- 05 8月, 2019 5 次提交
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using component->driver->ops->xxx, thus, it is deep nested, and makes code difficult to read, and is not good for encapsulation. This patch adds new snd_soc_component_open() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87ftmt5rnx.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
ALSA SoC is calling try_module_get()/module_put() based on component->driver->module_get_upon_open. To keep simple and readable code, we should create its function. This patch adds new snd_soc_component_get/put(). Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87h8795ro4.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
No ALSA SoC driver has .fill_silence at component->driver->ops. We can revive it if some-driver want to use it, but let's remove it so far to avoid maintaining complex code Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87k1c55rof.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
No ALSA SoC driver has .copy_kernel at component->driver->ops. We can revive it if some-driver want to use it, but let's remove it so far to avoid maintaining complex code Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87lfwl5rot.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
No ALSA SoC driver has .ack at component->driver->ops. We can revive it if some-driver want to use it, but let's remove it so far to avoid maintaining complex code Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87muh15roz.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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- 01 8月, 2019 2 次提交
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由 Greg Kroah-Hartman 提交于
When calling debugfs functions, there is no need to ever check the return value. The function can work or not, but the code logic should never do something different based on this. Also, there is no need to store the individual debugfs file name, just remove the whole directory all at once, saving a local variable. Note, the soc-pcm "state" file has now moved to a subdirectory, as it is only a good idea to save the dentries for debugfs directories, not individual files, as the individual file debugfs functions are changing to not return a dentry. Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Jaroslav Kysela <perex@perex.cz> Cc: Takashi Iwai <tiwai@suse.com> Cc: alsa-devel@alsa-project.org Signed-off-by: NGreg Kroah-Hartman <gregkh@linuxfoundation.org> Link: https://lore.kernel.org/r/20190731131716.9764-2-gregkh@linuxfoundation.orgSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Jerome Brunet 提交于
At the moment, codec to codec links uses an ephemeral variable for the struct snd_pcm_substream. Also the struct snd_soc_pcm_runtime does not have real struct snd_pcm. This might a problem if the functions used by a codec on codec to codec link expect these structures to exist, and keep on existing during the life of the codec. For example, it is the case of the hdmi-codec, which uses snd_pcm_add_chmap_ctls(). For the controls to works, the pcm and substream must to exist. This change is first step, it create pcm (and substreams) for codec to codec links, in the same way as dpcm backend links. Signed-off-by: NJerome Brunet <jbrunet@baylibre.com> Link: https://lore.kernel.org/r/20190725165949.29699-5-jbrunet@baylibre.comSigned-off-by: NMark Brown <broonie@kernel.org>
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- 24 7月, 2019 9 次提交
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由 Kuninori Morimoto 提交于
snd_soc_dai_stream_valid() is function to check stream validity. But, some code is using it, some code are checking stream->channels_min directly. Doing samethings by different method is confusable. This patch uses same funcntion for same purpose. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87ftmyhmzz.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_delay() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87o91mhn3i.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_bespoke_trigger() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87r26ihn3u.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_trigger() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87sgqyhn40.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_prepare() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87tvbehn46.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_shutdown() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87v9vuhn4b.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_startup() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87wogahn4i.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Current ALSA SoC is directly using dai->driver->ops->xxx, thus, it has deep nested bracket, and it makes code unreadable. This patch adds new snd_soc_dai_hw_free() and use it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87y30qhn4w.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Sometimes ALSA SoC naming is very random. Current soc_dai_hw_params() should use snd_soc_dai_xxx() style. And then, 1st parameter should be dai. Otherwise it is confusable. - soc_dai_hw_params(..., dai); + snd_soc_dai_hw_params(dai, ...); Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87zhl6hn5b.wl-kuninori.morimoto.gx@renesas.comSigned-off-by: NMark Brown <broonie@kernel.org>
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- 22 5月, 2019 1 次提交
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由 Kuninori Morimoto 提交于
soc_pcm_components_open/close() try to call try_module_get()/module_put() based on component->driver->module_get_upon_open. Here, the purpose why we need to call these functions are to checking module reference. Thus, we need to call try_module_open() even though it doesn't have .open callback. The same reason, we need to call module_put() even though it doesn't have .close This patch calls try_module_get()/module_put() regardless of .open/.close Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 13 5月, 2019 4 次提交
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由 Kuninori Morimoto 提交于
soc-pcm.c has soc_pcm_components_close() but not have its open() side function. This kind of unbalance function is very unreadable. And, current error handling is not correct. Because it is using for_each_rtdcom() loop, we need to call soc_pcm_components_close() anyway even though CPU DAI .startup() failed. This patch adds soc_pcm_components_open(), and fixup error handling issue. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
Codec side is setting codec_dai->rate = 0 when error case at soc_pcm_hw_params(), but there is not such setting for CPU side. This patch adds it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
cpu_dai related operation is separated by component operation at soc_pcm_hw_params() somehow. It is not readable, let's do it at same place Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Libin Yang 提交于
If playback/capture is paused and system enters S3, after system returns from suspend, BE dai needs to call prepare() callback when playback/capture is released from pause if RESUME_INFO flag is not set. Currently, the dpcm_be_dai_prepare() function will block calling prepare() if the pcm is in SND_SOC_DPCM_STATE_PAUSED state. This will cause the following test case fail if the pcm uses BE: playback -> pause -> S3 suspend -> S3 resume -> pause release The playback may exit abnormally when pause is released because the BE dai prepare() is not called. This patch allows dpcm_be_dai_prepare() to call dai prepare() callback in SND_SOC_DPCM_STATE_PAUSED state. Signed-off-by: NLibin Yang <libin.yang@intel.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 03 5月, 2019 2 次提交
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由 Jerome Brunet 提交于
Like for hw_params, hw_free should not be called on codec dai for which the current stream is invalid. Fixes: cde79035 ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: NJerome Brunet <jbrunet@baylibre.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Jerome Brunet 提交于
A stream may specify a rate range using 'rate_min' and 'rate_max', so a stream may be valid and not specify any rates. However, as stream cannot be valid and not have any channel. Let's use this condition instead to determine if a stream is valid or not. Fixes: cde79035 ("ASoC: Handle multiple codecs with split playback / capture") Signed-off-by: NJerome Brunet <jbrunet@baylibre.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 20 4月, 2019 1 次提交
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由 Libin Yang 提交于
Some drivers mandate setting up hw params after resuming from system sleep. Since, the hw_params ioctl is not invoked upon resuming, the fixed-up BE dai hw params should be saved so the driver can use it in its resume sequence. Signed-off-by: NLibin Yang <libin.yang@intel.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 10 4月, 2019 1 次提交
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由 Ranjani Sridharan 提交于
Handle error before returning when try_module_get() fails to prevent inconsistent mutex lock/unlock. Fixes: 52034add (ASoC: pcm: update module refcount if module_get_upon_open is set) Signed-off-by: NRanjani Sridharan <ranjani.sridharan@linux.intel.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 08 4月, 2019 1 次提交
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由 Ranjani Sridharan 提交于
Setting the module_get_upon_open field for component driver prevents the module refcount from being incremented during component probe(). This could lead to the module being allowed to be unloaded when a pcm stream is open. So, if this field is set, the module's refcount should be incremented during pcm open to prevent module removal when the component is in use. And, the refcount should be decremented upon pcm close. Signed-off-by: NRanjani Sridharan <ranjani.sridharan@linux.intel.com> Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 02 4月, 2019 1 次提交
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由 Jerome Brunet 提交于
If for any reason, the backend does not have the requested substream (like capture on a playback only backend), the BE will be skipped in dpcm_be_dai_startup(). However, dpcm_apply_symmetry() does not skip those BE and will dereference the be_substream (NULL) pointer anyway. Like in dpcm_be_dai_startup(), just skip those BE. Fixes: 906c7d69 ("ASoC: dpcm: Apply symmetry for DPCM") Signed-off-by: NJerome Brunet <jbrunet@baylibre.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 18 3月, 2019 1 次提交
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由 KaiChieh Chuang 提交于
The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: NKaiChieh Chuang <kaichieh.chuang@mediatek.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 12 3月, 2019 2 次提交
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由 KaiChieh Chuang 提交于
The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Use irq spin lock version, since the lock may be used in interrupts. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: NKaiChieh Chuang <kaichieh.chuang@mediatek.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Rander Wang 提交于
On HDaudio platforms, if playback is started when capture is working, there is no audible output. This can be root-caused to the use of the rx|tx_mask to store an HDaudio stream tag. If capture is stared before playback, rx_mask would be non-zero on HDaudio platform, then the channel number of playback, which is in the same codec dai with the capture, would be changed by soc_pcm_codec_params_fixup based on the tx_mask at first, then overwritten by this function based on rx_mask at last. According to the author of tx|rx_mask, tx_mask is for playback and rx_mask is for capture. And stream direction is checked at all other references of tx|rx_mask in ASoC, so here should be an error. This patch checks stream direction for tx|rx_mask for fixup function. This issue would affect not only HDaudio+ASoC, but also I2S codecs if the channel number based on rx_mask is not equal to the one for tx_mask. It could be rarely reproduecd because most drivers in kernel set the same channel number to tx|rx_mask or rx_mask is zero. Tested on all platforms using stream_tag & HDaudio and intel I2S platforms. Signed-off-by: NRander Wang <rander.wang@linux.intel.com> Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 03 2月, 2019 1 次提交
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由 Charles Keepax 提交于
Currently all widgets attached to a DAI link will be powered up when the DAI is active, however this may include routes that are not actually in use if there are unused channels available on the DAI. The macros for creating AIF widgets already include an entry for slot, it is proposed to change that to channel. The effective difference here being respresenting the logical channel index rather than the physical slot index. The CODECs currently using the slot entry on the DAPM_AIF macros are using it in a manner consistent with this, the CODECs not using it just have the field set to zero. A variable is added to snd_soc_dapm_widget to represent this channel index and then for each AIF widget attached to a DAI this is compared against the number of channels on the stream. Enabling the links for those which will be in use. This has the nice property that the CODECs which haven't used the slot/channel entry in the macro will function exactly as before due to all the AIF widgets having a channel of zero and a stream by definition having at least one channel. Signed-off-by: NCharles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 16 1月, 2019 1 次提交
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由 Takashi Iwai 提交于
Until now we rely on each driver calling snd_pcm_suspend*() explicitly at its own PM handling. However, this can be done far more easily by setting the PM ops to each actual snd_pcm device object. This patch adds the device_type object for PCM stream and assigns to each PCM stream object. The type contains only the PM ops for system suspend; we don't need to deal with the resume in general. The suspend hook simply calls snd_pcm_suspend_all() for the given PCM streams. This implies that the PM order is correctly put, i.e. PCM is suspended before the main (or codec) driver, which should be true in general. If a special ordering is needed, you'd need to adjust the device PM order manually later. This patch introduces a new flag, snd_pcm.no_device_suspend, too. With this flag set, the PCM device object won't invoke snd_pcm_suspend_all() by itself. This is needed for ASoC who wants to manage the PM call orders in its serialized way, and the flag is set in soc_new_pcm() as default. For the non-ASoC world, we can get rid of the manual snd_pcm_suspend calls. This will be done in the later patches. Reviewed-by: NJaroslav Kysela <perex@perex.cz> Signed-off-by: NTakashi Iwai <tiwai@suse.de>
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- 21 9月, 2018 5 次提交
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由 Kuninori Morimoto 提交于
To be more readable code, this patch adds new for_each_dpcm_be() macro, and replace existing code to it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
To be more readable code, this patch adds new for_each_dpcm_fe() macro, and replace existing code to it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
To be more readable code, this patch adds new for_each_card_rtds() macro, and replace existing code to it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
commit 0b7990e3 ("ASoC: add for_each_rtd_codec_dai() macro") added for_each_rtd_codec_dai_reverse(). but _rollback() is better naming than _reverse(). This patch rename it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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由 Kuninori Morimoto 提交于
commit 0b7990e3 ("ASoC: add for_each_rtd_codec_dai() macro") added for_each_rtd_codec_dai(), but it didn't convert few loop which is not using "rtd". This patch fixup it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 04 9月, 2018 1 次提交
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由 Kuninori Morimoto 提交于
ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai. To be more readable code, this patch adds new for_each_rtd_codec_dai() macro, and replace existing code to it. Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 29 8月, 2018 1 次提交
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由 Charles Keepax 提交于
If the CPU DAI does not initialise rate_max, say if using using KNOT or CONTINUOUS, then the rate_max field will be initialised to 0. A value of zero in the rate_max field of the hardware runtime will cause the sound card to support no sample rates at all. Obviously this is not desired, just a different mechanism is being used to apply the constraints. As such update the setting of rate_max in dpcm_init_runtime_hw to be consistent with the non-DPCM cases and set rate_max to UINT_MAX if nothing is defined on the CPU DAI. Signed-off-by: NCharles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: NMark Brown <broonie@kernel.org>
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- 01 8月, 2018 1 次提交
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由 Akshu Agrawal 提交于
Take into account the base delay set in pointer callback. There are cases where a pointer function populates runtime->delay, such as: ./sound/pci/hda/hda_controller.c ./sound/soc/intel/atom/sst-mfld-platform-pcm.c This delay was getting lost and was overwritten by delays from codec or cpu dai delay function if exposed. Now, Total delay = base delay + cpu_dai delay + codec_dai delay Signed-off-by: NAkshu Agrawal <akshu.agrawal@amd.com> Reviewed-by: NTakashi Iwai <tiwai@suse.de> Signed-off-by: NMark Brown <broonie@kernel.org>
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