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| author | Mark Brown <broonie@kernel.org> | 2024-12-13 20:33:09 +0300 |
|---|---|---|
| committer | Mark Brown <broonie@kernel.org> | 2024-12-13 20:33:09 +0300 |
| commit | 5ce3beed07b8145aff61f2cb41f1868f6221271f (patch) | |
| tree | f2496157a982db35fefb80543875272f84551f01 /include/uapi | |
| parent | 94c545aa535d7f5dcf54ad8e648f22943bbfcb32 (diff) | |
| parent | b62eaff0650dc6dc2a4bf0f50714f2357a23fc71 (diff) | |
| download | linux-5ce3beed07b8145aff61f2cb41f1868f6221271f.tar.xz | |
ASoC: fsl: add memory to memory function for ASRC
Merge series from Shengjiu Wang <shengjiu.wang@nxp.com>:
This function is base on the accelerator implementation
for compress API:
04177158cf98 ("ALSA: compress_offload: introduce accel operation mode")
Audio signal processing also has the requirement for memory to
memory similar as Video.
This asrc memory to memory (memory ->asrc->memory) case is a non
real time use case.
User fills the input buffer to the asrc module, after conversion, then asrc
sends back the output buffer to user. So it is not a traditional ALSA playback
and capture case.
Because we had implemented the "memory -> asrc ->i2s device-> codec"
use case in ALSA. Now the "memory->asrc->memory" needs
to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining
the code to make it can be shared by the "memory->asrc->memory"
driver.
Other change is to add memory to memory support for two kinds of i.MX ASRC
modules.
Diffstat (limited to 'include/uapi')
| -rw-r--r-- | include/uapi/sound/compress_params.h | 23 |
1 files changed, 21 insertions, 2 deletions
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index ddc77322d571..bc7648a30746 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -334,6 +334,14 @@ union snd_codec_options { struct snd_dec_wma wma_d; struct snd_dec_alac alac_d; struct snd_dec_ape ape_d; + struct { + __u32 out_sample_rate; + } src_d; +} __attribute__((packed, aligned(4))); + +struct snd_codec_desc_src { + __u32 out_sample_rate_min; + __u32 out_sample_rate_max; } __attribute__((packed, aligned(4))); /** struct snd_codec_desc - description of codec capabilities @@ -347,6 +355,9 @@ union snd_codec_options { * @modes: Supported modes. See SND_AUDIOMODE defines * @formats: Supported formats. See SND_AUDIOSTREAMFORMAT defines * @min_buffer: Minimum buffer size handled by codec implementation + * @pcm_formats: Output (for decoders) or input (for encoders) + * PCM formats (required to accel mode, 0 for other modes) + * @u_space: union space (for codec dependent data) * @reserved: reserved for future use * * This structure provides a scalar value for profiles, modes and stream @@ -370,7 +381,12 @@ struct snd_codec_desc { __u32 modes; __u32 formats; __u32 min_buffer; - __u32 reserved[15]; + __u32 pcm_formats; + union { + __u32 u_space[6]; + struct snd_codec_desc_src src; + } __attribute__((packed, aligned(4))); + __u32 reserved[8]; } __attribute__((packed, aligned(4))); /** struct snd_codec @@ -395,6 +411,8 @@ struct snd_codec_desc { * @align: Block alignment in bytes of an audio sample. * Only required for PCM or IEC formats. * @options: encoder-specific settings + * @pcm_format: Output (for decoders) or input (for encoders) + * PCM formats (required to accel mode, 0 for other modes) * @reserved: reserved for future use */ @@ -411,7 +429,8 @@ struct snd_codec { __u32 format; __u32 align; union snd_codec_options options; - __u32 reserved[3]; + __u32 pcm_format; + __u32 reserved[2]; } __attribute__((packed, aligned(4))); #endif |