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| author | Linus Torvalds <[email protected]> | 2019-07-09 09:59:43 -0700 |
|---|---|---|
| committer | Linus Torvalds <[email protected]> | 2019-07-09 09:59:43 -0700 |
| commit | 4cdd5f9186bbe80306e76f11da7ecb0b9720433c (patch) | |
| tree | 23c2f39933cd8253a65385eab00405beaf602f01 /sound/soc/qcom/common.c | |
| parent | 2d41ef5432b76ae90dc0db93026f1d981f874ec4 (diff) | |
| parent | 0dcb4efb1095d0a1f5f681c2b94e98b009cc5d77 (diff) | |
Merge tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Many updates in this development cycle are found in ASoC where it got
a wide range of changes for the continued refactoring.
Some highlights are below.
ASoC:
- Continued refactoring work by Morimoto-san toward the full
componentization; the changes are seen allover the places
- Support for force disconnecting muxes in DAPM
- Continued development of ASoC Intel SOF stuff
- New drivers for Cirrus Logic CS47L35, CS47L85 and CS47L90, Conexant
CX2072X, Realtek RT1011 and RT1308
HD-audio:
- More fixes and adjustments for ASoC SOF HD-audio
- Fix for resume problem on some Realtek codecs
USB-audio:
- A few fixes for the issues reported by syzbot USB fuzzer
- Fix for UAC2 extension unit parser
- Quirks for Line6 Helix, Emgaic Unitor 8
FireWire:
- Lots of code refactoring and fixes in most of its components"
* tag 'sound-5.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (626 commits)
ALSA: firewire-lib: code refactoring for local variables
ALSA: firewire-lib: code refactoring for post operation to data block counter
ALSA: firewire-lib: code refactoring for error path of parser for CIP header
ALSA: firewire-lib: fix different data block counter between probed event and transferred isochronous packet
ALSA: firewire-lib: fix initial value of data block count for IR context without CIP_DBC_IS_END_EVENT
ALSA: firewire-lib/fireface: fix initial value of data block counter for IR context with CIP_NO_HEADER
ALSA: firewire-lib: fix invalid length of rx packet payload for tracepoint events
ALSA: usb-audio: fix Line6 Helix audio format rates
firewire-motu: fix wrong reference count for stream functionality at error path of rawmidi interface
ALSA: firewire-digi00x: fix wrong reference count for stream functionality at error path of rawmidi interface
ALSA: dice: fix wrong reference count for stream functionality at error path of rawmidi interface
ALSA: oxfw: fix wrong reference count for stream functionality at error path of rawmidi interface
ALSA: fireworks: fix wrong reference count for stream functionality at error path of rawmidi interface
ALSA: bebob: fix wrong reference count for stream functionality at error path of rawmidi interface
ASoC: SOF: Intel: implement runtime idle for CNL/APL
ASoC: SOF: add runtime idle callback
ASoC: hdac_hdmi: report codec link up/down status to bus
ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write()
ASoC: sunxi: sun50i-codec-analog: Add earpiece
ASoC: rt5665: remove redundant assignment to variable idx
...
Diffstat (limited to 'sound/soc/qcom/common.c')
| -rw-r--r-- | sound/soc/qcom/common.c | 34 |
1 files changed, 27 insertions, 7 deletions
diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 5661025e8cec..2c7348ddbbb3 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -14,6 +14,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) struct device *dev = card->dev; struct snd_soc_dai_link *link; struct of_phandle_args args; + struct snd_soc_dai_link_component *dlc; int ret, num_links; ret = snd_soc_of_parse_card_name(card, "model"); @@ -40,7 +41,18 @@ int qcom_snd_parse_of(struct snd_soc_card *card) card->num_links = num_links; link = card->dai_link; + for_each_child_of_node(dev->of_node, np) { + dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return -ENOMEM; + + link->cpus = &dlc[0]; + link->platforms = &dlc[1]; + + link->num_cpus = 1; + link->num_platforms = 1; + cpu = of_get_child_by_name(np, "cpu"); platform = of_get_child_by_name(np, "platform"); codec = of_get_child_by_name(np, "codec"); @@ -57,20 +69,20 @@ int qcom_snd_parse_of(struct snd_soc_card *card) dev_err(card->dev, "error getting cpu phandle\n"); goto err; } - link->cpu_of_node = args.np; + link->cpus->of_node = args.np; link->id = args.args[0]; - ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); + ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name); if (ret) { dev_err(card->dev, "error getting cpu dai name\n"); goto err; } if (codec && platform) { - link->platform_of_node = of_parse_phandle(platform, + link->platforms->of_node = of_parse_phandle(platform, "sound-dai", 0); - if (!link->platform_of_node) { + if (!link->platforms->of_node) { dev_err(card->dev, "platform dai not found\n"); ret = -EINVAL; goto err; @@ -84,9 +96,16 @@ int qcom_snd_parse_of(struct snd_soc_card *card) link->no_pcm = 1; link->ignore_pmdown_time = 1; } else { - link->platform_of_node = link->cpu_of_node; - link->codec_dai_name = "snd-soc-dummy-dai"; - link->codec_name = "snd-soc-dummy"; + dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL); + if (!dlc) + return -ENOMEM; + + link->codecs = dlc; + link->num_codecs = 1; + + link->platforms->of_node = link->cpus->of_node; + link->codecs->dai_name = "snd-soc-dummy-dai"; + link->codecs->name = "snd-soc-dummy"; link->dynamic = 1; } @@ -97,6 +116,7 @@ int qcom_snd_parse_of(struct snd_soc_card *card) goto err; } + link->nonatomic = 1; link->dpcm_playback = 1; link->dpcm_capture = 1; link->stream_name = link->name; |