January 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: pcm: accept the OPEN state for snd_pcm_stop()
by Jaroslav Kysela 14 Feb '22

14 Feb '22

It may be useful to completely invalidate streaming under some circumstances like the USB gadget detach. This case is a bit different than the complete disconnection. The applications should be notified that the PCM streaming is no longer available, but the recovery may be expected. This patch adds support for SNDRV_PCM_STATE_OPEN passed to snd_pcm_stop() function. Only the hw_free ioctl is allowed to free the buffers in this state for a full recovery operation or the PCM file handle must be closed. In the OPEN state, ioctls return EBADFD error (with the added hw_free exception above). The applications which are not aware about this new state transition (and recovery) will fail with an error. This operation is expected. Link: https://lore.kernel.org/alsa-devel/4e17c99b-1d8a-8ebe-972c-9b1299952ff8@ivi… Cc: Pavel Hofman <pavel.hofman(a)ivitera.com> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/pcm.h | 1 + sound/core/pcm.c | 1 + sound/core/pcm_native.c | 12 +++++++++++- 3 files changed, 13 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 33451f8ff755..4de1c2c91109 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -467,6 +467,7 @@ struct snd_pcm_substream { /* -- assigned files -- */ int ref_count; atomic_t mmap_count; + atomic_t queued_hw_free; unsigned int f_flags; void (*pcm_release)(struct snd_pcm_substream *); struct pid *pid; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6fd3677685d7..8dc7e99b2113 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -694,6 +694,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) snd_pcm_group_init(&substream->self_group); list_add_tail(&substream->link_list, &substream->self_group.substreams); atomic_set(&substream->mmap_count, 0); + atomic_set(&substream->queued_hw_free, 0); prev = substream; } return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 621883e71194..118ad3f26f4a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -686,10 +686,13 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: + if (atomic_read(&substream->queued_hw_free)) + goto __badfd; case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: break; default: +__badfd: snd_pcm_stream_unlock_irq(substream); return -EBADFD; } @@ -829,6 +832,7 @@ static int do_hw_free(struct snd_pcm_substream *substream) result = substream->ops->hw_free(substream); if (substream->managed_buffer_alloc) snd_pcm_lib_free_pages(substream); + atomic_set(&substream->queued_hw_free, 0); return result; } @@ -1454,6 +1458,8 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, } wake_up(&runtime->sleep); wake_up(&runtime->tsleep); + if (state == SNDRV_PCM_STATE_OPEN) + atomic_set(&substream->queued_hw_free, 1); } static const struct action_ops snd_pcm_action_stop = { @@ -1469,6 +1475,9 @@ static const struct action_ops snd_pcm_action_stop = { * * The state of each stream is then changed to the given state unconditionally. * + * If the requested state is OPEN, the stream is invalidated and + * the application must call hw_free to recover the operation. + * * Return: Zero if successful, or a negative error code. */ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t state) @@ -2637,7 +2646,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) + if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN || + atomic_read(&substream->queued_hw_free)) do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; -- 2.33.1

5 7

[PATCH 00/19] soundwire: stream: cleanup of 'stream' support
by Bard Liao 11 Feb '22

11 Feb '22

This series revisits the SoundWire 'sdw_stream' support to split allocation and configuration steps. This is necessary if for example the routines are called multiple times from the hw_params stage. This also helps with better error handling. Pierre-Louis Bossart (19): soundwire: stream: remove unused parameter in sdw_stream_add_slave soundwire: stream: add slave runtime to list earlier soundwire: stream: simplify check on port range soundwire: stream: add alloc/config/free helpers for ports soundwire: stream: split port allocation and configuration loops soundwire: stream: split alloc and config in two functions soundwire: stream: add 'slave' prefix for port range checks soundwire: stream: group sdw_port and sdw_master/slave_port functions soundwire: stream: simplify sdw_alloc_master_rt() soundwire: stream: split sdw_alloc_master_rt() in alloc and config soundwire: stream: move sdw_alloc_slave_rt() before 'master' helpers soundwire: stream: split sdw_alloc_slave_rt() in alloc and config soundwire: stream: group sdw_stream_ functions soundwire: stream: rename and move master/slave_rt_free routines soundwire: stream: move list addition to sdw_slave_alloc_rt() soundwire: stream: separate alloc and config within sdw_stream_add_xxx() soundwire: stream: introduce sdw_slave_rt_find() helper soundwire: stream: sdw_stream_add_ functions can be called multiple times soundwire: stream: make enable/disable/deprepare idempotent drivers/soundwire/stream.c | 960 +++++++++++++++++++++---------------- 1 file changed, 547 insertions(+), 413 deletions(-) -- 2.17.1

2 20

[PATCH v2 0/6] ASoC: dt-bindings: samsung: convert to dtschema
by Krzysztof Kozlowski 10 Feb '22

10 Feb '22

Hi, The patchset is based on Rob's sound-dai changes: https://lore.kernel.org/all/20220126231427.1638089-1-robh@kernel.org/ Changes since v1: 1. Correct samsung,snow cpu/sound-dai. Best regards, Krzysztof Krzysztof Kozlowski (6): ASoC: dt-bindings: samsung,aries-wm8994: require sound-dai property ASoC: dt-bindings: samsung,arndale: convert to dtschema ASoC: dt-bindings: samsung,arndale: document ALC5631 ASoC: dt-bindings: samsung,smdk5250: convert to dtschema ASoC: dt-bindings: samsung,snow: convert to dtschema ASoC: dt-bindings: samsung,tm2: convert to dtschema .../devicetree/bindings/sound/arndale.txt | 25 ------ .../bindings/sound/samsung,aries-wm8994.yaml | 4 + .../bindings/sound/samsung,arndale.yaml | 45 +++++++++++ .../bindings/sound/samsung,smdk-wm8994.txt | 14 ---- .../bindings/sound/samsung,smdk5250.yaml | 38 +++++++++ .../bindings/sound/samsung,snow.yaml | 74 +++++++++++++++++ .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2.yaml | 80 +++++++++++++++++++ .../devicetree/bindings/sound/snow.txt | 31 ------- 9 files changed, 241 insertions(+), 112 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/arndale.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,arndale.yaml delete mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk-wm8994.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,smdk5250.yaml create mode 100644 Documentation/devicetree/bindings/sound/samsung,snow.yaml delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/snow.txt -- 2.32.0

3 13

[PATCH] soundwire: Replace acpi_bus_get_device()
by Rafael J. Wysocki 10 Feb '22

10 Feb '22

From: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> Replace acpi_bus_get_device() that is going to be dropped with acpi_fetch_acpi_dev(). No intentional functional impact. Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki(a)intel.com> --- drivers/soundwire/intel_init.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) Index: linux-pm/drivers/soundwire/intel_init.c =================================================================== --- linux-pm.orig/drivers/soundwire/intel_init.c +++ linux-pm/drivers/soundwire/intel_init.c @@ -180,7 +180,8 @@ static struct sdw_intel_ctx if (!res) return NULL; - if (acpi_bus_get_device(res->handle, &adev)) + adev = acpi_fetch_acpi_dev(res->handle); + if (!adev) return NULL; if (!res->count) @@ -294,13 +295,13 @@ err: static int sdw_intel_startup_controller(struct sdw_intel_ctx *ctx) { - struct acpi_device *adev; + struct acpi_device *adev = acpi_fetch_acpi_dev(ctx->handle); struct sdw_intel_link_dev *ldev; u32 caps; u32 link_mask; int i; - if (acpi_bus_get_device(ctx->handle, &adev)) + if (!adev) return -EINVAL; /* Check SNDWLCAP.LCOUNT */

2 1

[PATCH] soundwire: bus: add dev_warn() messages to track UNATTACHED devices
by Bard Liao 10 Feb '22

10 Feb '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> In rare cases, some devices seem to lose sync and never re-attach on the bus. This seems to happen only when there are more than one device per link, which suggests either an electrical issue, a race condition or a state machine issue. Add two dev_warn() messages to identify the sequence by which the devices become UNATTACHED. BugLink: https://github.com/thesofproject/linux/issues/3063 BugLink: https://github.com/thesofproject/linux/issues/3325 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 67369e941d0d..354d3f89366f 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1749,8 +1749,11 @@ int sdw_handle_slave_status(struct sdw_bus *bus, continue; if (status[i] == SDW_SLAVE_UNATTACHED && - slave->status != SDW_SLAVE_UNATTACHED) + slave->status != SDW_SLAVE_UNATTACHED) { + dev_warn(&slave->dev, "Slave %d state check1: UNATTACHED, status was %d\n", + i, slave->status); sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); + } } if (status[0] == SDW_SLAVE_ATTACHED) { @@ -1785,6 +1788,9 @@ int sdw_handle_slave_status(struct sdw_bus *bus, if (slave->status == SDW_SLAVE_UNATTACHED) break; + dev_warn(&slave->dev, "Slave %d state check2: UNATTACHED, status was %d\n", + i, slave->status); + sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); break; -- 2.17.1

2 1

[PATCH] soundwire: intel: fix wrong register name in intel_shim_wake
by Bard Liao 10 Feb '22

10 Feb '22

From: Libin Yang <libin.yang(a)intel.com> When clearing the sdw wakests status, we should use SDW_SHIM_WAKESTS. Fixes: 4a17c441c7cb ("soundwire: intel: revisit SHIM programming sequences.") Signed-off-by: Libin Yang <libin.yang(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 122f7a29d8ca..63101f1ba271 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -448,8 +448,8 @@ static void intel_shim_wake(struct sdw_intel *sdw, bool wake_enable) /* Clear wake status */ wake_sts = intel_readw(shim, SDW_SHIM_WAKESTS); - wake_sts |= (SDW_SHIM_WAKEEN_ENABLE << link_id); - intel_writew(shim, SDW_SHIM_WAKESTS_STATUS, wake_sts); + wake_sts |= (SDW_SHIM_WAKESTS_STATUS << link_id); + intel_writew(shim, SDW_SHIM_WAKESTS, wake_sts); } mutex_unlock(sdw->link_res->shim_lock); } -- 2.17.1

2 1

[PATCH 1/4] ALSA: hda/realtek: fix mute/micmute LEDs for HP 855 G8
by Jeremy Szu 08 Feb '22

08 Feb '22

The HP EliteBook 855 G8 Notebook PC is using ALC285 codec which needs ALC285_FIXUP_HP_MUTE_LED fixup to make it works. After applying the fixup, the mute/micmute LEDs work good. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 552e2cb73291..9d68f591c6bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8291,6 +8291,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), -- 2.31.1

3 32

[PATCH] ASoC: fsl_sai: Enable combine mode soft
by Sascha Hauer 08 Feb '22

08 Feb '22

The fsl_sai driver calculates the number of pins used and enables multiple channels if necessary. This means the SAI expects data in one FIFO per pin. The SDMA engine only services a single FIFO, so multi pin support doesn't work at all. This patch enables the software combine mode in chips that support it. With this the SAI presents only a single FIFO to the outside and distributes the data into the different FIFOs internally. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_sai.c | 9 +++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 10 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 10544fa27dc0..cab015d96889 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -517,6 +517,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } + if (sai->soc_data->pins > 1) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FCOMB_MASK, FSL_SAI_CR4_FCOMB_SOFT); + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx, ofs), FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE((1 << pins) - 1)); @@ -1195,6 +1199,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, .use_edma = false, .fifo_depth = 32, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, @@ -1204,6 +1209,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 32, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = true, .flags = 0, @@ -1213,6 +1219,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 16, + .pins = 2, .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = PMQOS_CPU_LATENCY, @@ -1222,6 +1229,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, .use_edma = false, .fifo_depth = 128, + .pins = 8, .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = 0, @@ -1231,6 +1239,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .use_edma = true, .fifo_depth = 64, + .pins = 1, .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 9aaf231bc024..410f6e6a9137 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -223,6 +223,7 @@ struct fsl_sai_soc_data { bool use_edma; bool mclk0_is_mclk1; unsigned int fifo_depth; + unsigned int pins; unsigned int reg_offset; unsigned int flags; }; -- 2.30.2

2 2

[PATCH v3] ASoC: SOF: compr: Add compress ops implementation
by Daniel Baluta 08 Feb '22

08 Feb '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> Implement snd_compress_ops. There are a lot of similarities with PCM implementation. For now we use sof_ipc_pcm_params to transfer compress parameters to SOF firmware. This will be changed in the future once we either add new compress parameters to SOF or enhance existing sof_ipc_pcm_params structure to support all native compress params. Note that get_caps and get_codec_caps are missing and will be added later. This is because we need to find a way to advertise DSP capabilities depending on supported platforms. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- Changes since v2: - dropped patch 2/2 because deciding on how to support get_caps and get_codec_caps needs some rework. - annotated commit message for 1/2 to mention about get_caps and get_codec_caps Changes since v1: (https://lore.kernel.org/lkml/20220113161341.371345-1-daniel.baluta@oss.nxp.…) - Addressed review from Cezary and Pierre - fixed layout of declaration blocks to be more consistent - avoid using rtd and runtime simultaneously inside a function (always used rtd and crtd) - check return code for create_page_table - completely remove sof_compr_stream and use snd_compr_tstmap instead to keep compress stream related info.· - add get_caps and get_codec_caps implementations (in patch 2)· sound/soc/sof/compress.c | 273 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 271 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 01ca85f0b87f..91a9c95929cd 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -10,6 +10,22 @@ #include "sof-audio.h" #include "sof-priv.h" +static void sof_set_transferred_bytes(struct snd_compr_tstamp *tstamp, + u64 host_pos, u64 buffer_size) +{ + u64 prev_pos; + unsigned int copied; + + div64_u64_rem(tstamp->copied_total, buffer_size, &prev_pos); + + if (host_pos < prev_pos) + copied = (buffer_size - prev_pos) + host_pos; + else + copied = host_pos - prev_pos; + + tstamp->copied_total += copied; +} + static void snd_sof_compr_fragment_elapsed_work(struct work_struct *work) { struct snd_sof_pcm_stream *sps = @@ -29,14 +45,16 @@ void snd_sof_compr_init_elapsed_work(struct work_struct *work) */ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) { + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; struct snd_soc_component *component; - struct snd_soc_pcm_runtime *rtd; + struct snd_compr_tstamp *tstamp; struct snd_sof_pcm *spcm; if (!cstream) return; - rtd = cstream->private_data; + tstamp = crtd->private_data; component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); spcm = snd_sof_find_spcm_dai(component, rtd); @@ -46,6 +64,257 @@ void snd_sof_compr_fragment_elapsed(struct snd_compr_stream *cstream) return; } + sof_set_transferred_bytes(tstamp, spcm->stream[cstream->direction].posn.host_posn, + crtd->buffer_size); + /* use the same workqueue-based solution as for PCM, cf. snd_sof_pcm_elapsed */ schedule_work(&spcm->stream[cstream->direction].period_elapsed_work); } + +static int create_page_table(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + unsigned char *dma_area, size_t size) +{ + struct snd_dma_buffer *dmab = cstream->runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + int dir = cstream->direction; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + return snd_sof_create_page_table(component->dev, dmab, + spcm->stream[dir].page_table.area, size); +} + +int sof_compr_open(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; + struct snd_compr_tstamp *tstamp; + struct snd_sof_pcm *spcm; + int dir; + + tstamp = kzalloc(sizeof(*tstamp), GFP_KERNEL); + if (!tstamp) + return -ENOMEM; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) { + kfree(tstamp); + return -EINVAL; + } + + dir = cstream->direction; + + if (spcm->stream[dir].cstream) { + kfree(tstamp); + return -EBUSY; + } + + spcm->stream[dir].cstream = cstream; + spcm->stream[dir].posn.host_posn = 0; + spcm->stream[dir].posn.dai_posn = 0; + spcm->prepared[dir] = false; + + crtd->private_data = tstamp; + + return 0; +} + +int sof_compr_free(struct snd_soc_component *component, + struct snd_compr_stream *cstream) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_compr_tstamp *tstamp = cstream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_sof_pcm *spcm; + int ret = 0; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_FREE; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + if (spcm->prepared[cstream->direction]) { + ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); + if (!ret) + spcm->prepared[cstream->direction] = false; + } + + cancel_work_sync(&spcm->stream[cstream->direction].period_elapsed_work); + spcm->stream[cstream->direction].cstream = NULL; + kfree(tstamp); + + return ret; +} + +int sof_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_compr_params *params) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_compr_runtime *crtd = cstream->runtime; + struct sof_ipc_pcm_params_reply ipc_params_reply; + struct snd_compr_tstamp *tstamp; + struct sof_ipc_pcm_params pcm; + struct snd_sof_pcm *spcm; + int ret; + + tstamp = crtd->private_data; + + spcm = snd_sof_find_spcm_dai(component, rtd); + + if (!spcm) + return -EINVAL; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, crtd->buffer_size); + if (ret < 0) + return ret; + + ret = create_page_table(component, cstream, crtd->dma_area, crtd->dma_bytes); + if (ret < 0) + return ret; + + memset(&pcm, 0, sizeof(pcm)); + + pcm.params.buffer.pages = PFN_UP(crtd->dma_bytes); + pcm.hdr.size = sizeof(pcm); + pcm.hdr.cmd = SOF_IPC_GLB_STREAM_MSG | SOF_IPC_STREAM_PCM_PARAMS; + + pcm.comp_id = spcm->stream[cstream->direction].comp_id; + pcm.params.hdr.size = sizeof(pcm.params); + pcm.params.buffer.phy_addr = spcm->stream[cstream->direction].page_table.addr; + pcm.params.buffer.size = crtd->dma_bytes; + pcm.params.direction = cstream->direction; + pcm.params.channels = params->codec.ch_out; + pcm.params.rate = params->codec.sample_rate; + pcm.params.buffer_fmt = SOF_IPC_BUFFER_INTERLEAVED; + pcm.params.frame_fmt = SOF_IPC_FRAME_S32_LE; + pcm.params.sample_container_bytes = + snd_pcm_format_physical_width(SNDRV_PCM_FORMAT_S32) >> 3; + pcm.params.host_period_bytes = params->buffer.fragment_size; + + ret = sof_ipc_tx_message(sdev->ipc, pcm.hdr.cmd, &pcm, sizeof(pcm), + &ipc_params_reply, sizeof(ipc_params_reply)); + if (ret < 0) { + dev_err(component->dev, "error ipc failed\n"); + return ret; + } + + tstamp->byte_offset = sdev->stream_box.offset + ipc_params_reply.posn_offset; + tstamp->sampling_rate = params->codec.sample_rate; + + spcm->prepared[cstream->direction] = true; + + return 0; +} + +int sof_compr_get_params(struct snd_soc_component *component, + struct snd_compr_stream *cstream, struct snd_codec *params) +{ + /* TODO: we don't query the supported codecs for now, if the + * application asks for an unsupported codec the set_params() will fail. + */ + return 0; +} + +int sof_compr_trigger(struct snd_soc_component *component, + struct snd_compr_stream *cstream, int cmd) +{ + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct sof_ipc_stream stream; + struct sof_ipc_reply reply; + struct snd_sof_pcm *spcm; + + spcm = snd_sof_find_spcm_dai(component, rtd); + if (!spcm) + return -EINVAL; + + stream.hdr.size = sizeof(stream); + stream.hdr.cmd = SOF_IPC_GLB_STREAM_MSG; + stream.comp_id = spcm->stream[cstream->direction].comp_id; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_START; + break; + case SNDRV_PCM_TRIGGER_STOP: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE; + break; + default: + dev_err(component->dev, "error: unhandled trigger cmd %d\n", cmd); + break; + } + + return sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, + &stream, sizeof(stream), + &reply, sizeof(reply)); +} + +int sof_compr_copy(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_available, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_from_user(ptr, buf, count); + } else { + ret = copy_from_user(ptr, buf, n); + ret += copy_from_user(rtd->dma_area, buf + n, count - n); + } + + return count - ret; +} + +static int sof_compr_pointer(struct snd_soc_component *component, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp) +{ + struct snd_compr_tstamp *pstamp = cstream->runtime->private_data; + + tstamp->sampling_rate = pstamp->sampling_rate; + tstamp->copied_total = pstamp->copied_total; + + return 0; +} + +struct snd_compress_ops sof_compressed_ops = { + .open = sof_compr_open, + .free = sof_compr_free, + .set_params = sof_compr_set_params, + .get_params = sof_compr_get_params, + .trigger = sof_compr_trigger, + .pointer = sof_compr_pointer, + .copy = sof_compr_copy, +}; +EXPORT_SYMBOL(sof_compressed_ops); -- 2.27.0

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[PATCH 0/2] ASoC: SOF: dma-trace: Change trace_init() ops parameter list
by Peter Ujfalusi 08 Feb '22

08 Feb '22

Hi, the DMA trace implementation on AMD platform assumes that the stream_tag pointer is pointing the stream_tag member of struct sof_ipc_dma_trace_params_ext, which is true at the moment, but it can not be guarantied and a change in the dtrace core can cause out of bound accesses for AMD. For this reason, change the API to pass the struct itself which will remove the assumption and makes it clear from both sides what is expected to be sent via the parameter list. This opens up a window to clean up the intel and AMD implementation at the same time. Regards, Peter --- Peter Ujfalusi (2): ASoC: SOF: intel: hda-trace: Pass the dma buffer pointer to hda_dsp_trace_prepare ASoC: SOF: dma-trace: Pass pointer to params_ext struct in trace_init() sound/soc/sof/amd/acp-trace.c | 38 ++++++++------------------------- sound/soc/sof/amd/acp.h | 3 ++- sound/soc/sof/intel/hda-trace.c | 17 ++++++++------- sound/soc/sof/intel/hda.h | 3 ++- sound/soc/sof/ops.h | 4 ++-- sound/soc/sof/sof-priv.h | 2 +- sound/soc/sof/trace.c | 2 +- 7 files changed, 26 insertions(+), 43 deletions(-) -- 2.35.0

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