This was done entirely with mindless brute force, using
git grep -l '\<k[vmz]*alloc_objs*(.*, GFP_KERNEL)' |
xargs sed -i 's/\(alloc_objs*(.*\), GFP_KERNEL)/\1)/'
to convert the new alloc_obj() users that had a simple GFP_KERNEL
argument to just drop that argument.
Note that due to the extreme simplicity of the scripting, any slightly
more complex cases spread over multiple lines would not be triggered:
they definitely exist, but this covers the vast bulk of the cases, and
the resulting diff is also then easier to check automatically.
For the same reason the 'flex' versions will be done as a separate
conversion.
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This is the result of running the Coccinelle script from
scripts/coccinelle/api/kmalloc_objs.cocci. The script is designed to
avoid scalar types (which need careful case-by-case checking), and
instead replace kmalloc-family calls that allocate struct or union
object instances:
Single allocations: kmalloc(sizeof(TYPE), ...)
are replaced with: kmalloc_obj(TYPE, ...)
Array allocations: kmalloc_array(COUNT, sizeof(TYPE), ...)
are replaced with: kmalloc_objs(TYPE, COUNT, ...)
Flex array allocations: kmalloc(struct_size(PTR, FAM, COUNT), ...)
are replaced with: kmalloc_flex(*PTR, FAM, COUNT, ...)
(where TYPE may also be *VAR)
The resulting allocations no longer return "void *", instead returning
"TYPE *".
Signed-off-by: Kees Cook <kees@kernel.org>
Add support for OPUS module, OPUS format ID, media format payload struct
and make it all recognizable by audioreach compress playback path.
At this moment this only supports raw or plain OPUS packets not
encapsulated in container (for instance OGG container). For this usecase
each OPUS packet needs to be prepended with 4-bytes long length field
which is expected to be done by userspace applications. This is
Qualcomm DSP specific requirement.
Cc: Annemarie Porter <annemari@quicinc.com>
Cc: Vinod Koul <vkoul@kernel.org>
Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Signed-off-by: Alexey Klimov <alexey.klimov@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The copied_total field in struct snd_compr_tstamp is a 32-bit
value that can overflow on long-running high-bitrate streams,
leading to incorrect calculations for buffer availablility.
This patch adds a 64-bit safe timestamping mechanism.
A new UAPI struct, snd_compr_tstamp64, is added which uses 64-bit
types for byte counters. The relevant ops structures across the
ASoC and core compress code are updated to use this new struct.
ASoC drivers are updated to use u64 counters.
Internal timestamps being u64 now, a compatibility function is added
to convert the 64-bit timestamp back to the 32-bit format for legacy
ioctl callers.
Reviewed-by: Miller Liang <millerliang@google.com>
Tested-by: Joris Verhaegen <verhaegen@google.com>
Signed-off-by: Joris Verhaegen <verhaegen@google.com>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@oss.qualcomm.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://patch.msgid.link/20250905091301.2711705-2-verhaegen@google.com
With the existing code, the buffer position is only reset in pointer
callback, which leaves the possiblity of it going over the size of
buffer size and reporting incorrect position to userspace.
Without this patch, its possible to see errors like:
snd-x1e80100 sound: invalid position: pcmC0D0p:0, pos = 12288, buffer size = 12288, period size = 1536
snd-x1e80100 sound: invalid position: pcmC0D0p:0, pos = 12288, buffer size = 12288, period size = 1536
Fixes: 9b4fe0f1cd ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Cc: stable@vger.kernel.org
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Tested-by: Johan Hovold <johan+linaro@kernel.org>
Link: https://patch.msgid.link/20250314174800.10142-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When constructing packets to DSP, the Audioreach code uses 'struct
audioreach_module_config' to configure parameters like number of
channels, bitrate, sample rate etc, but uses defaults for the channel
mapping.
Rework this code to copy the channel mapping from 'struct
audioreach_module_config', instead of using the default. This requires
all callers to fill that structure: add missing initialization of
channel mapping.
Entire patch makes code more logical and easier to follow:
1. q6apm-dai and q6apm-lpass-dais code which allocates 'struct
audioreach_module_config' initializes it fully, so fills both
the number of channels and the channel mapping.
2. Audioreach code, which uses 'struct audioreach_module_config' when
constructing packets, copies entire contents of passed config, not
only pieces of it.
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20240520-asoc-x1e80100-4-channel-mapping-v4-3-f657159b4aad@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
Static 'struct snd_pcm_hardware' is not modified by few drivers and its
copy is passed to the core, so it can be made const for increased code
safety.
Merge series from Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>:
Do not open-code snd_soc_substream_to_rtd() when accessing
snd_pcm_substream->private_data. This makes code more consistent with
rest of ASoC and allows in the future to move the field to any other
place or add additional checks in snd_soc_substream_to_rtd().
clang-16 points out a mismatch in function types that was hidden
by a typecast:
sound/soc/qcom/qdsp6/q6apm-dai.c:355:38: error: cast from 'void (*)(uint32_t, uint32_t, uint32_t *, void *)' (aka 'void (*)(unsigned int, unsigned int, unsigned int *, void *)') to 'q6apm_cb' (aka 'void (*)(unsigned int, unsigned int, void *, void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
355 | prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler, prtd, graph_id);
sound/soc/qcom/qdsp6/q6apm-dai.c:499:38: error: cast from 'void (*)(uint32_t, uint32_t, uint32_t *, void *)' (aka 'void (*)(unsigned int, unsigned int, unsigned int *, void *)') to 'q6apm_cb' (aka 'void (*)(unsigned int, unsigned int, void *, void *)') converts to incompatible function type [-Werror,-Wcast-function-type-strict]
499 | prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler_compr, prtd, graph_id);
The only difference here is the 'payload' argument, which is not even
used in this function, so just fix its type and remove the cast.
Fixes: 88b60bf047 ("ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://msgid.link/r/20240213101105.459402-1-arnd@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix few trivial code style issues, pointed out by checkpatch, so they do
not get copied to new code (when old code is used as template):
WARNING: Prefer "GPL" over "GPL v2" - see commit bf7fbeeae6 ("module: Cure the MODULE_LICENSE "GPL" vs. "GPL v2" bogosity")
WARNING: function definition argument 'struct platform_device *' should also have an identifier name
ERROR: code indent should use tabs where possible
WARNING: please, no spaces at the start of a line
WARNING: Missing a blank line after declarations
WARNING: unnecessary whitespace before a quoted newline
Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Link: https://msgid.link/r/20231204100048.211800-1-krzysztof.kozlowski@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The DT of_device.h and of_platform.h date back to the separate
of_platform_bus_type before it was merged into the regular platform bus.
As part of that merge prepping Arm DT support 13 years ago, they
"temporarily" include each other. They also include platform_device.h
and of.h. As a result, there's a pretty much random mix of those include
files used throughout the tree. In order to detangle these headers and
replace the implicit includes with struct declarations, users need to
explicitly include the correct includes.
Acked-by: Jernej Skrabec <jernej.skrabec@gmail.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: Claudiu Beznea <claudiu.beznea@tuxon.dev> # for at91
Signed-off-by: Rob Herring <robh@kernel.org>
Link: https://lore.kernel.org/r/20231006-dt-asoc-header-cleanups-v3-1-13a4f0f7fee6@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason we ended up with a setup without this flag.
This resulted in inconsistent sound card devices numbers which
are also not starting as expected at dai_link->id.
(Ex: MultiMedia1 pcm ended up with device number 4 instead of 0)
With this patch patch now the MultiMedia1 PCM ends up with device number 0
as expected.
[This is causing unstable numbering in userspace as other changes go in,
which in turn gets noticed by some userspace. There's been multiple
values so we can't simply pick one and revert to it. Do not backport
since it will introduce a change. -- broonie]
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230628092404.13927-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, playing audio with PulseAudio with the qdsp6 driver
results in distorted sound. It seems like its timer-based scheduling
does not work properly with qdsp6 since setting tsched=0 in
the PulseAudio configuration avoids the issue.
Apparently this happens when the pointer() callback is not accurate
enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop
PulseAudio from using timer-based scheduling by default.
According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html:
The flag is being used in the sense explained in the previous audio
meeting -- the data transfer granularity isn't fine enough but aligned
to the period size (or less).
q6apm-dai reports the position as multiple of
prtd->pcm_count = snd_pcm_lib_period_bytes(substream)
so it indeed just a multiple of the period size.
Therefore adding the flag here seems appropriate and makes audio
work out of the box.
Comment log inspired by Stephan Gerhold sent for q6asm-dai.c few years back.
Fixes: 9b4fe0f1cd ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20230209122806.18923-4-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Its possible that the sound card is just opened and closed without actually
playing stream, ex: if the audio file itself is missing.
Even in such cases we do call stop on graphs that are not yet started.
DSP can throw errors in such cases, so add a check to see if the graph
was started before stopping it.
Fixes: 9b4fe0f1cd ("ASoC: qdsp6: audioreach: add q6apm-dai support")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20220126113549.8853-5-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
