aboutsummaryrefslogtreecommitdiffstats
path: root/sound/soc/fsl/fsl-asoc-card.c
diff options
context:
space:
mode:
Diffstat (limited to 'sound/soc/fsl/fsl-asoc-card.c')
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c573
1 files changed, 573 insertions, 0 deletions
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
new file mode 100644
index 000000000000..cf3f1f47f1e8
--- /dev/null
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -0,0 +1,573 @@
1/*
2 * Freescale Generic ASoC Sound Card driver with ASRC
3 *
4 * Copyright (C) 2014 Freescale Semiconductor, Inc.
5 *
6 * Author: Nicolin Chen <nicoleotsuka@gmail.com>
7 *
8 * This file is licensed under the terms of the GNU General Public License
9 * version 2. This program is licensed "as is" without any warranty of any
10 * kind, whether express or implied.
11 */
12
13#include <linux/clk.h>
14#include <linux/i2c.h>
15#include <linux/module.h>
16#include <linux/of_platform.h>
17#include <sound/pcm_params.h>
18#include <sound/soc.h>
19
20#include "fsl_esai.h"
21#include "fsl_sai.h"
22#include "imx-audmux.h"
23
24#include "../codecs/sgtl5000.h"
25#include "../codecs/wm8962.h"
26
27#define RX 0
28#define TX 1
29
30/* Default DAI format without Master and Slave flag */
31#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
32
33/**
34 * CODEC private data
35 *
36 * @mclk_freq: Clock rate of MCLK
37 * @mclk_id: MCLK (or main clock) id for set_sysclk()
38 * @fll_id: FLL (or secordary clock) id for set_sysclk()
39 * @pll_id: PLL id for set_pll()
40 */
41struct codec_priv {
42 unsigned long mclk_freq;
43 u32 mclk_id;
44 u32 fll_id;
45 u32 pll_id;
46};
47
48/**
49 * CPU private data
50 *
51 * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
52 * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
53 * @sysclk_id[2]: SYSCLK ids for set_sysclk()
54 *
55 * Note: [1] for tx and [0] for rx
56 */
57struct cpu_priv {
58 unsigned long sysclk_freq[2];
59 u32 sysclk_dir[2];
60 u32 sysclk_id[2];
61};
62
63/**
64 * Freescale Generic ASOC card private data
65 *
66 * @dai_link[3]: DAI link structure including normal one and DPCM link
67 * @pdev: platform device pointer
68 * @codec_priv: CODEC private data
69 * @cpu_priv: CPU private data
70 * @card: ASoC card structure
71 * @sample_rate: Current sample rate
72 * @sample_format: Current sample format
73 * @asrc_rate: ASRC sample rate used by Back-Ends
74 * @asrc_format: ASRC sample format used by Back-Ends
75 * @dai_fmt: DAI format between CPU and CODEC
76 * @name: Card name
77 */
78
79struct fsl_asoc_card_priv {
80 struct snd_soc_dai_link dai_link[3];
81 struct platform_device *pdev;
82 struct codec_priv codec_priv;
83 struct cpu_priv cpu_priv;
84 struct snd_soc_card card;
85 u32 sample_rate;
86 u32 sample_format;
87 u32 asrc_rate;
88 u32 asrc_format;
89 u32 dai_fmt;
90 char name[32];
91};
92
93/**
94 * This dapm route map exsits for DPCM link only.
95 * The other routes shall go through Device Tree.
96 */
97static const struct snd_soc_dapm_route audio_map[] = {
98 {"CPU-Playback", NULL, "ASRC-Playback"},
99 {"Playback", NULL, "CPU-Playback"},
100 {"ASRC-Capture", NULL, "CPU-Capture"},
101 {"CPU-Capture", NULL, "Capture"},
102};
103
104/* Add all possible widgets into here without being redundant */
105static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
106 SND_SOC_DAPM_LINE("Line Out Jack", NULL),
107 SND_SOC_DAPM_LINE("Line In Jack", NULL),
108 SND_SOC_DAPM_HP("Headphone Jack", NULL),
109 SND_SOC_DAPM_SPK("Ext Spk", NULL),
110 SND_SOC_DAPM_MIC("Mic Jack", NULL),
111 SND_SOC_DAPM_MIC("AMIC", NULL),
112 SND_SOC_DAPM_MIC("DMIC", NULL),
113};
114
115static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
116 struct snd_pcm_hw_params *params)
117{
118 struct snd_soc_pcm_runtime *rtd = substream->private_data;
119 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
120 bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
121 struct cpu_priv *cpu_priv = &priv->cpu_priv;
122 struct device *dev = rtd->card->dev;
123 int ret;
124
125 priv->sample_rate = params_rate(params);
126 priv->sample_format = params_format(params);
127
128 if (priv->card.set_bias_level)
129 return 0;
130
131 /* Specific configurations of DAIs starts from here */
132 ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
133 cpu_priv->sysclk_freq[tx],
134 cpu_priv->sysclk_dir[tx]);
135 if (ret) {
136 dev_err(dev, "failed to set sysclk for cpu dai\n");
137 return ret;
138 }
139
140 return 0;
141}
142
143static struct snd_soc_ops fsl_asoc_card_ops = {
144 .hw_params = fsl_asoc_card_hw_params,
145};
146
147static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
148 struct snd_pcm_hw_params *params)
149{
150 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
151 struct snd_interval *rate;
152 struct snd_mask *mask;
153
154 rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
155 rate->max = rate->min = priv->asrc_rate;
156
157 mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
158 snd_mask_none(mask);
159 snd_mask_set(mask, priv->asrc_format);
160
161 return 0;
162}
163
164static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
165 /* Default ASoC DAI Link*/
166 {
167 .name = "HiFi",
168 .stream_name = "HiFi",
169 .ops = &fsl_asoc_card_ops,
170 },
171 /* DPCM Link between Front-End and Back-End (Optional) */
172 {
173 .name = "HiFi-ASRC-FE",
174 .stream_name = "HiFi-ASRC-FE",
175 .codec_name = "snd-soc-dummy",
176 .codec_dai_name = "snd-soc-dummy-dai",
177 .dpcm_playback = 1,
178 .dpcm_capture = 1,
179 .dynamic = 1,
180 },
181 {
182 .name = "HiFi-ASRC-BE",
183 .stream_name = "HiFi-ASRC-BE",
184 .platform_name = "snd-soc-dummy",
185 .be_hw_params_fixup = be_hw_params_fixup,
186 .ops = &fsl_asoc_card_ops,
187 .dpcm_playback = 1,
188 .dpcm_capture = 1,
189 .no_pcm = 1,
190 },
191};
192
193static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
194 struct snd_soc_dapm_context *dapm,
195 enum snd_soc_bias_level level)
196{
197 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
198 struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
199 struct codec_priv *codec_priv = &priv->codec_priv;
200 struct device *dev = card->dev;
201 unsigned int pll_out;
202 int ret;
203
204 if (dapm->dev != codec_dai->dev)
205 return 0;
206
207 switch (level) {
208 case SND_SOC_BIAS_PREPARE:
209 if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
210 break;
211
212 if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
213 pll_out = priv->sample_rate * 384;
214 else
215 pll_out = priv->sample_rate * 256;
216
217 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
218 codec_priv->mclk_id,
219 codec_priv->mclk_freq, pll_out);
220 if (ret) {
221 dev_err(dev, "failed to start FLL: %d\n", ret);
222 return ret;
223 }
224
225 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
226 pll_out, SND_SOC_CLOCK_IN);
227 if (ret) {
228 dev_err(dev, "failed to set SYSCLK: %d\n", ret);
229 return ret;
230 }
231 break;
232
233 case SND_SOC_BIAS_STANDBY:
234 if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
235 break;
236
237 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
238 codec_priv->mclk_freq,
239 SND_SOC_CLOCK_IN);
240 if (ret) {
241 dev_err(dev, "failed to switch away from FLL: %d\n", ret);
242 return ret;
243 }
244
245 ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
246 if (ret) {
247 dev_err(dev, "failed to stop FLL: %d\n", ret);
248 return ret;
249 }
250 break;
251
252 default:
253 break;
254 }
255
256 return 0;
257}
258
259static int fsl_asoc_card_audmux_init(struct device_node *np,
260 struct fsl_asoc_card_priv *priv)
261{
262 struct device *dev = &priv->pdev->dev;
263 u32 int_ptcr = 0, ext_ptcr = 0;
264 int int_port, ext_port;
265 int ret;
266
267 ret = of_property_read_u32(np, "mux-int-port", &int_port);
268 if (ret) {
269 dev_err(dev, "mux-int-port missing or invalid\n");
270 return ret;
271 }
272 ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
273 if (ret) {
274 dev_err(dev, "mux-ext-port missing or invalid\n");
275 return ret;
276 }
277
278 /*
279 * The port numbering in the hardware manual starts at 1, while
280 * the AUDMUX API expects it starts at 0.
281 */
282 int_port--;
283 ext_port--;
284
285 /*
286 * Use asynchronous mode (6 wires) for all cases.
287 * If only 4 wires are needed, just set SSI into
288 * synchronous mode and enable 4 PADs in IOMUX.
289 */
290 switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
291 case SND_SOC_DAIFMT_CBM_CFM:
292 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
293 IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
294 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
295 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
296 IMX_AUDMUX_V2_PTCR_RFSDIR |
297 IMX_AUDMUX_V2_PTCR_RCLKDIR |
298 IMX_AUDMUX_V2_PTCR_TFSDIR |
299 IMX_AUDMUX_V2_PTCR_TCLKDIR;
300 break;
301 case SND_SOC_DAIFMT_CBM_CFS:
302 int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
303 IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
304 IMX_AUDMUX_V2_PTCR_RCLKDIR |
305 IMX_AUDMUX_V2_PTCR_TCLKDIR;
306 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
307 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
308 IMX_AUDMUX_V2_PTCR_RFSDIR |
309 IMX_AUDMUX_V2_PTCR_TFSDIR;
310 break;
311 case SND_SOC_DAIFMT_CBS_CFM:
312 int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
313 IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
314 IMX_AUDMUX_V2_PTCR_RFSDIR |
315 IMX_AUDMUX_V2_PTCR_TFSDIR;
316 ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
317 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
318 IMX_AUDMUX_V2_PTCR_RCLKDIR |
319 IMX_AUDMUX_V2_PTCR_TCLKDIR;
320 break;
321 case SND_SOC_DAIFMT_CBS_CFS:
322 ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
323 IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
324 IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
325 IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
326 IMX_AUDMUX_V2_PTCR_RFSDIR |
327 IMX_AUDMUX_V2_PTCR_RCLKDIR |
328 IMX_AUDMUX_V2_PTCR_TFSDIR |
329 IMX_AUDMUX_V2_PTCR_TCLKDIR;
330 break;
331 default:
332 return -EINVAL;
333 }
334
335 /* Asynchronous mode can not be set along with RCLKDIR */
336 ret = imx_audmux_v2_configure_port(int_port, 0,
337 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
338 if (ret) {
339 dev_err(dev, "audmux internal port setup failed\n");
340 return ret;
341 }
342
343 ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
344 IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
345 if (ret) {
346 dev_err(dev, "audmux internal port setup failed\n");
347 return ret;
348 }
349
350 ret = imx_audmux_v2_configure_port(ext_port, 0,
351 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
352 if (ret) {
353 dev_err(dev, "audmux external port setup failed\n");
354 return ret;
355 }
356
357 ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
358 IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
359 if (ret) {
360 dev_err(dev, "audmux external port setup failed\n");
361 return ret;
362 }
363
364 return 0;
365}
366
367static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
368{
369 struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
370 struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
371 struct codec_priv *codec_priv = &priv->codec_priv;
372 struct device *dev = card->dev;
373 int ret;
374
375 ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
376 codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
377 if (ret) {
378 dev_err(dev, "failed to set sysclk in %s\n", __func__);
379 return ret;
380 }
381
382 return 0;
383}
384
385static int fsl_asoc_card_probe(struct platform_device *pdev)
386{
387 struct device_node *cpu_np, *codec_np, *asrc_np;
388 struct device_node *np = pdev->dev.of_node;
389 struct platform_device *asrc_pdev = NULL;
390 struct platform_device *cpu_pdev;
391 struct fsl_asoc_card_priv *priv;
392 struct i2c_client *codec_dev;
393 struct clk *codec_clk;
394 u32 width;
395 int ret;
396
397 priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
398 if (!priv)
399 return -ENOMEM;
400
401 cpu_np = of_parse_phandle(np, "audio-cpu", 0);
402 /* Give a chance to old DT binding */
403 if (!cpu_np)
404 cpu_np = of_parse_phandle(np, "ssi-controller", 0);
405 codec_np = of_parse_phandle(np, "audio-codec", 0);
406 if (!cpu_np || !codec_np) {
407 dev_err(&pdev->dev, "phandle missing or invalid\n");
408 ret = -EINVAL;
409 goto fail;
410 }
411
412 cpu_pdev = of_find_device_by_node(cpu_np);
413 if (!cpu_pdev) {
414 dev_err(&pdev->dev, "failed to find CPU DAI device\n");
415 ret = -EINVAL;
416 goto fail;
417 }
418
419 codec_dev = of_find_i2c_device_by_node(codec_np);
420 if (!codec_dev) {
421 dev_err(&pdev->dev, "failed to find codec platform device\n");
422 ret = -EINVAL;
423 goto fail;
424 }
425
426 asrc_np = of_parse_phandle(np, "audio-asrc", 0);
427 if (asrc_np)
428 asrc_pdev = of_find_device_by_node(asrc_np);
429
430 /* Get the MCLK rate only, and leave it controlled by CODEC drivers */
431 codec_clk = clk_get(&codec_dev->dev, NULL);
432 if (!IS_ERR(codec_clk)) {
433 priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
434 clk_put(codec_clk);
435 }
436
437 /* Default sample rate and format, will be updated in hw_params() */
438 priv->sample_rate = 44100;
439 priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
440
441 /* Assign a default DAI format, and allow each card to overwrite it */
442 priv->dai_fmt = DAI_FMT_BASE;
443
444 /* Diversify the card configurations */
445 if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
446 priv->card.set_bias_level = NULL;
447 priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
448 priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
449 priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
450 priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
451 priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
452 } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
453 priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
454 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
455 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
456 priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
457 priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
458 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
459 priv->codec_priv.pll_id = WM8962_FLL;
460 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
461 } else {
462 dev_err(&pdev->dev, "unknown Device Tree compatible\n");
463 return -EINVAL;
464 }
465
466 /* Common settings for corresponding Freescale CPU DAI driver */
467 if (strstr(cpu_np->name, "ssi")) {
468 /* Only SSI needs to configure AUDMUX */
469 ret = fsl_asoc_card_audmux_init(np, priv);
470 if (ret) {
471 dev_err(&pdev->dev, "failed to init audmux\n");
472 goto fail;
473 }
474 } else if (strstr(cpu_np->name, "esai")) {
475 priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
476 priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
477 } else if (strstr(cpu_np->name, "sai")) {
478 priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
479 priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
480 }
481
482 sprintf(priv->name, "%s-audio", codec_dev->name);
483
484 /* Initialize sound card */
485 priv->pdev = pdev;
486 priv->card.dev = &pdev->dev;
487 priv->card.name = priv->name;
488 priv->card.dai_link = priv->dai_link;
489 priv->card.dapm_routes = audio_map;
490 priv->card.late_probe = fsl_asoc_card_late_probe;
491 priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
492 priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
493 priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
494
495 memcpy(priv->dai_link, fsl_asoc_card_dai,
496 sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
497
498 /* Normal DAI Link */
499 priv->dai_link[0].cpu_of_node = cpu_np;
500 priv->dai_link[0].codec_of_node = codec_np;
501 priv->dai_link[0].codec_dai_name = codec_dev->name;
502 priv->dai_link[0].platform_of_node = cpu_np;
503 priv->dai_link[0].dai_fmt = priv->dai_fmt;
504 priv->card.num_links = 1;
505
506 if (asrc_pdev) {
507 /* DPCM DAI Links only if ASRC exsits */
508 priv->dai_link[1].cpu_of_node = asrc_np;
509 priv->dai_link[1].platform_of_node = asrc_np;
510 priv->dai_link[2].codec_dai_name = codec_dev->name;
511 priv->dai_link[2].codec_of_node = codec_np;
512 priv->dai_link[2].cpu_of_node = cpu_np;
513 priv->dai_link[2].dai_fmt = priv->dai_fmt;
514 priv->card.num_links = 3;
515
516 ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
517 &priv->asrc_rate);
518 if (ret) {
519 dev_err(&pdev->dev, "failed to get output rate\n");
520 ret = -EINVAL;
521 goto fail;
522 }
523
524 ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
525 if (ret) {
526 dev_err(&pdev->dev, "failed to get output rate\n");
527 ret = -EINVAL;
528 goto fail;
529 }
530
531 if (width == 24)
532 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
533 else
534 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
535 }
536
537 /* Finish card registering */
538 platform_set_drvdata(pdev, priv);
539 snd_soc_card_set_drvdata(&priv->card, priv);
540
541 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
542 if (ret)
543 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
544
545fail:
546 of_node_put(codec_np);
547 of_node_put(asrc_np);
548 of_node_put(cpu_np);
549
550 return ret;
551}
552
553static const struct of_device_id fsl_asoc_card_dt_ids[] = {
554 { .compatible = "fsl,imx-audio-cs42888", },
555 { .compatible = "fsl,imx-audio-sgtl5000", },
556 { .compatible = "fsl,imx-audio-wm8962", },
557 {}
558};
559
560static struct platform_driver fsl_asoc_card_driver = {
561 .probe = fsl_asoc_card_probe,
562 .driver = {
563 .name = "fsl-asoc-card",
564 .pm = &snd_soc_pm_ops,
565 .of_match_table = fsl_asoc_card_dt_ids,
566 },
567};
568module_platform_driver(fsl_asoc_card_driver);
569
570MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
571MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
572MODULE_ALIAS("platform:fsl-asoc-card");
573MODULE_LICENSE("GPL");
generated by cgit v1.2.2 (git 2.25.0) at 2025-09-14 05:49:10 -0400