[PATCH 0/4] Support CS42L42 on JSL platform
This series consists four patches. Patch 1 adds dai sequence support for cml/jsl/tgl platforms which is different from the sequence on glk platform. Patch 2 adds max98360a support to the maxim-common module. Patch 3 adds driver data for jsl_cs4242_mx98360a which supports cs42l42 and max98360a running on jsl boards. Patch 4 refactor the sof_rt5682 to use the max98360 code in the maxim-common module.
Brent Lu (4): ASoC: Intel: sof_cs42l42: support JSL DAI link sequence ASoC: Intel: maxim-common: support max98360a ASoC: intel: sof_cs42l42: add support for jsl_cs4242_mx98360a ASoC: Intel: sof_rt5682: code refactor for max98360a
sound/soc/intel/boards/sof_cs42l42.c | 340 ++++++++++++------ sound/soc/intel/boards/sof_maxim_common.c | 17 +- sound/soc/intel/boards/sof_maxim_common.h | 4 +- sound/soc/intel/boards/sof_rt5682.c | 52 +-- .../intel/common/soc-acpi-intel-jsl-match.c | 8 + 5 files changed, 254 insertions(+), 167 deletions(-)
The backend DAI link sequence of GLK platform is different from the sequence of other platforms. We refactor the sof_card_dai_links_create() function to support both style.
GLK: SPK - HP - DMIC - HDMI Other: HP - DMIC - HDMI - SPK
Signed-off-by: Brent Lu brent.lu@intel.com --- sound/soc/intel/boards/sof_cs42l42.c | 318 ++++++++++++++++++--------- 1 file changed, 208 insertions(+), 110 deletions(-)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 8919d3ba3c89..e3171242f612 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -259,133 +259,166 @@ static struct snd_soc_dai_link_component dmic_component[] = { } };
-static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, - int ssp_codec, - int ssp_amp, - int dmic_be_num, - int hdmi_num) +static int create_spk_amp_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_amp) { - struct snd_soc_dai_link_component *idisp_components; - struct snd_soc_dai_link_component *cpus; - struct snd_soc_dai_link *links; - int i, id = 0; - - links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * - sof_audio_card_cs42l42.num_links, GFP_KERNEL); - cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) * - sof_audio_card_cs42l42.num_links, GFP_KERNEL); - if (!links || !cpus) - goto devm_err; + int ret = 0;
/* speaker amp */ - if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) { - links[id].name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d-Codec", ssp_amp); - if (!links[id].name) - goto devm_err; + if (!(sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT)) + return 0;
- links[id].id = id; + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", + ssp_amp); + if (!links[*id].name) { + ret = -ENOMEM; + goto devm_err; + }
- if (sof_cs42l42_quirk & SOF_MAX98357A_SPEAKER_AMP_PRESENT) { - max_98357a_dai_link(&links[id]); - } else { - dev_err(dev, "no amp defined\n"); - goto devm_err; - } + links[*id].id = *id;
- links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].dpcm_playback = 1; - links[id].no_pcm = 1; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_amp); - if (!links[id].cpus->dai_name) - goto devm_err; + if (sof_cs42l42_quirk & SOF_MAX98357A_SPEAKER_AMP_PRESENT) { + max_98357a_dai_link(&links[*id]); + } else { + dev_err(dev, "no amp defined\n"); + ret = -EINVAL; + goto devm_err; + }
- id++; + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + links[*id].dpcm_playback = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", ssp_amp); + if (!links[*id].cpus->dai_name) { + ret = -ENOMEM; + goto devm_err; }
+ (*id)++; + +devm_err: + return ret; +} + +static int create_hp_codec_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int ssp_codec) +{ /* codec SSP */ - links[id].name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d-Codec", ssp_codec); - if (!links[id].name) + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", + ssp_codec); + if (!links[*id].name) goto devm_err;
- links[id].id = id; - links[id].codecs = cs42l42_component; - links[id].num_codecs = ARRAY_SIZE(cs42l42_component); - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].init = sof_cs42l42_init; - links[id].exit = sof_cs42l42_exit; - links[id].ops = &sof_cs42l42_ops; - links[id].dpcm_playback = 1; - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ssp_codec); - if (!links[id].cpus->dai_name) + links[*id].id = *id; + links[*id].codecs = cs42l42_component; + links[*id].num_codecs = ARRAY_SIZE(cs42l42_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + links[*id].init = sof_cs42l42_init; + links[*id].exit = sof_cs42l42_exit; + links[*id].ops = &sof_cs42l42_ops; + links[*id].dpcm_playback = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + + links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ssp_codec); + if (!links[*id].cpus->dai_name) goto devm_err;
- id++; + (*id)++; + + return 0; + +devm_err: + return -ENOMEM; +} + +static int create_dmic_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int dmic_be_num) +{ + int i;
/* dmic */ - if (dmic_be_num > 0) { - /* at least we have dmic01 */ - links[id].name = "dmic01"; - links[id].cpus = &cpus[id]; - links[id].cpus->dai_name = "DMIC01 Pin"; - links[id].init = dmic_init; - if (dmic_be_num > 1) { - /* set up 2 BE links at most */ - links[id + 1].name = "dmic16k"; - links[id + 1].cpus = &cpus[id + 1]; - links[id + 1].cpus->dai_name = "DMIC16k Pin"; - dmic_be_num = 2; - } + if (dmic_be_num <= 0) + return 0; + + /* at least we have dmic01 */ + links[*id].name = "dmic01"; + links[*id].cpus = &cpus[*id]; + links[*id].cpus->dai_name = "DMIC01 Pin"; + links[*id].init = dmic_init; + if (dmic_be_num > 1) { + /* set up 2 BE links at most */ + links[*id + 1].name = "dmic16k"; + links[*id + 1].cpus = &cpus[*id + 1]; + links[*id + 1].cpus->dai_name = "DMIC16k Pin"; + dmic_be_num = 2; }
for (i = 0; i < dmic_be_num; i++) { - links[id].id = id; - links[id].num_cpus = 1; - links[id].codecs = dmic_component; - links[id].num_codecs = ARRAY_SIZE(dmic_component); - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].ignore_suspend = 1; - links[id].dpcm_capture = 1; - links[id].no_pcm = 1; - id++; + links[*id].id = *id; + links[*id].num_cpus = 1; + links[*id].codecs = dmic_component; + links[*id].num_codecs = ARRAY_SIZE(dmic_component); + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + links[*id].ignore_suspend = 1; + links[*id].dpcm_capture = 1; + links[*id].no_pcm = 1; + + (*id)++; }
+ return 0; +} + +static int create_hdmi_dai_links(struct device *dev, + struct snd_soc_dai_link *links, + struct snd_soc_dai_link_component *cpus, + int *id, int hdmi_num) +{ + struct snd_soc_dai_link_component *idisp_components; + int i; + /* HDMI */ - if (hdmi_num > 0) { - idisp_components = devm_kzalloc(dev, - sizeof(struct snd_soc_dai_link_component) * - hdmi_num, GFP_KERNEL); - if (!idisp_components) - goto devm_err; - } + if (hdmi_num <= 0) + return 0; + + idisp_components = devm_kzalloc(dev, + sizeof(struct snd_soc_dai_link_component) * + hdmi_num, GFP_KERNEL); + if (!idisp_components) + goto devm_err; + for (i = 1; i <= hdmi_num; i++) { - links[id].name = devm_kasprintf(dev, GFP_KERNEL, - "iDisp%d", i); - if (!links[id].name) + links[*id].name = devm_kasprintf(dev, GFP_KERNEL, + "iDisp%d", i); + if (!links[*id].name) goto devm_err;
- links[id].id = id; - links[id].cpus = &cpus[id]; - links[id].num_cpus = 1; - links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "iDisp%d Pin", i); - if (!links[id].cpus->dai_name) + links[*id].id = *id; + links[*id].cpus = &cpus[*id]; + links[*id].num_cpus = 1; + links[*id].cpus->dai_name = devm_kasprintf(dev, + GFP_KERNEL, + "iDisp%d Pin", + i); + if (!links[*id].cpus->dai_name) goto devm_err;
idisp_components[i - 1].name = "ehdaudio0D2"; @@ -396,14 +429,79 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (!idisp_components[i - 1].dai_name) goto devm_err;
- links[id].codecs = &idisp_components[i - 1]; - links[id].num_codecs = 1; - links[id].platforms = platform_component; - links[id].num_platforms = ARRAY_SIZE(platform_component); - links[id].init = sof_hdmi_init; - links[id].dpcm_playback = 1; - links[id].no_pcm = 1; - id++; + links[*id].codecs = &idisp_components[i - 1]; + links[*id].num_codecs = 1; + links[*id].platforms = platform_component; + links[*id].num_platforms = ARRAY_SIZE(platform_component); + links[*id].init = sof_hdmi_init; + links[*id].dpcm_playback = 1; + links[*id].no_pcm = 1; + + (*id)++; + } + + return 0; + +devm_err: + return -ENOMEM; +} + +static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, + int ssp_codec, + int ssp_amp, + int dmic_be_num, + int hdmi_num) +{ + struct snd_soc_dai_link_component *cpus; + struct snd_soc_dai_link *links; + int ret, id = 0; + + links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * + sof_audio_card_cs42l42.num_links, GFP_KERNEL); + cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) * + sof_audio_card_cs42l42.num_links, GFP_KERNEL); + if (!links || !cpus) + goto devm_err; + + if (soc_intel_is_glk()) { + /* gemini lake starts from spk link */ + ret = create_spk_amp_dai_links(dev, links, cpus, &id, ssp_amp); + if (ret < 0) { + dev_err(dev, "fail to create spk amp dai links, ret %d\n", + ret); + goto devm_err; + } + } + + ret = create_hp_codec_dai_links(dev, links, cpus, &id, ssp_codec); + if (ret < 0) { + dev_err(dev, "fail to create hp codec dai links, ret %d\n", + ret); + goto devm_err; + } + + ret = create_dmic_dai_links(dev, links, cpus, &id, dmic_be_num); + if (ret < 0) { + dev_err(dev, "fail to create dmic dai links, ret %d\n", + ret); + goto devm_err; + } + + ret = create_hdmi_dai_links(dev, links, cpus, &id, hdmi_num); + if (ret < 0) { + dev_err(dev, "fail to create hdmi dai links, ret %d\n", + ret); + goto devm_err; + } + + if (!soc_intel_is_glk()) { + /* other platforms end with spk link */ + ret = create_spk_amp_dai_links(dev, links, cpus, &id, ssp_amp); + if (ret < 0) { + dev_err(dev, "fail to create spk amp dai links, ret %d\n", + ret); + goto devm_err; + } }
return links;
On 6/5/21 7:40 PM, Brent Lu wrote:
The backend DAI link sequence of GLK platform is different from the sequence of other platforms. We refactor the sof_card_dai_links_create() function to support both style.
GLK: SPK - HP - DMIC - HDMI Other: HP - DMIC - HDMI - SPK
I am really confused here. The dailink sequence is whatever we want it to be. What matters is that the dailink ID matches what is in the topology.
Is this saying that the GLK and JSL topologies did not follow any sort of convention? Can you elaborate more on what is the issue?
Put differently, why can't we fix the topology instead with a reorder of the dailinks?
Signed-off-by: Brent Lu brent.lu@intel.com
sound/soc/intel/boards/sof_cs42l42.c | 318 ++++++++++++++++++--------- 1 file changed, 208 insertions(+), 110 deletions(-)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index 8919d3ba3c89..e3171242f612 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -259,133 +259,166 @@ static struct snd_soc_dai_link_component dmic_component[] = { } };
-static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
int ssp_codec,int ssp_amp,int dmic_be_num,int hdmi_num)+static int create_spk_amp_dai_links(struct device *dev,
struct snd_soc_dai_link *links,struct snd_soc_dai_link_component *cpus,int *id, int ssp_amp)
- struct snd_soc_dai_link_component *idisp_components;
- struct snd_soc_dai_link_component *cpus;
- struct snd_soc_dai_link *links;
- int i, id = 0;
- links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) *
sof_audio_card_cs42l42.num_links, GFP_KERNEL);- cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) *
sof_audio_card_cs42l42.num_links, GFP_KERNEL);- if (!links || !cpus)
goto devm_err;
int ret = 0;
/* speaker amp */
- if (sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT) {
links[id].name = devm_kasprintf(dev, GFP_KERNEL,"SSP%d-Codec", ssp_amp);if (!links[id].name)goto devm_err;
- if (!(sof_cs42l42_quirk & SOF_SPEAKER_AMP_PRESENT))
return 0;
links[id].id = id;
- links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec",
ssp_amp);- if (!links[*id].name) {
ret = -ENOMEM;goto devm_err;- }
if (sof_cs42l42_quirk & SOF_MAX98357A_SPEAKER_AMP_PRESENT) {max_98357a_dai_link(&links[id]);} else {dev_err(dev, "no amp defined\n");goto devm_err;}
- links[*id].id = *id;
links[id].platforms = platform_component;links[id].num_platforms = ARRAY_SIZE(platform_component);links[id].dpcm_playback = 1;links[id].no_pcm = 1;links[id].cpus = &cpus[id];links[id].num_cpus = 1;links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,"SSP%d Pin",ssp_amp);if (!links[id].cpus->dai_name)goto devm_err;
- if (sof_cs42l42_quirk & SOF_MAX98357A_SPEAKER_AMP_PRESENT) {
max_98357a_dai_link(&links[*id]);- } else {
dev_err(dev, "no amp defined\n");ret = -EINVAL;goto devm_err;- }
id++;
links[*id].platforms = platform_component;
links[*id].num_platforms = ARRAY_SIZE(platform_component);
links[*id].dpcm_playback = 1;
links[*id].no_pcm = 1;
links[*id].cpus = &cpus[*id];
links[*id].num_cpus = 1;
links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
"SSP%d Pin", ssp_amp);if (!links[*id].cpus->dai_name) {
ret = -ENOMEM;goto devm_err;}
(*id)++;
+devm_err:
- return ret;
+}
+static int create_hp_codec_dai_links(struct device *dev,
struct snd_soc_dai_link *links,struct snd_soc_dai_link_component *cpus,int *id, int ssp_codec)+{ /* codec SSP */
- links[id].name = devm_kasprintf(dev, GFP_KERNEL,
"SSP%d-Codec", ssp_codec);- if (!links[id].name)
- links[*id].name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec",
ssp_codec);- if (!links[*id].name) goto devm_err;
- links[id].id = id;
- links[id].codecs = cs42l42_component;
- links[id].num_codecs = ARRAY_SIZE(cs42l42_component);
- links[id].platforms = platform_component;
- links[id].num_platforms = ARRAY_SIZE(platform_component);
- links[id].init = sof_cs42l42_init;
- links[id].exit = sof_cs42l42_exit;
- links[id].ops = &sof_cs42l42_ops;
- links[id].dpcm_playback = 1;
- links[id].dpcm_capture = 1;
- links[id].no_pcm = 1;
- links[id].cpus = &cpus[id];
- links[id].num_cpus = 1;
- links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
"SSP%d Pin",ssp_codec);- if (!links[id].cpus->dai_name)
- links[*id].id = *id;
- links[*id].codecs = cs42l42_component;
- links[*id].num_codecs = ARRAY_SIZE(cs42l42_component);
- links[*id].platforms = platform_component;
- links[*id].num_platforms = ARRAY_SIZE(platform_component);
- links[*id].init = sof_cs42l42_init;
- links[*id].exit = sof_cs42l42_exit;
- links[*id].ops = &sof_cs42l42_ops;
- links[*id].dpcm_playback = 1;
- links[*id].dpcm_capture = 1;
- links[*id].no_pcm = 1;
- links[*id].cpus = &cpus[*id];
- links[*id].num_cpus = 1;
- links[*id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,
"SSP%d Pin",ssp_codec);- if (!links[*id].cpus->dai_name) goto devm_err;
- id++;
- (*id)++;
- return 0;
+devm_err:
- return -ENOMEM;
+}
+static int create_dmic_dai_links(struct device *dev,
struct snd_soc_dai_link *links,struct snd_soc_dai_link_component *cpus,int *id, int dmic_be_num)+{
int i;
/* dmic */
- if (dmic_be_num > 0) {
/* at least we have dmic01 */links[id].name = "dmic01";links[id].cpus = &cpus[id];links[id].cpus->dai_name = "DMIC01 Pin";links[id].init = dmic_init;if (dmic_be_num > 1) {/* set up 2 BE links at most */links[id + 1].name = "dmic16k";links[id + 1].cpus = &cpus[id + 1];links[id + 1].cpus->dai_name = "DMIC16k Pin";dmic_be_num = 2;}
if (dmic_be_num <= 0)
return 0;/* at least we have dmic01 */
links[*id].name = "dmic01";
links[*id].cpus = &cpus[*id];
links[*id].cpus->dai_name = "DMIC01 Pin";
links[*id].init = dmic_init;
if (dmic_be_num > 1) {
/* set up 2 BE links at most */links[*id + 1].name = "dmic16k";links[*id + 1].cpus = &cpus[*id + 1];links[*id + 1].cpus->dai_name = "DMIC16k Pin";dmic_be_num = 2;}
for (i = 0; i < dmic_be_num; i++) {
links[id].id = id;links[id].num_cpus = 1;links[id].codecs = dmic_component;links[id].num_codecs = ARRAY_SIZE(dmic_component);links[id].platforms = platform_component;links[id].num_platforms = ARRAY_SIZE(platform_component);links[id].ignore_suspend = 1;links[id].dpcm_capture = 1;links[id].no_pcm = 1;id++;
links[*id].id = *id;links[*id].num_cpus = 1;links[*id].codecs = dmic_component;links[*id].num_codecs = ARRAY_SIZE(dmic_component);links[*id].platforms = platform_component;links[*id].num_platforms = ARRAY_SIZE(platform_component);links[*id].ignore_suspend = 1;links[*id].dpcm_capture = 1;links[*id].no_pcm = 1;(*id)++;}
return 0;
+}
+static int create_hdmi_dai_links(struct device *dev,
struct snd_soc_dai_link *links,struct snd_soc_dai_link_component *cpus,int *id, int hdmi_num)+{
- struct snd_soc_dai_link_component *idisp_components;
- int i;
- /* HDMI */
- if (hdmi_num > 0) {
idisp_components = devm_kzalloc(dev,sizeof(struct snd_soc_dai_link_component) *hdmi_num, GFP_KERNEL);if (!idisp_components)goto devm_err;- }
- if (hdmi_num <= 0)
return 0;- idisp_components = devm_kzalloc(dev,
sizeof(struct snd_soc_dai_link_component) *hdmi_num, GFP_KERNEL);- if (!idisp_components)
goto devm_err;- for (i = 1; i <= hdmi_num; i++) {
links[id].name = devm_kasprintf(dev, GFP_KERNEL,"iDisp%d", i);if (!links[id].name)
links[*id].name = devm_kasprintf(dev, GFP_KERNEL,"iDisp%d", i);if (!links[*id].name) goto devm_err;
links[id].id = id;links[id].cpus = &cpus[id];links[id].num_cpus = 1;links[id].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL,"iDisp%d Pin", i);if (!links[id].cpus->dai_name)
links[*id].id = *id;links[*id].cpus = &cpus[*id];links[*id].num_cpus = 1;links[*id].cpus->dai_name = devm_kasprintf(dev,GFP_KERNEL,"iDisp%d Pin",i);if (!links[*id].cpus->dai_name) goto devm_err;idisp_components[i - 1].name = "ehdaudio0D2";
@@ -396,14 +429,79 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (!idisp_components[i - 1].dai_name) goto devm_err;
links[id].codecs = &idisp_components[i - 1];links[id].num_codecs = 1;links[id].platforms = platform_component;links[id].num_platforms = ARRAY_SIZE(platform_component);links[id].init = sof_hdmi_init;links[id].dpcm_playback = 1;links[id].no_pcm = 1;id++;
links[*id].codecs = &idisp_components[i - 1];links[*id].num_codecs = 1;links[*id].platforms = platform_component;links[*id].num_platforms = ARRAY_SIZE(platform_component);links[*id].init = sof_hdmi_init;links[*id].dpcm_playback = 1;links[*id].no_pcm = 1;(*id)++;- }
- return 0;
+devm_err:
- return -ENOMEM;
+}
+static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
int ssp_codec,int ssp_amp,int dmic_be_num,int hdmi_num)+{
struct snd_soc_dai_link_component *cpus;
struct snd_soc_dai_link *links;
int ret, id = 0;
links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) *
sof_audio_card_cs42l42.num_links, GFP_KERNEL);cpus = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link_component) *
sof_audio_card_cs42l42.num_links, GFP_KERNEL);if (!links || !cpus)
goto devm_err;if (soc_intel_is_glk()) {
/* gemini lake starts from spk link */ret = create_spk_amp_dai_links(dev, links, cpus, &id, ssp_amp);if (ret < 0) {dev_err(dev, "fail to create spk amp dai links, ret %d\n",ret);goto devm_err;}}
ret = create_hp_codec_dai_links(dev, links, cpus, &id, ssp_codec);
if (ret < 0) {
dev_err(dev, "fail to create hp codec dai links, ret %d\n",ret);goto devm_err;}
ret = create_dmic_dai_links(dev, links, cpus, &id, dmic_be_num);
if (ret < 0) {
dev_err(dev, "fail to create dmic dai links, ret %d\n",ret);goto devm_err;}
ret = create_hdmi_dai_links(dev, links, cpus, &id, hdmi_num);
if (ret < 0) {
dev_err(dev, "fail to create hdmi dai links, ret %d\n",ret);goto devm_err;}
if (!soc_intel_is_glk()) {
/* other platforms end with spk link */ret = create_spk_amp_dai_links(dev, links, cpus, &id, ssp_amp);if (ret < 0) {dev_err(dev, "fail to create spk amp dai links, ret %d\n",ret);goto devm_err;}}
return links;
On 6/5/21 7:40 PM, Brent Lu wrote:
The backend DAI link sequence of GLK platform is different from the sequence of other platforms. We refactor the sof_card_dai_links_create() function to support both style.
GLK: SPK - HP - DMIC - HDMI Other: HP - DMIC - HDMI - SPK
I am really confused here. The dailink sequence is whatever we want it to be. What matters is that the dailink ID matches what is in the topology.
Is this saying that the GLK and JSL topologies did not follow any sort of convention? Can you elaborate more on what is the issue?
Put differently, why can't we fix the topology instead with a reorder of the dailinks?
snd_soc_find_dai_link() checked both dai link name and id when matching topology and machine driver. Soundcard registration would fail if their id doesn't match.
Cs42l42 is sharing topology with DA7219's topology source file sof-glk-da7219.m4 on GLK platform. The configuration is:
dai link id 0 is for spk #SSP 1 (ID: 0) with 19.2 MHz mclk with MCLK_ID 1 (unused), 1.536 MHz blck DAI_CONFIG(SSP, 1, 0, SSP1-Codec,
dai link id 1 is for headphone #SSP 2 (ID: 1) with 19.2 MHz mclk with MCLK_ID 1, 1.92 MHz bclk DAI_CONFIG(SSP, 2, 1, SSP2-Codec,
dai link id 2 is for dmic DAI_CONFIG(DMIC, 0, 2, dmic01,
dai link id 3/4/5 is for hdmi DAI_CONFIG(HDA, 3, 3, iDisp1, DAI_CONFIG(HDA, 4, 4, iDisp2, DAI_CONFIG(HDA, 5, 5, iDisp3,
When on JSL, we plan to share topology with rt5682 which has different dai link sequence: sof-jsl-rt5682.m4:
dai link id 0 is for headphone DAI_CONFIG(SSP, 0, 0, SSP0-Codec,
dai link id 6 is for spk # SSP 1 (ID: 6) DAI_CONFIG(SSP, SPK_INDEX, 6, SPK_NAME, SET_SSP_CONFIG)
dai link id 3/4/5 is for hdmi # 4 HDMI/DP outputs (ID: 3,4,5) DAI_CONFIG(HDA, 0, 3, iDisp1, DAI_CONFIG(HDA, 1, 4, iDisp2, DAI_CONFIG(HDA, 2, 5, iDisp3,
I'm not sure if there is convention about the sequence to follow?
Regards, Brent
Cs42l42 is sharing topology with DA7219's topology source file sof-glk-da7219.m4 on GLK platform. The configuration is:
dai link id 0 is for spk #SSP 1 (ID: 0) with 19.2 MHz mclk with MCLK_ID 1 (unused), 1.536 MHz blck DAI_CONFIG(SSP, 1, 0, SSP1-Codec,
dai link id 1 is for headphone #SSP 2 (ID: 1) with 19.2 MHz mclk with MCLK_ID 1, 1.92 MHz bclk DAI_CONFIG(SSP, 2, 1, SSP2-Codec,
dai link id 2 is for dmic DAI_CONFIG(DMIC, 0, 2, dmic01,
dai link id 3/4/5 is for hdmi DAI_CONFIG(HDA, 3, 3, iDisp1, DAI_CONFIG(HDA, 4, 4, iDisp2, DAI_CONFIG(HDA, 5, 5, iDisp3,
When on JSL, we plan to share topology with rt5682 which has different dai link sequence: sof-jsl-rt5682.m4:
dai link id 0 is for headphone DAI_CONFIG(SSP, 0, 0, SSP0-Codec,
dai link id 6 is for spk # SSP 1 (ID: 6) DAI_CONFIG(SSP, SPK_INDEX, 6, SPK_NAME, SET_SSP_CONFIG)
dai link id 3/4/5 is for hdmi # 4 HDMI/DP outputs (ID: 3,4,5) DAI_CONFIG(HDA, 0, 3, iDisp1, DAI_CONFIG(HDA, 1, 4, iDisp2, DAI_CONFIG(HDA, 2, 5, iDisp3,
I'm not sure if there is convention about the sequence to follow?
ok, now I get what you are trying to do.
Unfortunately there are no conventions so far, and since we have to be backwards-compatible with topology files already released we will need to deal with the different configurations in this machine driver, you're right about this.
The code you suggested is fine, but we can future-proof it a bit.
Instead of assuming any order depending on GLK or !GLK, we can add a BE 'base' for headphone, amp, DMIC and DMIC each (represented as a constant structure) and point to different configurations depending on a quirk. That way we can deal with other permutations such as HP - SPK - HDMI - DMIC
Move max98360a code to this common module so it could be shared between multiple SOF machine drivers. MAX98357A and MAX98360A are sharing same codec driver so here we also share some function and structures.
Signed-off-by: Brent Lu brent.lu@intel.com --- sound/soc/intel/boards/sof_maxim_common.c | 17 ++++++++++++++++- sound/soc/intel/boards/sof_maxim_common.h | 4 +++- 2 files changed, 19 insertions(+), 2 deletions(-)
diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e9c52f8b6428..e66dfe666915 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -134,7 +134,7 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON);
/* - * Maxim MAX98357A + * Maxim MAX98357A/MAX98360A */ static const struct snd_kcontrol_new max_98357a_kcontrols[] = { SOC_DAPM_PIN_SWITCH("Spk"), @@ -156,6 +156,13 @@ static struct snd_soc_dai_link_component max_98357a_components[] = { } };
+static struct snd_soc_dai_link_component max_98360a_components[] = { + { + .name = MAX_98360A_DEV0_NAME, + .dai_name = MAX_98357A_CODEC_DAI, + } +}; + static int max_98357a_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -193,5 +200,13 @@ void max_98357a_dai_link(struct snd_soc_dai_link *link) } EXPORT_SYMBOL_NS(max_98357a_dai_link, SND_SOC_INTEL_SOF_MAXIM_COMMON);
+void max_98360a_dai_link(struct snd_soc_dai_link *link) +{ + link->codecs = max_98360a_components; + link->num_codecs = ARRAY_SIZE(max_98360a_components); + link->init = max_98357a_init; +} +EXPORT_SYMBOL_NS(max_98360a_dai_link, SND_SOC_INTEL_SOF_MAXIM_COMMON); + MODULE_DESCRIPTION("ASoC Intel SOF Maxim helpers"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2674f1e373ef..3ff5e8fec4de 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -25,11 +25,13 @@ void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd);
/* - * Maxim MAX98357A + * Maxim MAX98357A/MAX98360A */ #define MAX_98357A_CODEC_DAI "HiFi" #define MAX_98357A_DEV0_NAME "MX98357A:00" +#define MAX_98360A_DEV0_NAME "MX98360A:00"
void max_98357a_dai_link(struct snd_soc_dai_link *link); +void max_98360a_dai_link(struct snd_soc_dai_link *link);
#endif /* __SOF_MAXIM_COMMON_H */
This patch adds driver data for jsl_cs4242_mx98360a which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on JSL platform. DAI format is leveraged from sof_rt5682 machine driver to reuse the topology.
Also use module device table to replace module alias.
Signed-off-by: Brent Lu brent.lu@intel.com --- sound/soc/intel/boards/sof_cs42l42.c | 22 +++++++++++++++---- .../intel/common/soc-acpi-intel-jsl-match.c | 8 +++++++ 2 files changed, 26 insertions(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index e3171242f612..d712cfb91fd1 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -36,7 +36,9 @@ #define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7)) #define SOF_CS42L42_NUM_HDMIDEV(quirk) \ (((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(10) +#define SOF_CS42L42_BCLK_2400000 BIT(10) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(11) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(12)
/* Default: SSP2 */ static unsigned long sof_cs42l42_quirk = SOF_CS42L42_SSP_CODEC(2); @@ -122,7 +124,10 @@ static int sof_cs42l42_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int clk_freq, ret;
- clk_freq = 3072000; /* BCLK freq */ + if (sof_cs42l42_quirk & SOF_CS42L42_BCLK_2400000) + clk_freq = 2400000; /* BCLK freq */ + else + clk_freq = 3072000; /* BCLK freq */
/* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, @@ -281,6 +286,8 @@ static int create_spk_amp_dai_links(struct device *dev,
if (sof_cs42l42_quirk & SOF_MAX98357A_SPEAKER_AMP_PRESENT) { max_98357a_dai_link(&links[*id]); + } else if (sof_cs42l42_quirk & SOF_MAX98360A_SPEAKER_AMP_PRESENT) { + max_98360a_dai_link(&links[*id]); } else { dev_err(dev, "no amp defined\n"); ret = -EINVAL; @@ -584,8 +591,17 @@ static const struct platform_device_id board_ids[] = { SOF_MAX98357A_SPEAKER_AMP_PRESENT | SOF_CS42L42_SSP_AMP(1)), }, + { + .name = "jsl_cs4242_mx98360a", + .driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_CS42L42_SSP_AMP(1)) | + SOF_CS42L42_BCLK_2400000, + }, { } }; +MODULE_DEVICE_TABLE(platform, board_ids);
static struct platform_driver sof_audio = { .probe = sof_audio_probe, @@ -601,7 +617,5 @@ module_platform_driver(sof_audio) MODULE_DESCRIPTION("SOF Audio Machine driver for CS42L42"); MODULE_AUTHOR("Brent Lu brent.lu@intel.com"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:sof_cs42l42"); -MODULE_ALIAS("platform:glk_cs4242_max98357a"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); 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 73fe4f89a82d..8e86476d48de 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -73,6 +73,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .id = "10134242", + .drv_name = "jsl_cs4242_mx98360a", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &mx98360a_spk, + .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines);
On 6/5/21 7:41 PM, Brent Lu wrote:
This patch adds driver data for jsl_cs4242_mx98360a which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on JSL platform. DAI format is leveraged from sof_rt5682 machine driver to reuse the topology.
This also looks like we have two topologies configuring the same DAIs differently on different platforms.
Why can't we pick one configuration that would work in all cases?
Also use module device table to replace module alias.
Humm, this looks like a missing dependency, I modified this a while ago.
Signed-off-by: Brent Lu brent.lu@intel.com
sound/soc/intel/boards/sof_cs42l42.c | 22 +++++++++++++++---- .../intel/common/soc-acpi-intel-jsl-match.c | 8 +++++++ 2 files changed, 26 insertions(+), 4 deletions(-)
diff --git a/sound/soc/intel/boards/sof_cs42l42.c b/sound/soc/intel/boards/sof_cs42l42.c index e3171242f612..d712cfb91fd1 100644 --- a/sound/soc/intel/boards/sof_cs42l42.c +++ b/sound/soc/intel/boards/sof_cs42l42.c @@ -36,7 +36,9 @@ #define SOF_CS42L42_NUM_HDMIDEV_MASK (GENMASK(9, 7)) #define SOF_CS42L42_NUM_HDMIDEV(quirk) \ (((quirk) << SOF_CS42L42_NUM_HDMIDEV_SHIFT) & SOF_CS42L42_NUM_HDMIDEV_MASK) -#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(10) +#define SOF_CS42L42_BCLK_2400000 BIT(10) +#define SOF_MAX98357A_SPEAKER_AMP_PRESENT BIT(11) +#define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(12)
/* Default: SSP2 */ static unsigned long sof_cs42l42_quirk = SOF_CS42L42_SSP_CODEC(2); @@ -122,7 +124,10 @@ static int sof_cs42l42_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); int clk_freq, ret;
- clk_freq = 3072000; /* BCLK freq */
if (sof_cs42l42_quirk & SOF_CS42L42_BCLK_2400000)
clk_freq = 2400000; /* BCLK freq */else
clk_freq = 3072000; /* BCLK freq *//* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, 0,
@@ -281,6 +286,8 @@ static int create_spk_amp_dai_links(struct device *dev,
if (sof_cs42l42_quirk & SOF_MAX98357A_SPEAKER_AMP_PRESENT) { max_98357a_dai_link(&links[*id]);
- } else if (sof_cs42l42_quirk & SOF_MAX98360A_SPEAKER_AMP_PRESENT) {
max_98360a_dai_link(&links[*id]);@@ -584,8 +591,17 @@ static const struct platform_device_id board_ids[] = { SOF_MAX98357A_SPEAKER_AMP_PRESENT | SOF_CS42L42_SSP_AMP(1)), },
- {
.name = "jsl_cs4242_mx98360a",.driver_data = (kernel_ulong_t)(SOF_CS42L42_SSP_CODEC(0) |SOF_SPEAKER_AMP_PRESENT |SOF_MAX98360A_SPEAKER_AMP_PRESENT |SOF_CS42L42_SSP_AMP(1)) |SOF_CS42L42_BCLK_2400000,- }, { } };
+MODULE_DEVICE_TABLE(platform, board_ids);
static struct platform_driver sof_audio = { .probe = sof_audio_probe, @@ -601,7 +617,5 @@ module_platform_driver(sof_audio) MODULE_DESCRIPTION("SOF Audio Machine driver for CS42L42"); MODULE_AUTHOR("Brent Lu brent.lu@intel.com"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:sof_cs42l42"); -MODULE_ALIAS("platform:glk_cs4242_max98357a"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); 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 73fe4f89a82d..8e86476d48de 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -73,6 +73,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", },
- {
.id = "10134242",.drv_name = "jsl_cs4242_mx98360a",.sof_fw_filename = "sof-jsl.ri",.machine_quirk = snd_soc_acpi_codec_list,.quirk_data = &mx98360a_spk,.sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg",- }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines);
On 6/5/21 7:41 PM, Brent Lu wrote:
This patch adds driver data for jsl_cs4242_mx98360a which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on JSL platform. DAI format is leveraged from sof_rt5682 machine driver to reuse the topology.
This also looks like we have two topologies configuring the same DAIs differently on different platforms.
Why can't we pick one configuration that would work in all cases?
The comment just say we are reusing rt5685's sof-jsl-rt5682-mx98360a.tplg. This patch does not care about the dai sequence. Maybe I should reword the commit log.
Regards, Brent
Also use module device table to replace module alias.
Humm, this looks like a missing dependency, I modified this a while ago.
On 6/7/21 9:29 AM, Lu, Brent wrote:
On 6/5/21 7:41 PM, Brent Lu wrote:
This patch adds driver data for jsl_cs4242_mx98360a which supports two max98360a speaker amplifiers on SSP1 and cs42l42 headphone codec on SSP0 running on JSL platform. DAI format is leveraged from sof_rt5682 machine driver to reuse the topology.
This also looks like we have two topologies configuring the same DAIs differently on different platforms.
Why can't we pick one configuration that would work in all cases?
The comment just say we are reusing rt5685's sof-jsl-rt5682-mx98360a.tplg. This patch does not care about the dai sequence. Maybe I should reword the commit log.
I was referring to the bclk frequency, one case uses 2.4 and the other 3.072MHz.
This also looks like we have two topologies configuring the same DAIs differently on different platforms.
Why can't we pick one configuration that would work in all cases?
The comment just say we are reusing rt5685's sof-jsl-rt5682-mx98360a.tplg. This patch does not care about the dai sequence. Maybe I should reword the commit log.
I was referring to the bclk frequency, one case uses 2.4 and the other 3.072MHz.
The 2.4MHz setting isn't ready when we enabled this codec so we selected 3.072MHz. Since we are updating topology for PLL issue soon, we can change bclk frequency to 2.4MHz as well. How do you think?
Regards, Brent
On 6/7/21 11:28 AM, Lu, Brent wrote:
This also looks like we have two topologies configuring the same DAIs differently on different platforms.
Why can't we pick one configuration that would work in all cases?
The comment just say we are reusing rt5685's sof-jsl-rt5682-mx98360a.tplg. This patch does not care about the dai sequence. Maybe I should reword the commit log.
I was referring to the bclk frequency, one case uses 2.4 and the other 3.072MHz.
The 2.4MHz setting isn't ready when we enabled this codec so we selected 3.072MHz. Since we are updating topology for PLL issue soon, we can change bclk frequency to 2.4MHz as well. How do you think?
The 3.072MHz clock will require the 24.576MHz PLL to be on on the SOC/PCH. If you can use 2.4 MHz without any loss of quality and the codec can deal with 25 bit slots with 24-bit data it's better power-wise.
We try to use 64.fs only when it's absolutely mandatory, e.g. if the codec or amplifier doesn't support the 25/24 configuration. IIRC this was the case with TI PCM512x and Maxim amps.
We've also used the 3.072 MHz bit clock when there are constraints on the clock sources and selectors. This isn't the case on GLK but the SOF commit 0a97c1a92f2d93bd4d45bc99d61e362cd214748c clarified the clock selection for newer platforms, including JSL. In the end we may be forced to use the 3.072 MHz PLL, you'd need to look at the various topologies used with this machine driver.
Refactor the machine driver by using the common code in maxim-common module to support max98360a.
Signed-off-by: Brent Lu brent.lu@intel.com --- sound/soc/intel/boards/sof_rt5682.c | 52 +---------------------------- 1 file changed, 1 insertion(+), 51 deletions(-)
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 3e69feaf052b..910c054b0b42 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -456,10 +456,6 @@ static const struct snd_kcontrol_new sof_controls[] = {
};
-static const struct snd_kcontrol_new speaker_controls[] = { - SOC_DAPM_PIN_SWITCH("Spk"), -}; - static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -467,10 +463,6 @@ static const struct snd_soc_dapm_widget sof_widgets[] = { SND_SOC_DAPM_SPK("Right Spk", NULL), };
-static const struct snd_soc_dapm_widget speaker_widgets[] = { - SND_SOC_DAPM_SPK("Spk", NULL), -}; - static const struct snd_soc_dapm_widget dmic_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), }; @@ -484,11 +476,6 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, };
-static const struct snd_soc_dapm_route speaker_map[] = { - /* speaker */ - { "Spk", NULL, "Speaker" }, -}; - static const struct snd_soc_dapm_route speaker_map_lr[] = { { "Left Spk", NULL, "Left SPO" }, { "Right Spk", NULL, "Right SPO" }, @@ -505,34 +492,6 @@ static int speaker_codec_init_lr(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(speaker_map_lr)); }
-static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - int ret; - - ret = snd_soc_dapm_new_controls(&card->dapm, speaker_widgets, - ARRAY_SIZE(speaker_widgets)); - if (ret) { - dev_err(rtd->dev, "unable to add dapm controls, ret %d\n", ret); - /* Don't need to add routes if widget addition failed */ - return ret; - } - - ret = snd_soc_add_card_controls(card, speaker_controls, - ARRAY_SIZE(speaker_controls)); - if (ret) { - dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); - return ret; - } - - ret = snd_soc_dapm_add_routes(&card->dapm, speaker_map, - ARRAY_SIZE(speaker_map)); - - if (ret) - dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); - return ret; -} - static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -594,13 +553,6 @@ static struct snd_soc_dai_link_component dmic_component[] = { } };
-static struct snd_soc_dai_link_component max98360a_component[] = { - { - .name = "MX98360A:00", - .dai_name = "HiFi", - } -}; - static struct snd_soc_dai_link_component rt1015_components[] = { { .name = "i2c-10EC1015:00", @@ -775,9 +727,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].dpcm_capture = 1; } else if (sof_rt5682_quirk & SOF_MAX98360A_SPEAKER_AMP_PRESENT) { - links[id].codecs = max98360a_component; - links[id].num_codecs = ARRAY_SIZE(max98360a_component); - links[id].init = speaker_codec_init; + max_98360a_dai_link(&links[id]); } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]);
participants (3)
-
Brent Lu -
Lu, Brent
-
Pierre-Louis Bossart