit message accordingly * [3/9] "ASoC: tegra: add Tegra210 based DMIC driver" - removed unnecessary initialization of 'ret' in probe() * [4/9] "ASoC: tegra: add Tegra210 based I2S driver" - removed unnecessary initialization of 'ret' in probe() - fixed indentation - added consistent bracing for if-else clauses - updated 'rx_fifo_th' type to 'unsigned int' - used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h * [5/9] "ASoC: tegra: add Tegra210 based AHUB driver" - used of_device_get_match_data() to get 'soc_data' and removed explicit of_match_device() - used devm_platform_ioremap_resource() and removed explicit platform_get_resource() - fixed indentation for devm_snd_soc_register_component() - updated commit message - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [6/9] "ASoC: tegra: add Tegra186 based DSPK driver" - removed unnecessary initialization of 'ret' in probe() - updated 'max_th' to 'unsigned int' - shortened lengthy macro names to avoid wrapping in tegra186_dspk_wr_reg() and to be consistent * [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver" - used of_device_get_match_data() and removed explicit of_match_device() - used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [8/9] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * common changes for patch [3/9] to [7/9] - sorted headers in alphabetical order - moved MODULE_DEVICE_TABLE() right below *_of_match table - removed macro DRV_NAME - removed explicit 'owner' field from platform_driver structure - added 'const' to snd_soc_dai_ops structure Sameer Pujar (11): ASoC: dt-bindings: tegra: Add DT bindings for Tegra210 ASoC: tegra: Add support for CIF programming ASoC: tegra: Add Tegra210 based DMIC driver ASoC: tegra: Add Tegra210 based I2S driver ASoC: tegra: Add Tegra210 based AHUB driver ASoC: tegra: Add Tegra186 based DSPK driver ASoC: tegra: Add Tegra210 based ADMAIF driver arm64: defconfig: Build AHUB component drivers arm64: defconfig: Build ADMA and ACONNECT driver arm64: tegra: Enable ACONNECT, ADMA and AGIC on Jetson Nano arm64: tegra: Add DT binding for AHUB components .../bindings/sound/nvidia,tegra186-dspk.yaml | 83 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 111 +++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 136 ++++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 83 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 217 +++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 225 +++++- arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 12 + arch/arm64/boot/dts/nvidia/tegra210.dtsi | 140 ++++ arch/arm64/configs/defconfig | 8 + sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra186_dspk.c | 442 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 70 ++ sound/soc/tegra/tegra210_admaif.c | 800 ++++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 162 ++++ sound/soc/tegra/tegra210_ahub.c | 676 +++++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 127 ++++ sound/soc/tegra/tegra210_dmic.c | 455 ++++++++++++ sound/soc/tegra/tegra210_dmic.h | 82 +++ sound/soc/tegra/tegra210_i2s.c | 812 +++++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 126 ++++ sound/soc/tegra/tegra_cif.h | 65 ++ sound/soc/tegra/tegra_pcm.c | 235 +++++- sound/soc/tegra/tegra_pcm.h | 21 +- 25 files changed, 5251 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

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[RFC PATCH v2 0/6] ALSA: compress: add support to change codec profile in gapless playback
by Srinivas Kandagatla 06 Aug '20

06 Aug '20

For gapless playback it is possible that each track can have different codec profile with same decoder, for example we have WMA album, we may have different tracks as WMA v9, WMA v10 and so on Or if DSP's like QDSP have abililty to switch decoders on single stream for each track, then this call could be used to set new codec parameters. Existing code does not allow to change this profile while doing gapless playback. This patchset adds new SNDRV_COMPRESS_SET_CODEC_PARAMS IOCTL along with flags in capablity structure to allow userspace to set this new parameters required which switching codec profile, either for gapless or cross fade usecase. thanks, srini Changes since v1: - split patch into smaller chuncks, - bump up the version - added flags in compress capablity structure - added user for this new functionality. - add this new call in Documentation. Srinivas Kandagatla (6): ALSA: compress: move codec parameter check to a function ALSA: compress: add new ioctl for setting codec parameters ALSA: compress: add flags to snd_compr_caps to expose dsp caps ASoC: compress: add snd_soc_dai_compr_set_codec_params() ALSA: compress: bump the version ASoC: q6asm-dai: add support to set_codec_params .../sound/designs/compress-offload.rst | 6 ++ include/sound/compress_driver.h | 5 ++ include/sound/soc-component.h | 3 + include/sound/soc-dai.h | 5 ++ include/uapi/sound/compress_offload.h | 10 ++- sound/core/compress_offload.c | 72 +++++++++++++++++-- sound/soc/qcom/qdsp6/q6asm-dai.c | 33 +++++++++ sound/soc/soc-compress.c | 30 ++++++++ sound/soc/soc-dai.c | 14 ++++ 9 files changed, 170 insertions(+), 8 deletions(-) -- 2.21.0

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[PATCH] soc: qmi: allow user to set handle wq to hiprio
by 王文虎 05 Aug '20

05 Aug '20

Currently the qmi_handle is initialized single threaded and strictly ordered with the active set to 1. This is pretty simple and safe but sometimes ineffency. So it is better to allow user to decide whether a high priority workqueue should be used. Signed-off-by: Wang Wenhu <wenhu.wang(a)vivo.com> --- drivers/net/ipa/ipa_qmi.c | 4 ++-- drivers/net/wireless/ath/ath10k/qmi.c | 2 +- drivers/net/wireless/ath/ath11k/qmi.c | 2 +- drivers/remoteproc/qcom_sysmon.c | 2 +- drivers/slimbus/qcom-ngd-ctrl.c | 4 ++-- drivers/soc/qcom/pdr_interface.c | 4 ++-- drivers/soc/qcom/qmi_interface.c | 9 +++++++-- include/linux/soc/qcom/qmi.h | 3 ++- samples/qmi/qmi_sample_client.c | 4 ++-- 9 files changed, 20 insertions(+), 14 deletions(-) diff --git a/drivers/net/ipa/ipa_qmi.c b/drivers/net/ipa/ipa_qmi.c index 5090f0f923ad..d78b0fe6bd83 100644 --- a/drivers/net/ipa/ipa_qmi.c +++ b/drivers/net/ipa/ipa_qmi.c @@ -486,7 +486,7 @@ int ipa_qmi_setup(struct ipa *ipa) */ ret = qmi_handle_init(&ipa_qmi->server_handle, IPA_QMI_SERVER_MAX_RCV_SZ, &ipa_server_ops, - ipa_server_msg_handlers); + ipa_server_msg_handlers, 0); if (ret) return ret; @@ -500,7 +500,7 @@ int ipa_qmi_setup(struct ipa *ipa) */ ret = qmi_handle_init(&ipa_qmi->client_handle, IPA_QMI_CLIENT_MAX_RCV_SZ, &ipa_client_ops, - ipa_client_msg_handlers); + ipa_client_msg_handlers, 0); if (ret) goto err_server_handle_release; diff --git a/drivers/net/wireless/ath/ath10k/qmi.c b/drivers/net/wireless/ath/ath10k/qmi.c index 5468a41e928e..02881882b4d9 100644 --- a/drivers/net/wireless/ath/ath10k/qmi.c +++ b/drivers/net/wireless/ath/ath10k/qmi.c @@ -1034,7 +1034,7 @@ int ath10k_qmi_init(struct ath10k *ar, u32 msa_size) ret = qmi_handle_init(&qmi->qmi_hdl, WLFW_BDF_DOWNLOAD_REQ_MSG_V01_MAX_MSG_LEN, - &ath10k_qmi_ops, qmi_msg_handler); + &ath10k_qmi_ops, qmi_msg_handler, 0); if (ret) goto err; diff --git a/drivers/net/wireless/ath/ath11k/qmi.c b/drivers/net/wireless/ath/ath11k/qmi.c index c00a99ad8dbc..91394d58d36e 100644 --- a/drivers/net/wireless/ath/ath11k/qmi.c +++ b/drivers/net/wireless/ath/ath11k/qmi.c @@ -2397,7 +2397,7 @@ int ath11k_qmi_init_service(struct ath11k_base *ab) ab->qmi.target_mem_mode = ATH11K_QMI_TARGET_MEM_MODE_DEFAULT; ret = qmi_handle_init(&ab->qmi.handle, ATH11K_QMI_RESP_LEN_MAX, - &ath11k_qmi_ops, ath11k_qmi_msg_handlers); + &ath11k_qmi_ops, ath11k_qmi_msg_handlers, 0); if (ret < 0) { ath11k_warn(ab, "failed to initialize qmi handle\n"); return ret; diff --git a/drivers/remoteproc/qcom_sysmon.c b/drivers/remoteproc/qcom_sysmon.c index 8d8996d714f0..4ec470e424ef 100644 --- a/drivers/remoteproc/qcom_sysmon.c +++ b/drivers/remoteproc/qcom_sysmon.c @@ -614,7 +614,7 @@ struct qcom_sysmon *qcom_add_sysmon_subdev(struct rproc *rproc, } ret = qmi_handle_init(&sysmon->qmi, SSCTL_MAX_MSG_LEN, &ssctl_ops, - qmi_indication_handler); + qmi_indication_handler, 0); if (ret < 0) { dev_err(sysmon->dev, "failed to initialize qmi handle\n"); kfree(sysmon); diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 743ee7b4e63f..ba76691fc5a5 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -446,7 +446,7 @@ static int qcom_slim_qmi_init(struct qcom_slim_ngd_ctrl *ctrl, return -ENOMEM; rc = qmi_handle_init(handle, SLIMBUS_QMI_POWER_REQ_MAX_MSG_LEN, - NULL, qcom_slim_qmi_msg_handlers); + NULL, qcom_slim_qmi_msg_handlers, 0); if (rc < 0) { dev_err(ctrl->dev, "QMI client init failed: %d\n", rc); goto qmi_handle_init_failed; @@ -1293,7 +1293,7 @@ static int qcom_slim_ngd_qmi_svc_event_init(struct qcom_slim_ngd_ctrl *ctrl) int ret; ret = qmi_handle_init(&qmi->svc_event_hdl, 0, - &qcom_slim_ngd_qmi_svc_event_ops, NULL); + &qcom_slim_ngd_qmi_svc_event_ops, NULL, 0); if (ret < 0) { dev_err(ctrl->dev, "qmi_handle_init failed: %d\n", ret); return ret; diff --git a/drivers/soc/qcom/pdr_interface.c b/drivers/soc/qcom/pdr_interface.c index bdcf16f88a97..cc1cb90c1968 100644 --- a/drivers/soc/qcom/pdr_interface.c +++ b/drivers/soc/qcom/pdr_interface.c @@ -685,7 +685,7 @@ struct pdr_handle *pdr_handle_alloc(void (*status)(int state, ret = qmi_handle_init(&pdr->locator_hdl, SERVREG_GET_DOMAIN_LIST_RESP_MAX_LEN, - &pdr_locator_ops, NULL); + &pdr_locator_ops, NULL, 0); if (ret < 0) goto destroy_indack; @@ -696,7 +696,7 @@ struct pdr_handle *pdr_handle_alloc(void (*status)(int state, ret = qmi_handle_init(&pdr->notifier_hdl, SERVREG_STATE_UPDATED_IND_MAX_LEN, &pdr_notifier_ops, - qmi_indication_handler); + qmi_indication_handler, 0); if (ret < 0) goto release_qmi_handle; diff --git a/drivers/soc/qcom/qmi_interface.c b/drivers/soc/qcom/qmi_interface.c index 1a03eaa38c46..01160dbfc4d0 100644 --- a/drivers/soc/qcom/qmi_interface.c +++ b/drivers/soc/qcom/qmi_interface.c @@ -609,6 +609,7 @@ static struct socket *qmi_sock_create(struct qmi_handle *qmi, * @recv_buf_size: maximum size of incoming message * @ops: reference to callbacks for QRTR notifications * @handlers: NULL-terminated list of QMI message handlers + * @hiprio: whether high priority worker is used for workqueue * * This initializes the QMI client handle to allow sending and receiving QMI * messages. As messages are received the appropriate handler will be invoked. @@ -617,9 +618,11 @@ static struct socket *qmi_sock_create(struct qmi_handle *qmi, */ int qmi_handle_init(struct qmi_handle *qmi, size_t recv_buf_size, const struct qmi_ops *ops, - const struct qmi_msg_handler *handlers) + const struct qmi_msg_handler *handlers, + unsigned int hiprio) { int ret; + unsigned int flags = WQ_UNBOUND; mutex_init(&qmi->txn_lock); mutex_init(&qmi->sock_lock); @@ -647,7 +650,9 @@ int qmi_handle_init(struct qmi_handle *qmi, size_t recv_buf_size, if (!qmi->recv_buf) return -ENOMEM; - qmi->wq = alloc_workqueue("qmi_msg_handler", WQ_UNBOUND, 1); + if (hiprio) + flags |= WQ_HIGHPRI; + qmi->wq = alloc_workqueue("qmi_msg_handler", flags, 1); if (!qmi->wq) { ret = -ENOMEM; goto err_free_recv_buf; diff --git a/include/linux/soc/qcom/qmi.h b/include/linux/soc/qcom/qmi.h index e712f94b89fc..24062fd7163d 100644 --- a/include/linux/soc/qcom/qmi.h +++ b/include/linux/soc/qcom/qmi.h @@ -244,7 +244,8 @@ int qmi_add_server(struct qmi_handle *qmi, unsigned int service, int qmi_handle_init(struct qmi_handle *qmi, size_t max_msg_len, const struct qmi_ops *ops, - const struct qmi_msg_handler *handlers); + const struct qmi_msg_handler *handlers, + unsigned int hiprio); void qmi_handle_release(struct qmi_handle *qmi); ssize_t qmi_send_request(struct qmi_handle *qmi, struct sockaddr_qrtr *sq, diff --git a/samples/qmi/qmi_sample_client.c b/samples/qmi/qmi_sample_client.c index c9e7276c3d83..a91d1633ea38 100644 --- a/samples/qmi/qmi_sample_client.c +++ b/samples/qmi/qmi_sample_client.c @@ -463,7 +463,7 @@ static int qmi_sample_probe(struct platform_device *pdev) ret = qmi_handle_init(&sample->qmi, TEST_DATA_REQ_MAX_MSG_LEN_V01, NULL, - qmi_sample_handlers); + qmi_sample_handlers, 0); if (ret < 0) return ret; @@ -590,7 +590,7 @@ static int qmi_sample_init(void) if (ret) goto err_remove_debug_dir; - ret = qmi_handle_init(&lookup_client, 0, &lookup_ops, NULL); + ret = qmi_handle_init(&lookup_client, 0, &lookup_ops, NULL, 0); if (ret < 0) goto err_unregister_driver; -- 2.17.1

3 3

[PATCH 0/3] ASoC: codec: tlv3204: Codec workaround
by Michael Sit Wei Hong 04 Aug '20

04 Aug '20

This patch series enables some features on the tlv3204 codec and also fixes some issues faced while testing v1: initial ASoC: codec: tlv3204: Codec workaround series Michael Sit Wei Hong (3): ASoC: codec: tlv3204: Enable 24 bit audio support ASoC: codec: tlv3204: Increased maximum supported channels ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset sound/soc/codecs/tlv320aic32x4.c | 60 +++++++++++++++++++++++--------- 1 file changed, 44 insertions(+), 16 deletions(-) -- 2.17.1

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[PATCH] ASoC: amd: renoir: restore two more registers during resume
by Hui Wang 03 Aug '20

03 Aug '20

Recently we found an issue about the suspend and resume. If dmic is recording the sound, and we run suspend and resume, after the resume, the dmic can't work well anymore. we need to close the app and reopen the app, then the dmic could record the sound again. For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2 -r 48000 test.wav", then suspend and resume, after the system resume back, we speak to the dmic. then stop the arecord, use aplay to play the test.wav, we could hear the sound recorded after resume is weird, it is not what we speak to the dmic. I found two registers are set in the dai_hw_params(), if the two registers are set in the resume() too, this issue could be fixed. Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 623dfd3ea705..8acb0315a169 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -474,6 +474,11 @@ static int acp_pdm_resume(struct device *dev) rtd = runtime->private_data; period_bytes = frames_to_bytes(runtime, runtime->period_size); buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + if (runtime->channels == TWO_CH) { + rn_writel(0x0 , rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); + rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + + ACP_WOV_PDM_DECIMATION_FACTOR); + } config_acp_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); init_pdm_ring_buffer(MEM_WINDOW_START, buffer_len, period_bytes, adata->acp_base); -- 2.17.1

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[PATCH v2] ASoC: amd: renoir: restore two more registers during resume
by Hui Wang 03 Aug '20

03 Aug '20

Recently we found an issue about the suspend and resume. If dmic is recording the sound, and we run suspend and resume, after the resume, the dmic can't work well anymore. we need to close the app and reopen the app, then the dmic could record the sound again. For example, we run "arecord -D hw:CARD=acp,DEV=0 -f S32_LE -c 2 -r 48000 test.wav", then suspend and resume, after the system resume back, we speak to the dmic. then stop the arecord, use aplay to play the test.wav, we could hear the sound recorded after resume is weird, it is not what we speak to the dmic. I found two registers are set in the dai_hw_params(), if the two registers are set during the resume, this issue could be fixed. Move the code of the dai_hw_params() into the pdm_dai_trigger(), then these two registers will be set during resume since pdm_dai_trigger() will be called during resume. And delete the empty function dai_hw_params(). Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/amd/renoir/acp3x-pdm-dma.c | 29 +++++++++------------------- 1 file changed, 9 insertions(+), 20 deletions(-) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 623dfd3ea705..7b14d9a81b97 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -314,40 +314,30 @@ static int acp_pdm_dma_close(struct snd_soc_component *component, return 0; } -static int acp_pdm_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) { struct pdm_stream_instance *rtd; + int ret; + bool pdm_status; unsigned int ch_mask; rtd = substream->runtime->private_data; - switch (params_channels(params)) { + ret = 0; + switch (substream->runtime->channels) { case TWO_CH: ch_mask = 0x00; break; default: return -EINVAL; } - rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); - rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + - ACP_WOV_PDM_DECIMATION_FACTOR); - return 0; -} - -static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct pdm_stream_instance *rtd; - int ret; - bool pdm_status; - - rtd = substream->runtime->private_data; - ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); + rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + + ACP_WOV_PDM_DECIMATION_FACTOR); rtd->bytescount = acp_pdm_get_byte_count(rtd, substream->stream); pdm_status = check_pdm_dma_status(rtd->acp_base); @@ -369,7 +359,6 @@ static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, } static struct snd_soc_dai_ops acp_pdm_dai_ops = { - .hw_params = acp_pdm_dai_hw_params, .trigger = acp_pdm_dai_trigger, }; -- 2.17.1

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[PATCH 00/10] Audio graph card updates and usage with Tegra210 audio
by Sameer Pujar 03 Aug '20

03 Aug '20

This series proposes following enhancements to audio-graph card driver. * Support multiple instances of a component. * Support Codec port with multiple endpoints. * Support open platforms with empty Codec endpoint. * Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Add new compatible to support DPCM based DAI chaining. This pushes DT support for Tegra210 based platforms which uses audio-graph card and above enhancements. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Sameer Pujar (10): ASoC: soc-core: Fix component name_prefix parsing ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Support Codec with multiple endpoints ASoC: audio-graph: Support for empty end point ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: soc-pcm: Get all BEs along DAPM path ASoC: dt-bindings: audio-graph-card: Support for component chaining ASoC: audio-graph: Add support for component chaining arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../devicetree/bindings/sound/audio-graph-card.txt | 1 + .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 120 +++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 196 +++++++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 110 ++++++++++++ include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 85 ++++++--- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- 8 files changed, 495 insertions(+), 24 deletions(-) create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi -- 2.7.4

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[PATCH] ASoC: wm8962: Do not access WM8962_GPIO_BASE
by Fabio Estevam 03 Aug '20

03 Aug '20

According to the WM8962 datasheet, there is no register at address 0x200. WM8962_GPIO_BASE is just a base address for the GPIO registers and not a real register, so remove it from wm8962_readable_register(). Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip its access. This fixes the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Hi, There is still one more soc_component_read_no_lock error left on register 48. I can get rid of it with the below change: --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -151,6 +151,7 @@ static const struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ + { 48, 0x0000 }, /* R48 - Additional control(4) */ { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ @@ -841,7 +842,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_SPKOUTL_VOLUME: case WM8962_SPKOUTR_VOLUME: case WM8962_THERMAL_SHUTDOWN_STATUS: - case WM8962_ADDITIONAL_CONTROL_4: case WM8962_CLASS_D_CONTROL_1: case WM8962_CLASS_D_CONTROL_2: case WM8962_CLOCKING_4: I haven't submitted it yet because I don't know if this is the correct approach. Please advise. Thanks sound/soc/codecs/wm8962.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index df8cdc71357d..8159a3866cde 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -956,7 +956,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_EQ39: case WM8962_EQ40: case WM8962_EQ41: - case WM8962_GPIO_BASE: case WM8962_GPIO_2: case WM8962_GPIO_3: case WM8962_GPIO_5: @@ -3437,7 +3436,13 @@ static int wm8962_probe(struct snd_soc_component *component) /* Save boards having to disable DMIC when not in use */ dmicclk = false; dmicdat = false; - for (i = 0; i < WM8962_MAX_GPIO; i++) { + for (i = 1; i < WM8962_MAX_GPIO; i++) { + /* + * Register 515 (WM8962_GPIO_BASE + 3) does not exist, + * so skip its access + */ + if (i == 3) + continue; switch (snd_soc_component_read(component, WM8962_GPIO_BASE + i) & WM8962_GP2_FN_MASK) { case WM8962_GPIO_FN_DMICCLK: -- 2.17.1

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[PATCH v3 0/2] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 03 Aug '20

03 Aug '20

Note: - The machine driver patch depends on LPASS patch series so it is not ready to be merged now. ASoC: qcom: Add support for SC7180 lpass variant https://patchwork.kernel.org/cover/11678133/ - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Thanks for the review! Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (1): ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 113 ++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 244 ++++++++++++++++++ 4 files changed, 371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.163.g6104cc2f0b6-goog

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[PATCH v2] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
by Shengjiu Wang 03 Aug '20

03 Aug '20

With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - replace is_stream_in_use with streams - add "out" error handler in hw_params() sound/soc/fsl/fsl-asoc-card.c | 155 ++++++++++++++++------------------ 1 file changed, 71 insertions(+), 84 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4848ba61d083..b6a2a527dc04 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @streams: Mask of current active streams * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->streams |= BIT(substream->stream); - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); - return ret; + goto out; } if (cpu_priv->slot_width) { @@ -182,6 +180,69 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); + goto out; + } + } + + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto out; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + goto out; + } + } + + return 0; + +out: + priv->streams &= ~BIT(substream->stream); + return ret; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->streams &= ~BIT(substream->stream); + + if (!priv->streams && codec_priv->pll_id && + codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); return ret; } } @@ -191,6 +252,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +316,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); - if (ret) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -611,7 +604,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -628,26 +620,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -657,7 +645,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; -- 2.27.0

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