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[PATCH] ASoC: SOF: prefix some terse and cryptic dev_dbg() with __func__
by Peter Ujfalusi 28 Sep '21

28 Sep '21

From: Marc Herbert <marc.herbert(a)intel.com> These helped troubleshoot some DMA issue in SOF. Signed-off-by: Marc Herbert <marc.herbert(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 7 ++++--- sound/soc/sof/trace.c | 5 +++-- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..ab051f67e0e8 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -89,13 +89,13 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, int remain, ioc; period_bytes = stream->period_bytes; - dev_dbg(sdev->dev, "period_bytes:0x%x\n", period_bytes); + dev_dbg(sdev->dev, "%s: period_bytes:0x%x\n", __func__, period_bytes); if (!period_bytes) period_bytes = stream->bufsize; periods = stream->bufsize / period_bytes; - dev_dbg(sdev->dev, "periods:%d\n", periods); + dev_dbg(sdev->dev, "%s: periods:%d\n", __func__, periods); remain = stream->bufsize % period_bytes; if (remain) @@ -244,7 +244,8 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) HDA_VS_INTEL_EM2_L1SEN, HDA_VS_INTEL_EM2_L1SEN); if (!found) { - dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag); + dev_dbg(sdev->dev, "%s: stream_tag %d not opened!\n", + __func__, stream_tag); return -ENODEV; } diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index 58f6ca5cf491..e3afc3dac7d1 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -417,7 +417,7 @@ int snd_sof_init_trace_ipc(struct snd_sof_dev *sdev) "error: fail in snd_sof_dma_trace_init %d\n", ret); return ret; } - dev_dbg(sdev->dev, "stream_tag: %d\n", params.stream_tag); + dev_dbg(sdev->dev, "%s: stream_tag: %d\n", __func__, params.stream_tag); /* send IPC to the DSP */ ret = sof_ipc_tx_message(sdev->ipc, @@ -480,7 +480,8 @@ int snd_sof_init_trace(struct snd_sof_dev *sdev) goto table_err; sdev->dma_trace_pages = ret; - dev_dbg(sdev->dev, "dma_trace_pages: %d\n", sdev->dma_trace_pages); + dev_dbg(sdev->dev, "%s: dma_trace_pages: %d\n", + __func__, sdev->dma_trace_pages); if (sdev->first_boot) { ret = trace_debugfs_create(sdev); -- 2.33.0

2 1

[PATCH] ASoC: intel: sof_rt5682: update platform device name for Maxim amplifier
by Brent Lu 28 Sep '21

28 Sep '21

To follow 20-character length limitation of platform device name, we have only 7 character space for amplifier. Therefore, the last character of mx98357a and mx98360a is removed to save space. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 8 ++++---- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 4 ++-- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 2 +- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 3f6f19d9b19e..ad42d4c7ade5 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -959,7 +959,7 @@ static const struct platform_device_id board_ids[] = { .name = "sof_rt5682", }, { - .name = "tgl_mx98357a_rt5682", + .name = "tgl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | @@ -989,7 +989,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "jsl_rt5682_mx98360a", + .name = "jsl_rt5682_mx98360", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | @@ -1039,7 +1039,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "adl_mx98357a_rt5682", + .name = "adl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | @@ -1067,7 +1067,7 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1)), }, { - .name = "jsl_rt5682s_mx98360a", + .name = "jsl_rt5682s_mx98360", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..e4ff280eac23 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -291,7 +291,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10EC5682", - .drv_name = "adl_mx98357a_rt5682", + .drv_name = "adl_mx98357_rt5682", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98357a_amp, .sof_fw_filename = "sof-adl.ri", diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index cfd6bf756215..20fd9dcc74af 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -67,7 +67,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { }, { .id = "10EC5682", - .drv_name = "jsl_rt5682_mx98360a", + .drv_name = "jsl_rt5682_mx98360", .sof_fw_filename = "sof-jsl.ri", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98360a_spk, @@ -99,7 +99,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { }, { .id = "RTL5682", - .drv_name = "jsl_rt5682s_mx98360a", + .drv_name = "jsl_rt5682s_mx98360", .sof_fw_filename = "sof-jsl.ri", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98360a_spk, diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 785d5f5f8a9c..91ef71c2535d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -333,7 +333,7 @@ static const struct snd_soc_acpi_codecs tgl_rt1011_amp = { struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC5682", - .drv_name = "tgl_mx98357a_rt5682", + .drv_name = "tgl_mx98357_rt5682", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &tgl_codecs, .sof_fw_filename = "sof-tgl.ri", -- 2.25.1

3 2

[PATCH] ASoC: SOF: ipc: Make the error prints consistent in tx_wait_done()
by Peter Ujfalusi 28 Sep '21

28 Sep '21

If we get an error on reply (msg->reply_error) then we should print the error value out. At the same time extend the print to include the message size as well and do the same in case of a timeout. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 5d41924f37a6..95b1d875c3c0 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -249,15 +249,17 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, msecs_to_jiffies(sdev->ipc_timeout)); if (ret == 0) { - dev_err(sdev->dev, "error: ipc timed out for 0x%x size %d\n", - hdr->cmd, hdr->size); + dev_err(sdev->dev, + "ipc tx timed out for %#x (msg/reply size: %d/%zu)\n", + hdr->cmd, hdr->size, msg->reply_size); snd_sof_handle_fw_exception(ipc->sdev); ret = -ETIMEDOUT; } else { ret = msg->reply_error; if (ret < 0) { - dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n", - hdr->cmd, msg->reply_size); + dev_err(sdev->dev, + "ipc tx error for %#x (msg/reply size: %d/%zu): %d\n", + hdr->cmd, hdr->size, msg->reply_size, ret); } else { ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); if (msg->reply_size) -- 2.33.0

2 1

[PATCH] ASoC: mediatek: mt8195: add missing of_node_put in probe
by Trevor Wu 28 Sep '21

28 Sep '21

dp node and hdmi node are retrieved from of_parse_phandle(), so using of_node_put() on them before return. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 25 +++++++++++-------- 1 file changed, 15 insertions(+), 10 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index a3fa8efc8f81..cb15467e5fc5 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -995,6 +995,8 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8195_mt6359_rt1019_rt5682_soc_card; struct device_node *platform_node; + struct device_node *dp_node; + struct device_node *hdmi_node; struct snd_soc_dai_link *dai_link; struct mt8195_mt6359_rt1019_rt5682_priv *priv = NULL; int ret, i; @@ -1013,12 +1015,12 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) dai_link->platforms->of_node = platform_node; if (strcmp(dai_link->name, "DPTX_BE") == 0) { - dai_link->codecs->of_node = - of_parse_phandle(pdev->dev.of_node, - "mediatek,dptx-codec", 0); - if (!dai_link->codecs->of_node) { + dp_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,dptx-codec", 0); + if (!dp_node) { dev_dbg(&pdev->dev, "No property 'dptx-codec'\n"); } else { + dai_link->codecs->of_node = dp_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_dptx_codec_init; @@ -1026,12 +1028,12 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - dai_link->codecs->of_node = - of_parse_phandle(pdev->dev.of_node, - "mediatek,hdmi-codec", 0); - if (!dai_link->codecs->of_node) { + hdmi_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,hdmi-codec", 0); + if (!hdmi_node) { dev_dbg(&pdev->dev, "No property 'hdmi-codec'\n"); } else { + dai_link->codecs->of_node = hdmi_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_hdmi_codec_init; @@ -1041,8 +1043,8 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) { - of_node_put(platform_node); - return -ENOMEM; + ret = -ENOMEM; + goto out; } snd_soc_card_set_drvdata(card, priv); @@ -1052,6 +1054,9 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +out: + of_node_put(hdmi_node); + of_node_put(dp_node); of_node_put(platform_node); return ret; } -- 2.18.0

2 1

[PATCH] ASoC: SOF: pm: fix a stale comment
by Peter Ujfalusi 28 Sep '21

28 Sep '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> There is no restore_stream flag anymmore. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/pm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index c83fb6255961..89b4b3d76539 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -191,7 +191,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) goto suspend; - /* set restore_stream for all streams during system suspend */ + /* prepare for streams to be resumed properly upon resume */ if (!runtime_suspend) { ret = sof_set_hw_params_upon_resume(sdev->dev); if (ret < 0) { -- 2.33.0

2 1

[PATCH] ASoC: SOF: Intel: hda-stream: Print stream name on STREAM_SD_OFFSET timeout
by Peter Ujfalusi 28 Sep '21

28 Sep '21

In order to provide more information in case of timeout observed while reading STREAM_SD_OFFSET, print out the stream name or in case there is no audio stream associated (like dma-trace), print "--" Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 85 ++++++++++++++++++++++---------- 1 file changed, 59 insertions(+), 26 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..db028b83c5f2 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -25,6 +25,33 @@ #define HDA_LTRP_GB_VALUE_US 95 +static inline const char *hda_hstream_direction_str(struct hdac_stream *hstream) +{ + if (hstream->direction == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + +static char *hda_hstream_dbg_get_stream_info_str(struct hdac_stream *hstream) +{ + struct snd_soc_pcm_runtime *rtd; + + if (hstream->substream) + rtd = asoc_substream_to_rtd(hstream->substream); + else if (hstream->cstream) + rtd = hstream->cstream->private_data; + else + /* Non audio DMA user, like dma-trace */ + return kasprintf(GFP_KERNEL, "-- (%s, stream_tag: %u)", + hda_hstream_direction_str(hstream), + hstream->stream_tag); + + return kasprintf(GFP_KERNEL, "dai_link \"%s\" (%s, stream_tag: %u)", + rtd->dai_link->name, hda_hstream_direction_str(hstream), + hstream->stream_tag); +} + /* * set up one of BDL entries for a stream */ @@ -257,7 +284,7 @@ int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, struct hdac_stream *hstream = &stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); u32 dma_start = SOF_HDA_SD_CTL_DMA_START; - int ret; + int ret = 0; u32 run; /* cmd must be for audio stream */ @@ -283,14 +310,9 @@ int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_STREAM_RUN_TIMEOUT); - if (ret < 0) { - dev_err(sdev->dev, - "error: %s: cmd %d: timeout on STREAM_SD_OFFSET read\n", - __func__, cmd); - return ret; - } + if (ret >= 0) + hstream->running = true; - hstream->running = true; break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -306,27 +328,32 @@ int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_STREAM_RUN_TIMEOUT); - if (ret < 0) { - dev_err(sdev->dev, - "error: %s: cmd %d: timeout on STREAM_SD_OFFSET read\n", - __func__, cmd); - return ret; - } + if (ret >= 0) { + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_STS, + SOF_HDA_CL_DMA_SD_INT_MASK); - snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, sd_offset + - SOF_HDA_ADSP_REG_CL_SD_STS, - SOF_HDA_CL_DMA_SD_INT_MASK); - - hstream->running = false; - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, - 1 << hstream->index, 0x0); + hstream->running = false; + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + 1 << hstream->index, 0x0); + } break; default: dev_err(sdev->dev, "error: unknown command: %d\n", cmd); return -EINVAL; } - return 0; + if (ret < 0) { + char *stream_name = hda_hstream_dbg_get_stream_info_str(hstream); + + dev_err(sdev->dev, + "%s: cmd %d on %s: timeout on STREAM_SD_OFFSET read\n", + __func__, cmd, stream_name ? stream_name : "unknown stream"); + kfree(stream_name); + } + + return ret; } /* minimal recommended programming for ICCMAX stream */ @@ -440,9 +467,12 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, HDA_DSP_STREAM_RUN_TIMEOUT); if (ret < 0) { + char *stream_name = hda_hstream_dbg_get_stream_info_str(hstream); + dev_err(sdev->dev, - "error: %s: timeout on STREAM_SD_OFFSET read1\n", - __func__); + "%s: on %s: timeout on STREAM_SD_OFFSET read1\n", + __func__, stream_name ? stream_name : "unknown stream"); + kfree(stream_name); return ret; } @@ -506,9 +536,12 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, HDA_DSP_STREAM_RUN_TIMEOUT); if (ret < 0) { + char *stream_name = hda_hstream_dbg_get_stream_info_str(hstream); + dev_err(sdev->dev, - "error: %s: timeout on STREAM_SD_OFFSET read2\n", - __func__); + "%s: on %s: timeout on STREAM_SD_OFFSET read1\n", + __func__, stream_name ? stream_name : "unknown stream"); + kfree(stream_name); return ret; } -- 2.33.0

2 1

[PATCH] ASoC: SOF: imx: add header file for ops
by Peter Ujfalusi 28 Sep '21

28 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Sparse (make C=2) complains about undeclared variables. Fix by adding a real prototype instead of 'extern' in sof-of-dev.c Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/imx/imx-ops.h | 10 ++++++++++ sound/soc/sof/imx/imx8.c | 1 + sound/soc/sof/imx/imx8m.c | 1 + sound/soc/sof/sof-of-dev.c | 5 +---- 4 files changed, 13 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/imx/imx-ops.h diff --git a/sound/soc/sof/imx/imx-ops.h b/sound/soc/sof/imx/imx-ops.h new file mode 100644 index 000000000000..24235ef8c8fa --- /dev/null +++ b/sound/soc/sof/imx/imx-ops.h @@ -0,0 +1,10 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ + +#ifndef __IMX_OPS_H__ +#define __IMX_OPS_H__ + +extern struct snd_sof_dsp_ops sof_imx8_ops; +extern struct snd_sof_dsp_ops sof_imx8x_ops; +extern struct snd_sof_dsp_ops sof_imx8m_ops; + +#endif diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 82ca920a69ab..cf27ee66b577 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -22,6 +22,7 @@ #include <dt-bindings/firmware/imx/rsrc.h> #include "../ops.h" #include "imx-common.h" +#include "imx-ops.h" /* DSP memories */ #define IRAM_OFFSET 0x10000 diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 76f4ffd9d540..932c0e73a8b0 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -18,6 +18,7 @@ #include "../ops.h" #include "imx-common.h" +#include "imx-ops.h" #define MBOX_OFFSET 0x800000 #define MBOX_SIZE 0x1000 diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index d1a21edfa05d..f0f819a46456 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -11,10 +11,7 @@ #include <sound/sof.h> #include "ops.h" - -extern struct snd_sof_dsp_ops sof_imx8_ops; -extern struct snd_sof_dsp_ops sof_imx8x_ops; -extern struct snd_sof_dsp_ops sof_imx8m_ops; +#include "imx/imx-ops.h" /* platform specific devices */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) -- 2.33.0

2 1

[PATCH v8 00/22] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 28 Sep '21

28 Sep '21

Many thanks for reviewing v7 This version addresses all the comments raised as part of v8 review. This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v7: - removed returning errors in dsp command callbacks as it has no effect - removed uncessary err prints - removed default topology file as its impossible to have one for all boards. - removed references of Elite in commit log - return correct error codes returned from functions. Srinivas Kandagatla (22): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 177 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 287 ++++- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1135 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 415 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 823 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++ 31 files changed, 7170 insertions(+), 1415 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

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[patch 00/11] hrtimers: Cleanup hrtimer_forward() [ab]use
by Thomas Gleixner 28 Sep '21

28 Sep '21

A recent syzbot report unearthed abuse of hrtimer_forward() which can cause runaway timers hogging the CPU in timer expiry context by rearming the timer in the past over and over. This happens when the caller uses timer->expiry for the 'now' argument of hrtimer_forward(). That works as long as the timer expiry is on time, but can cause a long period of rearm/fire loops which hog the CPU. Expiring late can have various causes, but obviously virtualization is prone to that due to VCPU scheduling. The correct usage of hrtimer_forward() is to hand the current time to the 'now' argument which ensures that the next event on the periodic time line is past now. This is what hrtimer_forward_now() provides. The following series addresses this: 1) Add a debug mechanism to the hrtimer expiry loop 2) Convert all hrtimer_forward() usage outside of kernel/time/ to use hrtimer_forward_now(). 3) Confine hrtimer_forward() to kernel/time/ core code. The mac80211_hwsim patch has already been picked up by the wireless maintainer and all other patches which affect usage outside the core code can be picked up by the relevant subsystems. If a maintainer wants me to pick a particular patch up, please let me know. The last patch which confines hrtimer_forward() will be postponed until all other patches have been merged into Linus tree. The series is also available from git: git://git.kernel.org/pub/scm/linux/kernel/git/tglx/devel.git hrtimer Thanks, tglx --- drivers/gpu/drm/i915/i915_pmu.c | 2 - drivers/net/wireless/mac80211_hwsim.c | 4 +- drivers/net/wwan/iosm/iosm_ipc_imem.c | 4 +- drivers/power/reset/ltc2952-poweroff.c | 4 -- include/linux/hrtimer.h | 26 ----------------- include/linux/posix-timers.h | 3 ++ kernel/signal.c | 14 +-------- kernel/time/hrtimer.c | 48 ++++++++++++++++++++++++++++++++- kernel/time/itimer.c | 13 ++++++++ kernel/time/posix-timers.c | 42 +++++++++++----------------- kernel/time/tick-internal.h | 1 net/can/bcm.c | 2 - sound/drivers/pcsp/pcsp_lib.c | 2 - 13 files changed, 92 insertions(+), 73 deletions(-)

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[PATCH] snd-usb-audio: fix comment reference in __uac_clock_find_source
by Geraldo Nascimento 28 Sep '21

28 Sep '21

snd_usb_find_clock_source and snd_usb_find_clock_selector are helper macros that look at an entity id and validate that this entity id is in fact a clock source or a clock selector. The present comments inside __uac_clock_find_source give the reader the impression we're looking for an entity id. We're looking for an entity id indeed, the clock source, but since __uac_clock_find_source is recursive, we're also looking *at* the entity ids, in the search for the one clock source. Fix the comment so we don't give readers a wrong idea. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 81d5ce07d548..7dd71d342443 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -271,7 +271,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } - /* first, see if the ID we're looking for is a clock source already */ + /* first, see if the ID we're looking at is a clock source already */ source = snd_usb_find_clock_source(chip, entity_id, proto); if (source) { entity_id = GET_VAL(source, proto, bClockID); @@ -297,7 +297,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, goto find_source; } - /* the entity ID we are looking for is a selector. + /* the entity ID we are looking at is a selector. * find out what it currently selects */ ret = uac_clock_selector_get_val(chip, clock_id); if (ret < 0) {

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