December 2024 - Alsa-devel - mailman.alsa-project.org

[RFC PATCH] soc: audio-graph-card2: use correct endpoint when getting link parameters
by Ivaylo Dimitrov 20 Dec '24

20 Dec '24

We may have multiple links between ports, with each link having different parameters. Currently, no matter the topology, it is always port endpoint 0 that is used when setting parameters. On a complex sound system, like the one found on Motorola droid4, hifi and voice DAIs require differents formats (i2s vs dsp_a) and curently it is impossible to use DT to set that. Implementing the change leads to partially dropping of at least 0dedbde5062d (ASoC: cpcap: Implement set_tdm_slot for voice call support), as core does most of what is needed to configure voice DAI. We (on Maemo Leste ) use the patch (along with few others) to have voice calls working properly on d4 through UCM. The patch is for linux 6.6, I want to know whether the approach would be accepted before sending a proper patch for current master. the original commit message follows: When link parameters are parsed, it is always endpoint@0 that is used and parameters set to other endpoints are ignored. Fix that by using endpoint that is set in DT when parsing link parameters. Signed-off-by: Ivaylo Dimitrov <ivo.g.dimitrov.75(a)gmail.com> --- sound/soc/generic/audio-graph-card2.c | 59 +++++++++++++-------------- 1 file changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..163a20c8ffee 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -508,17 +508,16 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, static int graph_parse_node(struct asoc_simple_priv *priv, enum graph_type gtype, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu) { - struct device_node *ep; int ret = 0; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); - if (graph_lnk_is_multi(port)) { + if (is_multi) { int idx; - of_node_get(port); - for (idx = 0;; idx++) { ep = graph_get_next_multi_ep(&port); if (!ep) @@ -532,9 +531,8 @@ static int graph_parse_node(struct asoc_simple_priv *priv, } } else { /* Single CPU / Codec */ - ep = port_to_endpoint(port); + of_node_put(port); ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); } return ret; @@ -591,22 +589,20 @@ static void graph_parse_daifmt(struct device_node *node, } static void graph_link_init(struct asoc_simple_priv *priv, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu_node) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct device_node *ep; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; unsigned int bit_frame = 0; - if (graph_lnk_is_multi(port)) { - of_node_get(port); + if (is_multi) { ep = graph_get_next_multi_ep(&port); port = of_get_parent(ep); - } else { - ep = port_to_endpoint(port); } ports = of_get_parent(port); @@ -642,6 +638,9 @@ static void graph_link_init(struct asoc_simple_priv *priv, dai_link->ops = &graph_ops; if (priv->ops) dai_link->ops = priv->ops; + + of_node_put(port); + of_node_put(ports); } int audio_graph2_link_normal(struct asoc_simple_priv *priv, @@ -650,7 +649,7 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, { struct device_node *cpu_port = lnk; struct device_node *cpu_ep = port_to_endpoint(cpu_port); - struct device_node *codec_port = of_graph_get_remote_port(cpu_ep); + struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; /* @@ -658,20 +657,20 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0); + ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0); if (ret < 0) goto err; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1); + ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1); if (ret < 0) goto err; - graph_link_init(priv, cpu_port, li, 1); + graph_link_init(priv, cpu_ep, li, 1); err: - of_node_put(codec_port); + of_node_put(codec_ep); of_node_put(cpu_ep); return ret; @@ -684,7 +683,6 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, { struct device_node *ep = port_to_endpoint(lnk); struct device_node *rep = of_graph_get_remote_endpoint(ep); - struct device_node *rport = of_graph_get_remote_port(ep); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); int is_cpu = asoc_graph_is_ports0(lnk); @@ -718,7 +716,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 1); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 1); if (ret) goto err; } else { @@ -751,7 +749,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 0); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 0); if (ret < 0) goto err; } @@ -761,11 +759,10 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, rport, li, is_cpu); + graph_link_init(priv, rep, li, is_cpu); err: of_node_put(ep); of_node_put(rep); - of_node_put(rport); return ret; } @@ -777,7 +774,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct device_node *port0, *port1, *ports; - struct device_node *codec0_port, *codec1_port; + struct device_node *codec0_ep, *codec1_ep; struct device_node *ep0, *ep1; u32 val = 0; int ret = -EINVAL; @@ -834,31 +831,31 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); - codec0_port = of_graph_get_remote_port(ep0); - codec1_port = of_graph_get_remote_port(ep1); + codec0_ep = of_graph_get_remote_endpoint(ep0); + codec1_ep = of_graph_get_remote_endpoint(ep1); /* * call Codec first. * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0); + ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0); if (ret < 0) goto err2; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1); + ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1); if (ret < 0) goto err2; - graph_link_init(priv, codec0_port, li, 1); + graph_link_init(priv, codec0_ep, li, 1); err2: of_node_put(ep0); of_node_put(ep1); - of_node_put(codec0_port); - of_node_put(codec1_port); + of_node_put(codec0_ep); + of_node_put(codec1_ep); err1: of_node_put(ports); of_node_put(port0); -- 2.30.2

1 0

CS42L43 headset mic enable/disable not applied when jack insert state changes
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-ucm-conf issue #489 was opened from rfvirgil: Originally the configuration for sof-soundwire/cs42l43*.conf had the headset mic and internal mics on the same ALSA capture device 4. The Headset Mic enable and disable sequences ran as expected when the headset jack was inserted and removed. But SOF expects that device 4 is the internal mics, and headset mic should be on device 1. So we made this change to move the headset mic to capture device 1: https://github.com/alsa-project/alsa-ucm-conf/commit/30989bd0c2aa3f9f4b6f5e… But now the Headset Mic enable and disable sequences do not run when the headset jack state changes. Seen on multiple systems with Pipewire + alsa-ucm + CS42L43 on sof-soundwire. Example from a system that booted with jack unplugged. The jack was then plugged in. The CS42L43 has reported that the headset mic is present: ``` Simple mixer control 'Headset Mic',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] ``` But the enable sequence has not been applied (DP2TX1 and DP2TX2 should have been set to Decimator 1): ``` Simple mixer control 'cs42l43 DP2TX1 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' Simple mixer control 'cs42l43 DP2TX2 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/489 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

snd_pcm_writei() takes to much time to write data
by GitHub issues - edited 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was edited from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_readi() takes to much time to write data
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was opened from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 1/1] ASoC: codecs: nau8824: fix max volume for Speaker output
by Maxim Kochetkov 19 Dec '24

19 Dec '24

There is no audio output if Speaker volume is set above 25. According to datasheet Rev 2.5 maximum allowed value for the Speaker output is 0b11001 (25) 0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2: Left/Right Channel Class-D Driver Gain For DAC Left/Right Input (Step size is 1dB.) 00000 = Mute (DEFAULT) 00001 = 0dB 00002 = 1dB ... 11000 = 23dB 11001 = 24dB So adjust this value in accordance with the datasheet. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/nau8824.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d..5aaf8c496300 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -368,13 +368,13 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), SOC_SINGLE_TLV("Speaker Right DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Right DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), -- 2.45.2

1 0

[PATCH v2] ASoC: amd: yc: Fix the wrong return value
by Venkata Prasad Potturu 17 Dec '24

17 Dec '24

With the current implementation, when ACP driver fails to read ACPI _WOV entry then the DMI overrides code won't invoke, may cause regressions for some BIOS versions. Add a condition check to jump to check the DMI entries incase of ACP driver fail to read ACPI _WOV method. Fixes: 4095cf872084 (ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry) Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: - removed redundant "&& wov_en" check sound/soc/amd/yc/acp6x-mach.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e38c5885dadf..ecf57a6cb7c3 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev) handle = ACPI_HANDLE(pdev->dev.parent); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); - if (!ACPI_FAILURE(ret)) + if (!ACPI_FAILURE(ret)) { wov_en = dmic_status; + if (!wov_en) + return -ENODEV; + } else { + /* Incase of ACPI method read failure then jump to check_dmi_entry */ + goto check_dmi_entry; + } - if (is_dmic_enable && wov_en) + if (is_dmic_enable) platform_set_drvdata(pdev, &acp6x_card); - else - return 0; +check_dmi_entry: /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.25.1

5 5

UCM2: Intel: sof-hda-dsp: Enable Dmic0 pre-processing and SDW speaker post-processing
by GitHub pull_request - edited 16 Dec '24

16 Dec '24

alsa-project/alsa-ucm-conf pull request #481 was edited from singalsu: This patch adds enable of the microphone capture pre-processing features as BootSequence inital values if the controls for them are available. If the user later switches these off, they will remain off. The DRC is dynamic range control that boosts the capture level, and the TDFB (time-domain fixed beamformer) improves capture signa-to-noise ratio for set beam angle direction. The used 0 degrees angle is defined as angle towards notebook computer user. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: amd: ps: Fix for enabling DMIC on acp63 platform via _DSD entry
by Venkata Prasad Potturu 16 Dec '24

16 Dec '24

Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 0386d765f27a ("ASoC: amd: ps: refactor acp device configuration read logic") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index a7583844f5b4..983d7fa13938 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -374,11 +374,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a { struct acpi_device *pdm_dev; const union acpi_object *obj; + acpi_handle handle; + acpi_integer dmic_status; u32 config; bool is_dmic_dev = false; bool is_sdw_dev = false; + bool wov_en, dmic_en; int ret; + /* IF WOV entry not found, enable dmic based on acp-audio-device-type entry*/ + wov_en = true; + dmic_en = false; + config = readl(acp_data->acp63_base + ACP_PIN_CONFIG); switch (config) { case ACP_CONFIG_4: @@ -411,10 +418,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == ACP_DMIC_DEV) - is_dmic_dev = true; + dmic_en = true; } + + handle = ACPI_HANDLE(&pci->dev); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; } + if (dmic_en && wov_en) + is_dmic_dev = true; + if (acp_data->is_sdw_config) { ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR); if (!ret && acp_data->info.link_mask) -- 2.25.1

3 2

[PATCH v7 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 13 Dec '24

13 Dec '24

This function is base on the accelerator implementation for compress API: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v7: - add 'select SND_COMPRESS_OFFLOAD' for fixing compile issue with imx_v6_v7_defconfig changes in v6: - rebase to latest tree, change to use 'MODULE_IMPORT_NS("DMA_BUF")' - Add Acked-by: Vinod Koul for patch 1/6 changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1261 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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[PATCH v6 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 13 Dec '24

13 Dec '24

This function is base on the accelerator implementation for compress API: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v6: - rebase to latest tree, change to use 'MODULE_IMPORT_NS("DMA_BUF")' - Add Acked-by: Vinod Koul for patch 1/6 changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

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