December 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: simple-card: Don't create separate link when platform is present
by Daniel Baluta 10 Dec '19

10 Dec '19

In normal sound case all DAIs are detected as CPU-Codec. simple_dai_link_of supports the presence of a platform but it counts it as a CPU DAI resulting in the creation of an extra link. Adding a platform property to a link description like: simple-audio-card,dai-link { cpu { sound-dai = <&sai1>; }; plat { sound-dai = <&dsp>; }; codec { sound-dai = <&wm8960>; } will result in the creation of two links: * sai1 <-> wm8960 * dsp <-> wm8960 which is obviously not what we want. We just want one single link with: * sai1 <-> wm8960 (and platform set to dsp). Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/generic/simple-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 10b82bf043d1..55e9f8800b3e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -371,6 +371,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, do { struct asoc_simple_data adata; struct device_node *codec; + struct device_node *plat; struct device_node *np; int num = of_get_child_count(node); @@ -381,6 +382,9 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, ret = -ENODEV; goto error; } + /* get platform */ + plat = of_get_child_by_name(node, is_top ? + PREFIX "plat" : "plat"); /* get convert-xxx property */ memset(&adata, 0, sizeof(adata)); @@ -389,6 +393,8 @@ static int simple_for_each_link(struct asoc_simple_priv *priv, /* loop for all CPU/Codec node */ for_each_child_of_node(node, np) { + if (plat == np) + continue; /* * It is DPCM * if it has many CPUs, -- 2.17.1

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[alsa-devel] [PATCH 0/8] ASoC: SOF: improvements for v5.6
by Pierre-Louis Bossart 10 Dec '19

10 Dec '19

Mix of new interfaces for firmware functionality (channel map, ASRC), corrections to loader code, BYT/CHT improvements w/ BYT-CR support, HDaudio codec mask and CometLake machine driver detection. Amery Song (1): ASoC: Intel: common: work-around incorrect ACPI HID for CML boards Curtis Malainey (1): ASoC: SOF: Intel: split cht and byt debug window sizes Kai Vehmanen (1): ASoC: SOF: Intel: add codec_mask module parameter Karol Trzcinski (2): ASoC: SOF: loader: snd_sof_fw_parse_ext_data log warning on unknown header ASoC: SOF: loader: fix snd_sof_fw_parse_ext_data Pierre-Louis Bossart (1): ASoC: SOF: Intel: byt: fixup topology filename for BYT-CR Seppo Ingalsuo (1): ASoC: SOF: Add asynchronous sample rate converter topology support Slawomir Blauciak (1): ASoC: SOF: ipc: channel map structures include/sound/sof/channel_map.h | 61 +++++++++++++++ include/sound/sof/topology.h | 27 +++++++ include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 6 ++ .../intel/common/soc-acpi-intel-cml-match.c | 41 ++++++---- sound/soc/sof/intel/byt.c | 54 ++++++++++++- sound/soc/sof/intel/hda-ctrl.c | 13 ++++ sound/soc/sof/loader.c | 7 +- sound/soc/sof/topology.c | 78 +++++++++++++++++++ 9 files changed, 265 insertions(+), 24 deletions(-) create mode 100644 include/sound/sof/channel_map.h -- 2.20.1

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[alsa-devel] [PATCH v2 00/15][resend] ASoC: soc-core cleanup step6
by Kuninori Morimoto 10 Dec '19

10 Dec '19

Hi Mark These are resend of v2 of soc-core cleanup step6. These doesn't add new features. Kuninori Morimoto (15): ASoC: soc-core: remove dai_link_list ASoC: soc-core: remove snd_soc_disconnect_sync() ASoC: soc-core: remove snd_soc_get_dai_substream() ASoC: soc-core: move snd_soc_get_pcm_runtime() ASoC: soc-core: find rtd via dai_link pointer at snd_soc_get_pcm_runtime() ASoC: soc-core: move snd_soc_find_dai_link() ASoC: soc-core: rename snd_soc_add_dai_link() to snd_soc_add_pcm_runtime() ASoC: soc-core: rename snd_soc_remove_dai_link() to snd_soc_remove_pcm_runtime() ASoC: soc-core: move soc_link_dai_pcm_new() ASoC: soc-core: rename soc_link_dai_pcm_new() to soc_dai_pcm_new() ASoC: soc-core: move soc_link_init() ASoC: soc-core: add missing return value check for soc_link_init() ASoC: soc-core: rename soc_link_init() to soc_init_pcm_runtime() ASoC: soc-core: soc_set_name_prefix(): tidyup loop condition ASoC: soc-core: soc_set_name_prefix(): get component device_node at out of loop include/sound/soc.h | 22 +-- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/samsung/bells.c | 12 +- sound/soc/samsung/littlemill.c | 10 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/speyside.c | 4 +- sound/soc/samsung/tm2_wm5110.c | 6 +- sound/soc/samsung/tobermory.c | 6 +- sound/soc/sh/rcar/core.c | 2 - sound/soc/soc-core.c | 339 ++++++++++++++++------------------------- sound/soc/soc-topology.c | 47 +++++- sound/soc/tegra/tegra_wm8903.c | 2 +- 13 files changed, 204 insertions(+), 252 deletions(-) -- 2.7.4

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[alsa-devel] broonie-sound/for-next bisection: boot on rk3399-gru-kevin
by kernelci.org bot 10 Dec '19

10 Dec '19

* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * This automated bisection report was sent to you on the basis * * that you may be involved with the breaking commit it has * * found. No manual investigation has been done to verify it, * * and the root cause of the problem may be somewhere else. * * * * If you do send a fix, please include this trailer: * * Reported-by: "kernelci.org bot" <bot(a)kernelci.org> * * * * Hope this helps! * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * broonie-sound/for-next bisection: boot on rk3399-gru-kevin Summary: Start: 3778790e1d13 Merge branch 'asoc-5.6' into asoc-next Details: https://kernelci.org/boot/id/5deef68acb72c66093960f08 Plain log: https://storage.kernelci.org//broonie-sound/for-next/v5.5-rc1-46-g3778790e1… HTML log: https://storage.kernelci.org//broonie-sound/for-next/v5.5-rc1-46-g3778790e1… Result: abe3b6727b65 ASoC: cs42l51: add dac mux widget in codec routes Checks: revert: PASS verify: PASS Parameters: Tree: broonie-sound URL: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Branch: for-next Target: rk3399-gru-kevin CPU arch: arm64 Lab: lab-collabora Compiler: gcc-8 Config: defconfig Test suite: boot Breaking commit found: ------------------------------------------------------------------------------- commit abe3b6727b653307c27870a2d4ecbf9de4e914a5 Author: Olivier Moysan <olivier.moysan(a)st.com> Date: Tue Dec 3 15:16:27 2019 +0100 ASoC: cs42l51: add dac mux widget in codec routes Add "DAC mux" DAPM widget in CS42l51 audio codec routes, to support DAC mux control and to remove error trace "DAC Mux has no paths" at widget creation. Note: ADC path of DAC mux is not routed in this patch. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Link: https://lore.kernel.org/r/20191203141627.29471-1-olivier.moysan@st.com Signed-off-by: Mark Brown <broonie(a)kernel.org> diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 55408c8fcb4e..e47758e4fb36 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -214,12 +214,10 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = { SND_SOC_DAPM_ADC_E("Right ADC", "Right HiFi Capture", CS42L51_POWER_CTL1, 2, 1, cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), - SND_SOC_DAPM_DAC_E("Left DAC", "Left HiFi Playback", - CS42L51_POWER_CTL1, 5, 1, - cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), - SND_SOC_DAPM_DAC_E("Right DAC", "Right HiFi Playback", - CS42L51_POWER_CTL1, 6, 1, - cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Left DAC", NULL, CS42L51_POWER_CTL1, 5, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), + SND_SOC_DAPM_DAC_E("Right DAC", NULL, CS42L51_POWER_CTL1, 6, 1, + cs42l51_pdn_event, SND_SOC_DAPM_PRE_POST_PMD), /* analog/mic */ SND_SOC_DAPM_INPUT("AIN1L"), @@ -255,6 +253,12 @@ static const struct snd_soc_dapm_route cs42l51_routes[] = { {"HPL", NULL, "Left DAC"}, {"HPR", NULL, "Right DAC"}, + {"Right DAC", NULL, "DAC Mux"}, + {"Left DAC", NULL, "DAC Mux"}, + + {"DAC Mux", "Direct PCM", "Playback"}, + {"DAC Mux", "DSP PCM", "Playback"}, + {"Left ADC", NULL, "Left PGA"}, {"Right ADC", NULL, "Right PGA"}, ------------------------------------------------------------------------------- Git bisection log: ------------------------------------------------------------------------------- git bisect start # good: [e42617b825f8073569da76dc4510bfa019b1c35a] Linux 5.5-rc1 git bisect good e42617b825f8073569da76dc4510bfa019b1c35a # bad: [3778790e1d1329223601c63602286ad1d73c480f] Merge branch 'asoc-5.6' into asoc-next git bisect bad 3778790e1d1329223601c63602286ad1d73c480f # bad: [3e62579436c6a7fc35de7318e6c5f495b8d0046c] ASoC: SOF: core: modify the signature for snd_sof_create_page_table git bisect bad 3e62579436c6a7fc35de7318e6c5f495b8d0046c # good: [62d5ae4cafb7ffeeec6ba2dd1814cafeeea7dd8f] ASoC: max98090: save and restore SHDN when changing sensitive registers git bisect good 62d5ae4cafb7ffeeec6ba2dd1814cafeeea7dd8f # bad: [bc9a665581b3c6c82c9220a47f6573b49ce2df0b] ASoC: fix soc-core.c kernel-doc warning git bisect bad bc9a665581b3c6c82c9220a47f6573b49ce2df0b # bad: [abe3b6727b653307c27870a2d4ecbf9de4e914a5] ASoC: cs42l51: add dac mux widget in codec routes git bisect bad abe3b6727b653307c27870a2d4ecbf9de4e914a5 # good: [49df1e3925824cf44e590daac635974270185841] ASoC: rsnd: Calculate DALIGN inversion at run-time git bisect good 49df1e3925824cf44e590daac635974270185841 # first bad commit: [abe3b6727b653307c27870a2d4ecbf9de4e914a5] ASoC: cs42l51: add dac mux widget in codec routes -------------------------------------------------------------------------------

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[alsa-devel] [PATCH] ASoC: cs42l42: add missed regulator_bulk_disable in remove and fix probe failure
by Chuhong Yuan 10 Dec '19

10 Dec '19

The driver forgets to call regulator_bulk_disable() in remove like that in probe failure. Besides, some failed branches in probe do not handle failure correctly. Add the missed call and revise wrong direct returns to fix it. Fixes: 2c394ca79604b ("ASoC: Add support for CS42L42 codec") Signed-off-by: Chuhong Yuan <hslester96(a)gmail.com> --- sound/soc/codecs/cs42l42.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 5125bb9b37b5..96b3cff50ce9 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1796,8 +1796,10 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, /* Reset the Device */ cs42l42->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, "reset", GPIOD_OUT_LOW); - if (IS_ERR(cs42l42->reset_gpio)) - return PTR_ERR(cs42l42->reset_gpio); + if (IS_ERR(cs42l42->reset_gpio)) { + ret = PTR_ERR(cs42l42->reset_gpio); + goto err_disable; + } if (cs42l42->reset_gpio) { dev_dbg(&i2c_client->dev, "Found reset GPIO\n"); @@ -1831,13 +1833,13 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L42 Device ID (%X). Expected %X\n", devid, CS42L42_CHIP_ID); - return ret; + goto err_disable; } ret = regmap_read(cs42l42->regmap, CS42L42_REVID, &reg); if (ret < 0) { dev_err(&i2c_client->dev, "Get Revision ID failed\n"); - return ret; + goto err_disable; } dev_info(&i2c_client->dev, @@ -1863,7 +1865,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, if (i2c_client->dev.of_node) { ret = cs42l42_handle_device_data(i2c_client, cs42l42); if (ret != 0) - return ret; + goto err_disable; } /* Setup headset detection */ @@ -1892,6 +1894,8 @@ static int cs42l42_i2c_remove(struct i2c_client *i2c_client) /* Hold down reset */ gpiod_set_value_cansleep(cs42l42->reset_gpio, 0); + regulator_bulk_disable(ARRAY_SIZE(cs42l42->supplies), + cs42l42->supplies); return 0; } -- 2.24.0

2 5

[alsa-devel] [PATCH 1/3] treewide: Fix -Wformat=2 warnings
by Rosen Penev 10 Dec '19

10 Dec '19

Allows the compiler to check these formats at compile time. Fixed several such warnings. Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- include/use-case.h | 1 + src/error.c | 2 ++ src/input.c | 6 ++++-- src/output.c | 6 ++++-- src/pcm/pcm.c | 4 ++-- src/pcm/pcm_direct.c | 2 +- src/pcm/pcm_file.c | 6 +++--- src/topology/builder.c | 3 ++- src/ucm/ucm_local.h | 2 ++ 9 files changed, 21 insertions(+), 11 deletions(-) diff --git a/include/use-case.h b/include/use-case.h index 2efcb4d8..3f0cf58e 100644 --- a/include/use-case.h +++ b/include/use-case.h @@ -188,6 +188,7 @@ typedef struct snd_use_case_mgr snd_use_case_mgr_t; * \param ... Optional arguments for sprintf like format * \return Allocated string identifier or NULL on error */ +__attribute__((format(printf, 1, 2))) char *snd_use_case_identifier(const char *fmt, ...); /** diff --git a/src/error.c b/src/error.c index 2e617f87..ad281411 100644 --- a/src/error.c +++ b/src/error.c @@ -95,6 +95,7 @@ snd_local_error_handler_t snd_lib_error_set_local(snd_local_error_handler_t func * \ref snd_lib_error_set_local, it is called. Otherwise, prints the error * message including location to \c stderr. */ +__attribute__((format(printf, 5, 6))) static void snd_lib_error_default(const char *file, int line, const char *function, int err, const char *fmt, ...) { va_list arg; @@ -149,6 +150,7 @@ const char *snd_asoundlib_version(void) /* * internal error handling */ +__attribute__((format(printf, 5, 6))) static void snd_err_msg_default(const char *file, int line, const char *function, int err, const char *fmt, ...) { va_list arg; diff --git a/src/input.c b/src/input.c index 35324f1f..c204fe7b 100644 --- a/src/input.c +++ b/src/input.c @@ -132,6 +132,7 @@ static int snd_input_stdio_close(snd_input_t *input ATTRIBUTE_UNUSED) return 0; } +__attribute__((format(scanf, 2, 0))) static int snd_input_stdio_scan(snd_input_t *input, const char *format, va_list args) { snd_input_stdio_t *stdio = input->private_data; @@ -144,7 +145,7 @@ static char *snd_input_stdio_gets(snd_input_t *input, char *str, size_t size) snd_input_stdio_t *stdio = input->private_data; return fgets(str, (int) size, stdio->fp); } - + static int snd_input_stdio_getc(snd_input_t *input) { snd_input_stdio_t *stdio = input->private_data; @@ -236,6 +237,7 @@ static int snd_input_buffer_close(snd_input_t *input) return 0; } +__attribute__((format(scanf, 2, 0))) static int snd_input_buffer_scan(snd_input_t *input, const char *format, va_list args) { snd_input_buffer_t *buffer = input->private_data; @@ -262,7 +264,7 @@ static char *snd_input_buffer_gets(snd_input_t *input, char *str, size_t size) *str = '\0'; return str; } - + static int snd_input_buffer_getc(snd_input_t *input) { snd_input_buffer_t *buffer = input->private_data; diff --git a/src/output.c b/src/output.c index 7e3a91b3..935ab003 100644 --- a/src/output.c +++ b/src/output.c @@ -141,6 +141,7 @@ static int snd_output_stdio_close(snd_output_t *output) return 0; } +__attribute__((format(printf, 2, 0))) static int snd_output_stdio_print(snd_output_t *output, const char *format, va_list args) { snd_output_stdio_t *stdio = output->private_data; @@ -152,7 +153,7 @@ static int snd_output_stdio_puts(snd_output_t *output, const char *str) snd_output_stdio_t *stdio = output->private_data; return fputs(str, stdio->fp); } - + static int snd_output_stdio_putc(snd_output_t *output, int c) { snd_output_stdio_t *stdio = output->private_data; @@ -268,6 +269,7 @@ static int snd_output_buffer_need(snd_output_t *output, size_t size) return buffer->alloc - buffer->size; } +__attribute__((format(printf, 2, 0))) static int snd_output_buffer_print(snd_output_t *output, const char *format, va_list args) { snd_output_buffer_t *buffer = output->private_data; @@ -304,7 +306,7 @@ static int snd_output_buffer_puts(snd_output_t *output, const char *str) buffer->size += size; return size; } - + static int snd_output_buffer_putc(snd_output_t *output, int c) { snd_output_buffer_t *buffer = output->private_data; diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1064044c..09dfe967 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -8241,7 +8241,7 @@ int snd_pcm_chmap_print(const snd_pcm_chmap_t *map, size_t maxlen, char *buf) return -ENOMEM; } if (map->pos[i] & SND_CHMAP_DRIVER_SPEC) - len += snprintf(buf + len, maxlen - len, "%d", p); + len += snprintf(buf + len, maxlen - len, "%u", p); else { const char *name = chmap_names[p]; if (name) @@ -8249,7 +8249,7 @@ int snd_pcm_chmap_print(const snd_pcm_chmap_t *map, size_t maxlen, char *buf) "%s", name); else len += snprintf(buf + len, maxlen - len, - "Ch%d", p); + "Ch%u", p); } if (len >= maxlen) return -ENOMEM; diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 54d99005..719e1692 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1406,7 +1406,7 @@ int snd_pcm_direct_initialize_poll_fd(snd_pcm_direct_t *dmix) SNDERR("unable to info for slave pcm"); return ret; } - sprintf(name, "hw:CLASS=%i,SCLASS=0,CARD=%i,DEV=%i,SUBDEV=%i", + sprintf(name, "hw:CLASS=%i,SCLASS=0,CARD=%i,DEV=%u,SUBDEV=%u", (int)SND_TIMER_CLASS_PCM, snd_pcm_info_get_card(&info), snd_pcm_info_get_device(&info), diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index a02b2dc0..c874e863 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -155,7 +155,7 @@ static int snd_pcm_file_replace_fname(snd_pcm_file_t *file, char **new_fname_p) next char */ switch (*(++old_index_ch)) { case RATE_KEY: - snprintf(value, sizeof(value), "%d", + snprintf(value, sizeof(value), "%u", pcm->rate); err = snd_pcm_file_append_value(&new_fname, &new_index_ch, &new_len, value); @@ -164,7 +164,7 @@ static int snd_pcm_file_replace_fname(snd_pcm_file_t *file, char **new_fname_p) break; case CHANNELS_KEY: - snprintf(value, sizeof(value), "%d", + snprintf(value, sizeof(value), "%u", pcm->channels); err = snd_pcm_file_append_value(&new_fname, &new_index_ch, &new_len, value); @@ -173,7 +173,7 @@ static int snd_pcm_file_replace_fname(snd_pcm_file_t *file, char **new_fname_p) break; case BWIDTH_KEY: - snprintf(value, sizeof(value), "%d", + snprintf(value, sizeof(value), "%u", pcm->frame_bits/pcm->channels); err = snd_pcm_file_append_value(&new_fname, &new_index_ch, &new_len, value); diff --git a/src/topology/builder.c b/src/topology/builder.c index 5ae3ae89..3adbad45 100644 --- a/src/topology/builder.c +++ b/src/topology/builder.c @@ -21,6 +21,7 @@ #include "tplg_local.h" /* verbose output detailing each object size and file position */ +__attribute__((format(printf, 2, 3))) static void verbose(snd_tplg_t *tplg, const char *fmt, ...) { int offset; @@ -244,7 +245,7 @@ static int write_manifest_data(snd_tplg_t *tplg) return ret; } - verbose(tplg, "manifest : write %d bytes\n", sizeof(tplg->manifest)); + verbose(tplg, "manifest : write %zu bytes\n", sizeof(tplg->manifest)); ret = write(tplg->out_fd, &tplg->manifest, sizeof(tplg->manifest)); if (ret < 0) { SNDERR("error: failed to write manifest %d\n", ret); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index ba961507..696a60dc 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -246,7 +246,9 @@ struct snd_use_case_mgr { #define uc_dbg(fmt, arg...) do { } while (0) #endif +__attribute__((format(printf, 1, 2))) void uc_mgr_error(const char *fmt, ...); +__attribute__((format(printf, 1, 2))) void uc_mgr_stdout(const char *fmt, ...); int uc_mgr_config_load(int format, const char *file, snd_config_t **cfg); -- 2.23.0

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[alsa-devel] [PATCH 00/55] ALSA: Support PCM sync_stop
by Takashi Iwai 10 Dec '19

10 Dec '19

Hi, this is the last episode of the trilogy of PCM enhancements for non-ASoC drivers. Basically this either adds the setup for card->sync_irq and removes the superfluous synchronize_irq() calls, or replaces the synchronization code with the new sync_irq PCM ops. For the explanation about PCM ssync_stop and card sync_irq, see the commits: 1e850beea2781d30418743dd99250291cef37919 ALSA: pcm: Add the support for sync-stop operation fabb26dcd104027b971c018275fe40f2ebe09ae3 ALSA: pcm: Add card sync_irq field thanks, Takashi === Takashi Iwai (55): ALSA: ad1889: Support PCM sync_stop ALSA: ali5451: Support PCM sync_stop ALSA: als300: Support PCM sync_stop ALSA: atiixp: Support PCM sync_stop ALSA: au88x0: Support PCM sync_stop ALSA: aw2: Support PCM sync_stop ALSA: azt3328: Support PCM sync_stop ALSA: bt87x: Support PCM sync_stop ALSA: ca0106: Support PCM sync_stop ALSA: cmipci: Support PCM sync_stop ALSA: cs4281: Support PCM sync_stop ALSA: cs46xx: Support PCM sync_stop ALSA: cs5535audio: Support PCM sync_stop ALSA: ctxfi: Support PCM sync_stop ALSA: echoaudio: Support PCM sync_stop ALSA: emu10k1: Support PCM sync_stop ALSA: ens137x: Support PCM sync_stop ALSA: es1938: Support PCM sync_stop ALSA: es1968: Support PCM sync_stop ALSA: fm801: Support PCM sync_stop ALSA: hda: Support PCM sync_stop ALSA: ice1712: Support PCM sync_stop ALSA: ice1724: Support PCM sync_stop ALSA: intel8x0: Support PCM sync_stop ALSA: korg1212: Support PCM sync_stop ALSA: lola: Support PCM sync_stop ALSA: maestro3: Support PCM sync_stop ALSA: nm256: Support PCM sync_stop ALSA: oxygen: Support PCM sync_stop ALSA: riptide: Support PCM sync_stop ALSA: rme32: Support PCM sync_stop ALSA: rme96: Support PCM sync_stop ALSA: rme9652: Support PCM sync_stop ALSA: sis7019: Support PCM sync_stop ALSA: sonicvibes: Support PCM sync_stop ALSA: trident: Support PCM sync_stop ALSA: via82xx: Support PCM sync_stop ALSA: ymfpci: Support PCM sync_stop ALSA: ad1816a: Support PCM sync_stop ALSA: es1688: Support PCM sync_stop ALSA: es18xx: Support PCM sync_stop ALSA: gus: Support PCM sync_stop ALSA: msnd: Support PCM sync_stop ALSA: opl3sa2: Support PCM sync_stop ALSA: opti9xx: Support PCM sync_stop ALSA: sb: Support PCM sync_stop ALSA: wavefront: Support PCM sync_stop ALSA: wss: Support PCM sync_stop ALSA: lx6464es: Support PCM sync_stop ALSA: mixart: Support PCM sync_stop ALSA: pcxhr: Support PCM sync_stop ALSA: vx222: Support PCM sync_stop ALSA: pdaudiocf: Support PCM sync_stop ALSA: vxpocket: Support PCM sync_stop ALSA: usb-audio: Support PCM sync_stop sound/isa/ad1816a/ad1816a_lib.c | 1 + sound/isa/es1688/es1688_lib.c | 1 + sound/isa/es18xx.c | 1 + sound/isa/gus/gus_main.c | 1 + sound/isa/gus/gusmax.c | 3 ++- sound/isa/gus/interwave.c | 1 + sound/isa/msnd/msnd_pinnacle.c | 1 + sound/isa/opl3sa2.c | 1 + sound/isa/opti9xx/opti92x-ad1848.c | 1 + sound/isa/sb/sb_common.c | 1 + sound/isa/wavefront/wavefront.c | 1 + sound/isa/wss/wss_lib.c | 1 + sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 3 +-- sound/pci/als300.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.c | 1 + sound/pci/aw2/aw2-alsa.c | 1 + sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 2 +- sound/pci/ca0106/ca0106_main.c | 1 + sound/pci/cmipci.c | 1 + sound/pci/cs4281.c | 5 +---- sound/pci/cs46xx/cs46xx_lib.c | 1 + sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/cthw20k1.c | 4 +--- sound/pci/ctxfi/cthw20k2.c | 1 + sound/pci/echoaudio/echoaudio.c | 3 +++ sound/pci/emu10k1/emu10k1_main.c | 1 + sound/pci/emu10k1/emu10k1x.c | 1 + sound/pci/ens1370.c | 4 +--- sound/pci/es1938.c | 3 +++ sound/pci/es1968.c | 3 +-- sound/pci/fm801.c | 1 + sound/pci/hda/hda_intel.c | 4 +++- sound/pci/hda/hda_tegra.c | 4 +--- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 4 +++- sound/pci/intel8x0m.c | 3 +++ sound/pci/korg1212/korg1212.c | 1 + sound/pci/lola/lola.c | 2 +- sound/pci/lx6464es/lx6464es.c | 1 + sound/pci/maestro3.c | 1 + sound/pci/mixart/mixart.c | 1 + sound/pci/nm256/nm256.c | 2 ++ sound/pci/oxygen/oxygen_lib.c | 2 +- sound/pci/pcxhr/pcxhr.c | 1 + sound/pci/riptide/riptide.c | 1 + sound/pci/rme32.c | 1 + sound/pci/rme96.c | 1 + sound/pci/rme9652/hdsp.c | 1 + sound/pci/rme9652/hdspm.c | 1 + sound/pci/rme9652/rme9652.c | 1 + sound/pci/sis7019.c | 1 + sound/pci/sonicvibes.c | 1 + sound/pci/trident/trident_main.c | 1 + sound/pci/via82xx.c | 3 +-- sound/pci/via82xx_modem.c | 3 +-- sound/pci/vx222/vx222.c | 1 + sound/pci/ymfpci/ymfpci_main.c | 1 + sound/pcmcia/pdaudiocf/pdaudiocf.c | 1 + sound/pcmcia/vx/vxpocket.c | 1 + sound/usb/pcm.c | 37 +++++++++++++++++++++++++------------ 65 files changed, 101 insertions(+), 47 deletions(-) -- 2.16.4

1 55

[alsa-devel] [PATCH][resend] ASoC: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 10 Dec '19

10 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/simple-card.txt | 351 --------------- .../devicetree/bindings/sound/simple-card.yaml | 472 +++++++++++++++++++++ 2 files changed, 472 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6..0000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 0000000..ac3975f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,472 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle-array + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle-array + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + hp-det-gpio: + description: | + Reference to GPIO that signals when headphones are attached. + $ref: /schemas/types.yaml#/definitions/phandle + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + mic-det-gpio: + description: | + Reference to GPIO that signals when a microphone is attached. + $ref: /schemas/types.yaml#/definitions/phandle + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + oneOf: + - items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + clocks: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/phandle-array + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": true + "#size-cells": true + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + simple-audio-card,routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + "^simple-audio-card,cpu.*": + description: CPU node + $ref: "#/definitions/dai" + "^simple-audio-card,codec.*": + description: CPU node + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,hp-det-gpio$": + $ref: "#/definitions/hp-det-gpio" + "^simple-audio-card,mic-det-gpio$": + $ref: "#/definitions/mic-det-gpio" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" + + "^simple-audio-card,dai-link": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + hp-det-gpio: + $ref: "#/definitions/hp-det-gpio" + mic-det-gpio: + $ref: "#/definitions/mic-det-gpio" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + + patternProperties: + "^cpu.*": + $ref: "#/definitions/dai" + "^codec.*": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.7.4

1 0

[alsa-devel] [PATCH][resend] ASoC: renesas, rsnd: switch to yaml base Documentation
by Kuninori Morimoto 10 Dec '19

10 Dec '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. It is still keeping detail explanations at .txt Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 518 -------------------- .../devicetree/bindings/sound/renesas,rsnd.yaml | 527 +++++++++++++++++++++ 2 files changed, 527 insertions(+), 518 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 797fd03..b731f16 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -253,521 +253,3 @@ This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. see "Example: simple sound card for TDM" - -============================================= -Required properties: -============================================= - -- compatible : "renesas,rcar_sound-<soctype>", fallbacks - "renesas,rcar_sound-gen1" if generation1, and - "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) - Examples with soctypes are: - - "renesas,rcar_sound-r8a7743" (RZ/G1M) - - "renesas,rcar_sound-r8a7744" (RZ/G1N) - - "renesas,rcar_sound-r8a7745" (RZ/G1E) - - "renesas,rcar_sound-r8a77470" (RZ/G1C) - - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - - "renesas,rcar_sound-r8a7778" (R-Car M1A) - - "renesas,rcar_sound-r8a7779" (R-Car H1) - - "renesas,rcar_sound-r8a7790" (R-Car H2) - - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - - "renesas,rcar_sound-r8a7794" (R-Car E2) - - "renesas,rcar_sound-r8a7795" (R-Car H3) - - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - - "renesas,rcar_sound-r8a77990" (R-Car E3) - - "renesas,rcar_sound-r8a77995" (R-Car D3) -- reg : Should contain the register physical address. - required register is - SRU/ADG/SSI if generation1 - SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3 - Select extended AUDIO-DMAC-periperi address if SoC has it, - otherwise select normal AUDIO-DMAC-periperi address. -- reg-names : Should contain the register names. - scu/adg/ssi if generation1 - scu/adg/ssiu/ssi/audmapp if generation2/generation3 -- rcar_sound,ssi : Should contain SSI feature. - The number of SSI subnode should be same as HW. - see below for detail. -- rcar_sound,ssiu : Should contain SSIU feature. - The number of SSIU subnode should be same as HW. - see below for detail. -- rcar_sound,src : Should contain SRC feature. - The number of SRC subnode should be same as HW. - see below for detail. -- rcar_sound,ctu : Should contain CTU feature. - The number of CTU subnode should be same as HW. - see below for detail. -- rcar_sound,mix : Should contain MIX feature. - The number of MIX subnode should be same as HW. - see below for detail. -- rcar_sound,dvc : Should contain DVC feature. - The number of DVC subnode should be same as HW. - see below for detail. -- rcar_sound,dai : DAI contents. - The number of DAI subnode should be same as HW. - see below for detail. -- #sound-dai-cells : it must be 0 if your system is using single DAI - it must be 1 if your system is using multi DAI -- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. -- clock-names : List of necessary clock names. - "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", - "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" - -Optional properties: -- #clock-cells : it must be 0 if your system has audio_clkout - it must be 1 if your system has audio_clkout0/1/2/3 -- clock-frequency : for all audio_clkout0/1/2/3 -- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn - is asynchronizes with lr-clock. -- resets : References to SSI resets. -- reset-names : List of valid reset names. - "ssi-all", "ssi.X" - -SSI subnode properties: -- interrupts : Should contain SSI interrupt for PIO transfer -- shared-pin : if shared clock pin -- pio-transfer : use PIO transfer mode -- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case -- dma : Should contain Audio DMAC entry -- dma-names : SSI case "rx" (=playback), "tx" (=capture) - Deprecated: see SSIU subnode properties - SSIU case "rxu" (=playback), "txu" (=capture) - -SSIU subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -SRC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -DVC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "tx" (=playback/capture) - -DAI subnode properties: -- playback : list of playback modules -- capture : list of capture modules - - -============================================= -Example: -============================================= - -rcar_sound: sound@ec500000 { - #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; - reg = <0 0xec500000 0 0x1000>, /* SCU */ - <0 0xec5a0000 0 0x100>, /* ADG */ - <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ - reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; - - clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, - <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, - <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, - <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, - <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, - <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, - <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, - <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, - <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, - <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, - <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, - <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, - <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; - clock-names = "ssi-all", - "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", - "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", - "src.9", "src.8", "src.7", "src.6", "src.5", - "src.4", "src.3", "src.2", "src.1", "src.0", - "dvc.0", "dvc.1", - "clk_a", "clk_b", "clk_c", "clk_i"; - - rcar_sound,dvc { - dvc0: dvc-0 { - dmas = <&audma0 0xbc>; - dma-names = "tx"; - }; - dvc1: dvc-1 { - dmas = <&audma0 0xbe>; - dma-names = "tx"; - }; - }; - - rcar_sound,mix { - mix0: mix-0 { }; - mix1: mix-1 { }; - }; - - rcar_sound,ctu { - ctu00: ctu-0 { }; - ctu01: ctu-1 { }; - ctu02: ctu-2 { }; - ctu03: ctu-3 { }; - ctu10: ctu-4 { }; - ctu11: ctu-5 { }; - ctu12: ctu-6 { }; - ctu13: ctu-7 { }; - }; - - rcar_sound,src { - src0: src-0 { - interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x85>, <&audma1 0x9a>; - dma-names = "rx", "tx"; - }; - src1: src-1 { - interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x87>, <&audma1 0x9c>; - dma-names = "rx", "tx"; - }; - src2: src-2 { - interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x89>, <&audma1 0x9e>; - dma-names = "rx", "tx"; - }; - src3: src-3 { - interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8b>, <&audma1 0xa0>; - dma-names = "rx", "tx"; - }; - src4: src-4 { - interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8d>, <&audma1 0xb0>; - dma-names = "rx", "tx"; - }; - src5: src-5 { - interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8f>, <&audma1 0xb2>; - dma-names = "rx", "tx"; - }; - src6: src-6 { - interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x91>, <&audma1 0xb4>; - dma-names = "rx", "tx"; - }; - src7: src-7 { - interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x93>, <&audma1 0xb6>; - dma-names = "rx", "tx"; - }; - src8: src-8 { - interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x95>, <&audma1 0xb8>; - dma-names = "rx", "tx"; - }; - src9: src-9 { - interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x97>, <&audma1 0xba>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssiu { - ssiu00: ssiu-0 { - dmas = <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx"; - }; - ssiu01: ssiu-1 { - dmas = <&audma0 0x35>, <&audma1 0x36>; - dma-names = "rx", "tx"; - }; - - ... - - ssiu95: ssiu-49 { - dmas = <&audma0 0xA5>, <&audma1 0xA6>; - dma-names = "rx", "tx"; - }; - ssiu96: ssiu-50 { - dmas = <&audma0 0xA7>, <&audma1 0xA8>; - dma-names = "rx", "tx"; - }; - ssiu97: ssiu-51 { - dmas = <&audma0 0xA9>, <&audma1 0xAA>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssi { - ssi0: ssi-0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>; - dma-names = "rx", "tx"; - }; - ssi1: ssi-1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>; - dma-names = "rx", "tx"; - }; - - ... - - ssi8: ssi-8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>; - dma-names = "rx", "tx"; - }; - ssi9: ssi-9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,dai { - dai0 { - playback = <&ssi5 &src5>; - capture = <&ssi6>; - }; - dai1 { - playback = <&ssi3>; - }; - dai2 { - capture = <&ssi4>; - }; - dai3 { - playback = <&ssi7>; - }; - dai4 { - capture = <&ssi8>; - }; - }; -}; - -============================================= -Example: simple sound card -============================================= - - rsnd_ak4643: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; - }; - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; -}; - -&ssi1 { - shared-pin; -}; - -============================================= -Example: simple sound card for Asynchronous mode -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * SRC Asynchronous mode setting - * Playback: - * All input data will be converted to 48kHz - * Capture: - * Inputed 48kHz data will be converted to - * system specified Hz - */ - simple-audio-card,convert-rate = <48000>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for channel convert -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * CTU setting - * All input data will be converted to 2ch - * as output data - */ - simple-audio-card,convert-channels = <2>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for MIXer -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - simple-audio-card,cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - simple-audio-card,cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - simple-audio-card,codec { - ... - }; -}; - -&rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; - }; - dai1 { - playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; - }; - }; -}; - -============================================= -Example: simple sound card for TDM -============================================= - -rsnd_tdm: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - dai-tdm-slot-num = <6>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&xxx>; - }; -}; - -============================================= -Example: simple sound card for TDM Split -============================================= - -sound_card: sound { - compatible = "audio-graph-scu-card"; - prefix = "xxxx"; - routing = "xxxx Playback", "DAI0 Playback", - "xxxx Playback", "DAI1 Playback", - "xxxx Playback", "DAI2 Playback", - "xxxx Playback", "DAI3 Playback"; - convert-channels = <8>; /* TDM Split */ - - dais = <&rsnd_port0 /* playback ch1/ch2 */ - &rsnd_port1 /* playback ch3/ch4 */ - &rsnd_port2 /* playback ch5/ch6 */ - &rsnd_port3 /* playback ch7/ch8 */ - >; -}; - -audio-codec { - ... - port { - codec_0: endpoint@1 { - remote-endpoint = <&rsnd_ep0>; - }; - codec_1: endpoint@2 { - remote-endpoint = <&rsnd_ep1>; - }; - codec_2: endpoint@3 { - remote-endpoint = <&rsnd_ep2>; - }; - codec_3: endpoint@4 { - remote-endpoint = <&rsnd_ep3>; - }; - }; -}; - -&rcar_sound { - ... - ports { - rsnd_port0: port@0 { - rsnd_ep0: endpoint { - remote-endpoint = <&codec_0>; - ... - playback = <&ssiu30 &ssi3>; - }; - }; - rsnd_port1: port@1 { - rsnd_ep1: endpoint { - remote-endpoint = <&codec_1>; - ... - playback = <&ssiu31 &ssi3>; - }; - }; - rsnd_port2: port@2 { - rsnd_ep2: endpoint { - remote-endpoint = <&codec_2>; - ... - playback = <&ssiu32 &ssi3>; - }; - }; - rsnd_port3: port@3 { - rsnd_ep3: endpoint { - remote-endpoint = <&codec_3>; - ... - playback = <&ssiu33 &ssi3>; - }; - }; - }; -}; - -============================================= -Example: simple sound card for Multi channel -============================================= - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 0000000..5fb036c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,527 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + + reg-names: + description: | + Should contain the register names. + scu/adg/ssi if generation1 + scu/adg/ssiu/ssi/audmapp if generation2/generation3 + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + allOf: + - items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + $ref: /schemas/types.yaml#/definitions/uint32-array + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + description: References to SSI resets + $ref: /schemas/types.yaml#/definitions/phandle-array + + reset-names: + description: List of valid reset names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + + clocks: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + + clock-names: + description: List of necessary clock names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + - src.9 + - src.8 + - src.7 + - src.6 + - src.5 + - src.4 + - src.3 + - src.2 + - src.1 + - src.0 + - mix.1 + - mix.0 + - ctu.1 + - ctu.0 + - dvc.1 + - dvc.0 + - clk_a + - clk_b + - clk_c + - clk_i + + # For OF-graph + port: + description: OF-Graph subnode + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + endpoint: + type: object + properties: + remote-endpoint: + $ref: /schemas/types.yaml#/definitions/phandle-array + dai-format: + $ref: "simple-card.yaml#/definitions/format" + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - remote-endpoint + + # For multi OF-graph + ports: + description: multi OF-Graph subnode + type: object + patternProperties: + "port.*": + $ref: "#properties/port" + +patternProperties: + "^rcar_sound,dvc$": + description: DVC subnode. + type: object + patternProperties: + "dvc.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + const: "tx" + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,mix$": + description: MIX subnode. + type: object + patternProperties: + "mix.*": + type: object + additionalProperties: false + + "^rcar_sound,ctu$": + description: CTU subnode. + type: object + patternProperties: + "ctu.*": + type: object + additionalProperties: false + + "^rcar_sound,src$": + description: SRC subnode. + type: object + patternProperties: + "src.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + if: + properties: + status: + const: "ok" + then: + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssiu$": + description: SSIU subnode. + type: object + patternProperties: + "ssiu.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssi$": + description: SSI subnode. + type: object + patternProperties: + "ssi.*": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + "^rcar_sound,dai$": + description: DAI subnode. + type: object + patternProperties: + "dai.*": + type: object + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0 0xec500000 0 0x1000>, /* SCU */ + <0 0xec5a0000 0 0x100>, /* ADG */ + <0 0xec540000 0 0x1000>, /* SSIU */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* DVC0, DVC1 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CLKA, CLKB */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* OF-Graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + + /* Multi OF-Graph */ + ports { + #address-cells = <1>; + #size-cells = <0>; + rsnd_port0: port@0 { + reg = <0>; + rsnd_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + rsnd_port1: port@1 { + reg = <1>; + rsnd_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint2>; + + dai-format = "i2s"; + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + }; + }; + + + codec { + ports { + port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + port@1 { + codec_endpoint1: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + port@2 { + codec_endpoint2: endpoint { + remote-endpoint = <&rsnd_endpoint1>; + }; + }; + }; + }; -- 2.7.4

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Re: [alsa-devel] [PATCH v3 08/15] ASoC: tegra: Add audio mclk control through clk_out_1 and extern1
by Dmitry Osipenko 10 Dec '19

10 Dec '19

10.12.2019 02:05, Sowjanya Komatineni пишет: > > On 12/9/19 12:06 PM, Dmitry Osipenko wrote: >> 07.12.2019 22:20, Sowjanya Komatineni пишет: >>> On 12/7/19 6:58 AM, Dmitry Osipenko wrote: >>>> 06.12.2019 05:48, Sowjanya Komatineni пишет: >>>>> Current ASoC driver uses extern1 as cdev1 clock from Tegra30 onwards >>>>> through device tree. >>>>> >>>>> Actual audio mclk is clk_out_1 and to use PLLA for mclk rate control, >>>>> need to clk_out_1_mux parent to extern1 and extern1 parent to >>>>> PLLA_OUT0. >>>>> >>>>> Currently Tegra clock driver init sets the parents and enables both >>>>> clk_out_1 and extern1 clocks. But these clocks parent and enables >>>>> should >>>>> be controlled by ASoC driver. >>>>> >>>>> Clock parents can be specified in device tree using assigned-clocks >>>>> and assigned-clock-parents. >>>>> >>>>> To enable audio mclk, both clk_out_1 and extern1 clocks need to be >>>>> enabled. >>>>> >>>>> This patch configures parents for clk_out_1 and extern1 clocks if >>>>> device >>>>> tree does not specify clock parents inorder to support old device tree >>>>> and controls mclk using both clk_out_1 and extern1 clocks. >>>>> >>>>> Signed-off-by: Sowjanya Komatineni <skomatineni(a)nvidia.com> >>>>> --- >>>>> sound/soc/tegra/tegra_asoc_utils.c | 66 >>>>> ++++++++++++++++++++++++++++++++++++++ >>>>> sound/soc/tegra/tegra_asoc_utils.h | 1 + >>>>> 2 files changed, 67 insertions(+) >>>>> >>>>> diff --git a/sound/soc/tegra/tegra_asoc_utils.c >>>>> b/sound/soc/tegra/tegra_asoc_utils.c >>>>> index 536a578e9512..8e3a3740df7c 100644 >>>>> --- a/sound/soc/tegra/tegra_asoc_utils.c >>>>> +++ b/sound/soc/tegra/tegra_asoc_utils.c >>>>> @@ -60,6 +60,7 @@ int tegra_asoc_utils_set_rate(struct >>>>> tegra_asoc_utils_data *data, int srate, >>>>> data->set_mclk = 0; >>>>> clk_disable_unprepare(data->clk_cdev1); >>>>> + clk_disable_unprepare(data->clk_extern1); >>>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>>> clk_disable_unprepare(data->clk_pll_a); >>>>> @@ -89,6 +90,14 @@ int tegra_asoc_utils_set_rate(struct >>>>> tegra_asoc_utils_data *data, int srate, >>>>> return err; >>>>> } >>>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>>> + err = clk_prepare_enable(data->clk_extern1); >>>>> + if (err) { >>>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>>> + return err; >>>>> + } >>>>> + } >>>>> + >>>>> err = clk_prepare_enable(data->clk_cdev1); >>>>> if (err) { >>>>> dev_err(data->dev, "Can't enable cdev1: %d\n", err); >>>>> @@ -109,6 +118,7 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>>> tegra_asoc_utils_data *data) >>>>> int err; >>>>> clk_disable_unprepare(data->clk_cdev1); >>>>> + clk_disable_unprepare(data->clk_extern1); >>>>> clk_disable_unprepare(data->clk_pll_a_out0); >>>>> clk_disable_unprepare(data->clk_pll_a); >>>>> @@ -142,6 +152,14 @@ int tegra_asoc_utils_set_ac97_rate(struct >>>>> tegra_asoc_utils_data *data) >>>>> return err; >>>>> } >>>>> + if (!IS_ERR_OR_NULL(data->clk_extern1)) { >>>>> + err = clk_prepare_enable(data->clk_extern1); >>>>> + if (err) { >>>>> + dev_err(data->dev, "Can't enable extern1: %d\n", err); >>>>> + return err; >>>>> + } >>>>> + } >>>> Why this is needed given that clk_extern1 is either a child of MCLK or >>>> MCLK itself (on T20)? The child clocks are enabled when the parent is >>>> enabled. >>> For T30 and later, clk_extern1 is one of the source for clk_out_1_mux. >>> clk_extern1 is in CAR and it has its own gate and mux. >>> >>> As audio mclk related clocks (clk_out_1, clk_out_1_mux, and extern1) are >>> moved into ASoC driver from clock driver >>> >>> need to enable extern1 gate as well along with clk_out1 for T30 through >>> T210. >>> >>> Just FYI, extern1 enable here happens only when data->clk_extern1 is >>> available which is for T30 onwards. >> clk_out_1 is the parent of extern1, thus extern1 is enabled by the clk >> core whenever clk_out_1 is enabled because data->clk_cdev1=clk_out_1. An >> I missing something? >> >> [snip] > extern1 is the parent for clk_out_1. explained extern1 clock path to > clk_out in reply to your comment in other patch of this series. Right, I meant extern1 the parent of clk_out_1, sorry for the confusion. So when clk_out_1 (child) is enabled, extern1 (parent) is enabled as well. I'll take a closer look at the other email tomorrow.

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