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-rw-r--r--sound/arm/Kconfig11
-rw-r--r--sound/arm/Makefile3
-rw-r--r--sound/arm/aaci.c6
-rw-r--r--sound/arm/pxa2xx-ac97-lib.c1
-rw-r--r--sound/arm/pxa2xx-ac97.c2
-rw-r--r--sound/arm/pxa2xx-pcm-lib.c3
-rw-r--r--sound/arm/sa11xx-uda1341.c984
7 files changed, 4 insertions, 1006 deletions
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index f8e6de48d816..885683a3b0bd 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -11,17 +11,6 @@ menuconfig SND_ARM
11 11
12if SND_ARM 12if SND_ARM
13 13
14config SND_SA11XX_UDA1341
15 tristate "SA11xx UDA1341TS driver (iPaq H3600)"
16 depends on ARCH_SA1100 && L3
17 select SND_PCM
18 help
19 Say Y here if you have a Compaq iPaq H3x00 handheld computer
20 and want to use its Philips UDA 1341 audio chip.
21
22 To compile this driver as a module, choose M here: the module
23 will be called snd-sa11xx-uda1341.
24
25config SND_ARMAACI 14config SND_ARMAACI
26 tristate "ARM PrimeCell PL041 AC Link support" 15 tristate "ARM PrimeCell PL041 AC Link support"
27 depends on ARM_AMBA 16 depends on ARM_AMBA
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
index 2054de11de8a..5a549ed6c8aa 100644
--- a/sound/arm/Makefile
+++ b/sound/arm/Makefile
@@ -2,9 +2,6 @@
2# Makefile for ALSA 2# Makefile for ALSA
3# 3#
4 4
5obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o
6snd-sa11xx-uda1341-objs := sa11xx-uda1341.o
7
8obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o 5obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o
9snd-aaci-objs := aaci.o devdma.o 6snd-aaci-objs := aaci.o devdma.o
10 7
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 7d39aac9ec14..7fbd68fab944 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
90 */ 90 */
91 do { 91 do {
92 v = readl(aaci->base + AACI_SLFR); 92 v = readl(aaci->base + AACI_SLFR);
93 } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--); 93 } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout);
94 94
95 if (!timeout) 95 if (!timeout)
96 dev_err(&aaci->dev->dev, 96 dev_err(&aaci->dev->dev,
@@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
126 */ 126 */
127 do { 127 do {
128 v = readl(aaci->base + AACI_SLFR); 128 v = readl(aaci->base + AACI_SLFR);
129 } while ((v & SLFR_1TXB) && timeout--); 129 } while ((v & SLFR_1TXB) && --timeout);
130 130
131 if (!timeout) { 131 if (!timeout) {
132 dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n"); 132 dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n");
@@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
147 do { 147 do {
148 cond_resched(); 148 cond_resched();
149 v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV); 149 v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV);
150 } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--); 150 } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout);
151 151
152 if (!timeout) { 152 if (!timeout) {
153 dev_err(&aaci->dev->dev, "timeout on RX valid\n"); 153 dev_err(&aaci->dev->dev, "timeout on RX valid\n");
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 71bef45e9d31..0afd1a8226fb 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -21,7 +21,6 @@
21#include <sound/pxa2xx-lib.h> 21#include <sound/pxa2xx-lib.h>
22 22
23#include <asm/irq.h> 23#include <asm/irq.h>
24#include <mach/hardware.h>
25#include <mach/regs-ac97.h> 24#include <mach/regs-ac97.h>
26#include <mach/pxa2xx-gpio.h> 25#include <mach/pxa2xx-gpio.h>
27#include <mach/audio.h> 26#include <mach/audio.h>
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 7ed100c80a5f..c570ebd9d177 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -20,8 +20,6 @@
20#include <sound/initval.h> 20#include <sound/initval.h>
21#include <sound/pxa2xx-lib.h> 21#include <sound/pxa2xx-lib.h>
22 22
23#include <mach/hardware.h>
24#include <mach/pxa-regs.h>
25#include <mach/regs-ac97.h> 23#include <mach/regs-ac97.h>
26#include <mach/audio.h> 24#include <mach/audio.h>
27 25
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 75a0d746fb60..108b643229ba 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -12,8 +12,7 @@
12#include <sound/pcm_params.h> 12#include <sound/pcm_params.h>
13#include <sound/pxa2xx-lib.h> 13#include <sound/pxa2xx-lib.h>
14 14
15#include <asm/dma.h> 15#include <mach/dma.h>
16#include <mach/pxa-regs.h>
17 16
18#include "pxa2xx-pcm.h" 17#include "pxa2xx-pcm.h"
19 18
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
deleted file mode 100644
index 51d708c31e65..000000000000
--- a/sound/arm/sa11xx-uda1341.c
+++ /dev/null
@@ -1,984 +0,0 @@
1/*
2 * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
3 * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License.
7 *
8 * History:
9 *
10 * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS
11 * 2002-03-20 Tomas Kasparek playback over ALSA is working
12 * 2002-03-28 Tomas Kasparek playback over OSS emulation is working
13 * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA)
14 * 2002-03-29 Tomas Kasparek capture is working (OSS emulation)
15 * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates)
16 * 2003-02-14 Brian Avery fixed full duplex mode, other updates
17 * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL)
18 * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
19 * working suspend and resume
20 * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again
21 * merged HAL layer (patches from Brian)
22 */
23
24/***************************************************************************************************
25*
26* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
27* available in the Alsa doc section on the website
28*
29* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
30* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
31* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
32* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
33* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
34* is a mem loc that always decodes to 0's w/ no off chip access.
35*
36* Some alsa terminology:
37* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes
38* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
39* buffer and 4 periods in the runtime structure this means we'll get an int every 256
40* bytes or 4 times per buffer.
41* A number of the sizes are in frames rather than bytes, use frames_to_bytes and
42* bytes_to_frames to convert. The easiest way to tell the units is to look at the
43* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
44*
45* Notes about the pointer fxn:
46* The pointer fxn needs to return the offset into the dma buffer in frames.
47* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
48*
49* Notes about pause/resume
50* Implementing this would be complicated so it's skipped. The problem case is:
51* A full duplex connection is going, then play is paused. At this point you need to start xmitting
52* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd
53* need to save off the dma info, and restore it properly on a resume. Yeach!
54*
55* Notes about transfer methods:
56* The async write calls fail. I probably need to implement something else to support them?
57*
58***************************************************************************************************/
59
60#include <linux/module.h>
61#include <linux/moduleparam.h>
62#include <linux/init.h>
63#include <linux/err.h>
64#include <linux/platform_device.h>
65#include <linux/errno.h>
66#include <linux/ioctl.h>
67#include <linux/delay.h>
68#include <linux/slab.h>
69
70#ifdef CONFIG_PM
71#include <linux/pm.h>
72#endif
73
74#include <mach/hardware.h>
75#include <mach/h3600.h>
76#include <asm/mach-types.h>
77#include <asm/dma.h>
78
79#include <sound/core.h>
80#include <sound/pcm.h>
81#include <sound/initval.h>
82
83#include <linux/l3/l3.h>
84
85#undef DEBUG_MODE
86#undef DEBUG_FUNCTION_NAMES
87#include <sound/uda1341.h>
88
89/*
90 * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
91 * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
92 * module for Familiar 0.6.1
93 */
94
95/* {{{ Type definitions */
96
97MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
98MODULE_LICENSE("GPL");
99MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
100MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
101
102static char *id; /* ID for this card */
103
104module_param(id, charp, 0444);
105MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
106
107struct audio_stream {
108 char *id; /* identification string */
109 int stream_id; /* numeric identification */
110 dma_device_t dma_dev; /* device identifier for DMA */
111#ifdef HH_VERSION
112 dmach_t dmach; /* dma channel identification */
113#else
114 dma_regs_t *dma_regs; /* points to our DMA registers */
115#endif
116 unsigned int active:1; /* we are using this stream for transfer now */
117 int period; /* current transfer period */
118 int periods; /* current count of periods registerd in the DMA engine */
119 int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */
120 unsigned int old_offset;
121 spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */
122 struct snd_pcm_substream *stream;
123};
124
125struct sa11xx_uda1341 {
126 struct snd_card *card;
127 struct l3_client *uda1341;
128 struct snd_pcm *pcm;
129 long samplerate;
130 struct audio_stream s[2]; /* playback & capture */
131};
132
133static unsigned int rates[] = {
134 8000, 10666, 10985, 14647,
135 16000, 21970, 22050, 24000,
136 29400, 32000, 44100, 48000,
137};
138
139static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
140 .count = ARRAY_SIZE(rates),
141 .list = rates,
142 .mask = 0,
143};
144
145static struct platform_device *device;
146
147/* }}} */
148
149/* {{{ Clock and sample rate stuff */
150
151/*
152 * Stop-gap solution until rest of hh.org HAL stuff is merged.
153 */
154#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12)
155#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13)
156
157#ifdef CONFIG_SA1100_H3XXX
158#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x)
159#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x)
160#else
161#error This driver could serve H3x00 handhelds only!
162#endif
163
164static void sa11xx_uda1341_set_audio_clock(long val)
165{
166 switch (val) {
167 case 24000: case 32000: case 48000: /* 00: 12.288 MHz */
168 GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
169 break;
170
171 case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */
172 GPSR = GPIO_H3600_CLK_SET0;
173 GPCR = GPIO_H3600_CLK_SET1;
174 break;
175
176 case 8000: case 10666: case 16000: /* 10: 4.096 MHz */
177 GPCR = GPIO_H3600_CLK_SET0;
178 GPSR = GPIO_H3600_CLK_SET1;
179 break;
180
181 case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */
182 GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
183 break;
184 }
185}
186
187static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
188{
189 int clk_div = 0;
190 int clk=0;
191
192 /* We don't want to mess with clocks when frames are in flight */
193 Ser4SSCR0 &= ~SSCR0_SSE;
194 /* wait for any frame to complete */
195 udelay(125);
196
197 /*
198 * We have the following clock sources:
199 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
200 * Those can be divided either by 256, 384 or 512.
201 * This makes up 12 combinations for the following samplerates...
202 */
203 if (rate >= 48000)
204 rate = 48000;
205 else if (rate >= 44100)
206 rate = 44100;
207 else if (rate >= 32000)
208 rate = 32000;
209 else if (rate >= 29400)
210 rate = 29400;
211 else if (rate >= 24000)
212 rate = 24000;
213 else if (rate >= 22050)
214 rate = 22050;
215 else if (rate >= 21970)
216 rate = 21970;
217 else if (rate >= 16000)
218 rate = 16000;
219 else if (rate >= 14647)
220 rate = 14647;
221 else if (rate >= 10985)
222 rate = 10985;
223 else if (rate >= 10666)
224 rate = 10666;
225 else
226 rate = 8000;
227
228 /* Set the external clock generator */
229
230 sa11xx_uda1341_set_audio_clock(rate);
231
232 /* Select the clock divisor */
233 switch (rate) {
234 case 8000:
235 case 10985:
236 case 22050:
237 case 24000:
238 clk = F512;
239 clk_div = SSCR0_SerClkDiv(16);
240 break;
241 case 16000:
242 case 21970:
243 case 44100:
244 case 48000:
245 clk = F256;
246 clk_div = SSCR0_SerClkDiv(8);
247 break;
248 case 10666:
249 case 14647:
250 case 29400:
251 case 32000:
252 clk = F384;
253 clk_div = SSCR0_SerClkDiv(12);
254 break;
255 }
256
257 /* FMT setting should be moved away when other FMTs are added (FIXME) */
258 l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
259
260 l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);
261 Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
262 sa11xx_uda1341->samplerate = rate;
263}
264
265/* }}} */
266
267/* {{{ HW init and shutdown */
268
269static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
270{
271 unsigned long flags;
272
273 /* Setup DMA stuff */
274 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
275 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
276 sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
277
278 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
279 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
280 sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
281
282 /* Initialize the UDA1341 internal state */
283
284 /* Setup the uarts */
285 local_irq_save(flags);
286 GAFR |= (GPIO_SSP_CLK);
287 GPDR &= ~(GPIO_SSP_CLK);
288 Ser4SSCR0 = 0;
289 Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
290 Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
291 Ser4SSCR0 |= SSCR0_SSE;
292 local_irq_restore(flags);
293
294 /* Enable the audio power */
295
296 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
297 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
298 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
299
300 /* Wait for the UDA1341 to wake up */
301 mdelay(1); //FIXME - was removed by Perex - Why?
302
303 /* Initialize the UDA1341 internal state */
304 l3_open(sa11xx_uda1341->uda1341);
305
306 /* external clock configuration (after l3_open - regs must be initialized */
307 sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
308
309 /* Wait for the UDA1341 to wake up */
310 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
311 mdelay(1);
312
313 /* make the left and right channels unswapped (flip the WS latch) */
314 Ser4SSDR = 0;
315
316 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
317}
318
319static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
320{
321 /* mute on */
322 set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
323
324 /* disable the audio power and all signals leading to the audio chip */
325 l3_close(sa11xx_uda1341->uda1341);
326 Ser4SSCR0 = 0;
327 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
328
329 /* power off and mute off */
330 /* FIXME - is muting off necesary??? */
331
332 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
333 clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
334}
335
336/* }}} */
337
338/* {{{ DMA staff */
339
340/*
341 * these are the address and sizes used to fill the xmit buffer
342 * so we can get a clock in record only mode
343 */
344#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS
345#define FORCE_CLOCK_SIZE 4096 // was 2048
346
347// FIXME Why this value exactly - wrote comment
348#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
349
350#ifdef HH_VERSION
351
352static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
353{
354 int ret;
355
356 ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
357 if (ret < 0) {
358 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
359 return ret;
360 }
361 sa1100_dma_set_callback(s->dmach, callback);
362 return 0;
363}
364
365static inline void audio_dma_free(struct audio_stream *s)
366{
367 sa1100_free_dma(s->dmach);
368 s->dmach = -1;
369}
370
371#else
372
373static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
374{
375 int ret;
376
377 ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
378 if (ret < 0)
379 printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
380 return ret;
381}
382
383static void audio_dma_free(struct audio_stream *s)
384{
385 sa1100_free_dma(s->dma_regs);
386 s->dma_regs = 0;
387}
388
389#endif
390
391static u_int audio_get_dma_pos(struct audio_stream *s)
392{
393 struct snd_pcm_substream *substream = s->stream;
394 struct snd_pcm_runtime *runtime = substream->runtime;
395 unsigned int offset;
396 unsigned long flags;
397 dma_addr_t addr;
398
399 // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
400 spin_lock_irqsave(&s->dma_lock, flags);
401#ifdef HH_VERSION
402 sa1100_dma_get_current(s->dmach, NULL, &addr);
403#else
404 addr = sa1100_get_dma_pos((s)->dma_regs);
405#endif
406 offset = addr - runtime->dma_addr;
407 spin_unlock_irqrestore(&s->dma_lock, flags);
408
409 offset = bytes_to_frames(runtime,offset);
410 if (offset >= runtime->buffer_size)
411 offset = 0;
412
413 return offset;
414}
415
416/*
417 * this stops the dma and clears the dma ptrs
418 */
419static void audio_stop_dma(struct audio_stream *s)
420{
421 unsigned long flags;
422
423 spin_lock_irqsave(&s->dma_lock, flags);
424 s->active = 0;
425 s->period = 0;
426 /* this stops the dma channel and clears the buffer ptrs */
427#ifdef HH_VERSION
428 sa1100_dma_flush_all(s->dmach);
429#else
430 sa1100_clear_dma(s->dma_regs);
431#endif
432 spin_unlock_irqrestore(&s->dma_lock, flags);
433}
434
435static void audio_process_dma(struct audio_stream *s)
436{
437 struct snd_pcm_substream *substream = s->stream;
438 struct snd_pcm_runtime *runtime;
439 unsigned int dma_size;
440 unsigned int offset;
441 int ret;
442
443 /* we are requested to process synchronization DMA transfer */
444 if (s->tx_spin) {
445 if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK))
446 return;
447 /* fill the xmit dma buffers and return */
448#ifdef HH_VERSION
449 sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
450#else
451 while (1) {
452 ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
453 if (ret)
454 return;
455 }
456#endif
457 return;
458 }
459
460 /* must be set here - only valid for running streams, not for forced_clock dma fills */
461 runtime = substream->runtime;
462 while (s->active && s->periods < runtime->periods) {
463 dma_size = frames_to_bytes(runtime, runtime->period_size);
464 if (s->old_offset) {
465 /* a little trick, we need resume from old position */
466 offset = frames_to_bytes(runtime, s->old_offset - 1);
467 s->old_offset = 0;
468 s->periods = 0;
469 s->period = offset / dma_size;
470 offset %= dma_size;
471 dma_size = dma_size - offset;
472 if (!dma_size)
473 continue; /* special case */
474 } else {
475 offset = dma_size * s->period;
476 snd_BUG_ON(dma_size > DMA_BUF_SIZE);
477 }
478#ifdef HH_VERSION
479 ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
480 if (ret)
481 return; //FIXME
482#else
483 ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
484 if (ret) {
485 printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
486 return;
487 }
488#endif
489
490 s->period++;
491 s->period %= runtime->periods;
492 s->periods++;
493 }
494}
495
496#ifdef HH_VERSION
497static void audio_dma_callback(void *data, int size)
498#else
499static void audio_dma_callback(void *data)
500#endif
501{
502 struct audio_stream *s = data;
503
504 /*
505 * If we are getting a callback for an active stream then we inform
506 * the PCM middle layer we've finished a period
507 */
508 if (s->active)
509 snd_pcm_period_elapsed(s->stream);
510
511 spin_lock(&s->dma_lock);
512 if (!s->tx_spin && s->periods > 0)
513 s->periods--;
514 audio_process_dma(s);
515 spin_unlock(&s->dma_lock);
516}
517
518/* }}} */
519
520/* {{{ PCM setting */
521
522/* {{{ trigger & timer */
523
524static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
525{
526 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
527 int stream_id = substream->pstr->stream;
528 struct audio_stream *s = &chip->s[stream_id];
529 struct audio_stream *s1 = &chip->s[stream_id ^ 1];
530 int err = 0;
531
532 /* note local interrupts are already disabled in the midlevel code */
533 spin_lock(&s->dma_lock);
534 switch (cmd) {
535 case SNDRV_PCM_TRIGGER_START:
536 /* now we need to make sure a record only stream has a clock */
537 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
538 /* we need to force fill the xmit DMA with zeros */
539 s1->tx_spin = 1;
540 audio_process_dma(s1);
541 }
542 /* this case is when you were recording then you turn on a
543 * playback stream so we stop (also clears it) the dma first,
544 * clear the sync flag and then we let it turned on
545 */
546 else {
547 s->tx_spin = 0;
548 }
549
550 /* requested stream startup */
551 s->active = 1;
552 audio_process_dma(s);
553 break;
554 case SNDRV_PCM_TRIGGER_STOP:
555 /* requested stream shutdown */
556 audio_stop_dma(s);
557
558 /*
559 * now we need to make sure a record only stream has a clock
560 * so if we're stopping a playback with an active capture
561 * we need to turn the 0 fill dma on for the xmit side
562 */
563 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
564 /* we need to force fill the xmit DMA with zeros */
565 s->tx_spin = 1;
566 audio_process_dma(s);
567 }
568 /*
569 * we killed a capture only stream, so we should also kill
570 * the zero fill transmit
571 */
572 else {
573 if (s1->tx_spin) {
574 s1->tx_spin = 0;
575 audio_stop_dma(s1);
576 }
577 }
578
579 break;
580 case SNDRV_PCM_TRIGGER_SUSPEND:
581 s->active = 0;
582#ifdef HH_VERSION
583 sa1100_dma_stop(s->dmach);
584#else
585 //FIXME - DMA API
586#endif
587 s->old_offset = audio_get_dma_pos(s) + 1;
588#ifdef HH_VERSION
589 sa1100_dma_flush_all(s->dmach);
590#else
591 //FIXME - DMA API
592#endif
593 s->periods = 0;
594 break;
595 case SNDRV_PCM_TRIGGER_RESUME:
596 s->active = 1;
597 s->tx_spin = 0;
598 audio_process_dma(s);
599 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
600 s1->tx_spin = 1;
601 audio_process_dma(s1);
602 }
603 break;
604 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
605#ifdef HH_VERSION
606 sa1100_dma_stop(s->dmach);
607#else
608 //FIXME - DMA API
609#endif
610 s->active = 0;
611 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
612 if (s1->active) {
613 s->tx_spin = 1;
614 s->old_offset = audio_get_dma_pos(s) + 1;
615#ifdef HH_VERSION
616 sa1100_dma_flush_all(s->dmach);
617#else
618 //FIXME - DMA API
619#endif
620 audio_process_dma(s);
621 }
622 } else {
623 if (s1->tx_spin) {
624 s1->tx_spin = 0;
625#ifdef HH_VERSION
626 sa1100_dma_flush_all(s1->dmach);
627#else
628 //FIXME - DMA API
629#endif
630 }
631 }
632 break;
633 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
634 s->active = 1;
635 if (s->old_offset) {
636 s->tx_spin = 0;
637 audio_process_dma(s);
638 break;
639 }
640 if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
641 s1->tx_spin = 1;
642 audio_process_dma(s1);
643 }
644#ifdef HH_VERSION
645 sa1100_dma_resume(s->dmach);
646#else
647 //FIXME - DMA API
648#endif
649 break;
650 default:
651 err = -EINVAL;
652 break;
653 }
654 spin_unlock(&s->dma_lock);
655 return err;
656}
657
658static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
659{
660 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
661 struct snd_pcm_runtime *runtime = substream->runtime;
662 struct audio_stream *s = &chip->s[substream->pstr->stream];
663
664 /* set requested samplerate */
665 sa11xx_uda1341_set_samplerate(chip, runtime->rate);
666
667 /* set requestd format when available */
668 /* set FMT here !!! FIXME */
669
670 s->period = 0;
671 s->periods = 0;
672
673 return 0;
674}
675
676static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
677{
678 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
679 return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
680}
681
682/* }}} */
683
684static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
685{
686 .info = (SNDRV_PCM_INFO_INTERLEAVED |
687 SNDRV_PCM_INFO_BLOCK_TRANSFER |
688 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
689 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
690 .formats = SNDRV_PCM_FMTBIT_S16_LE,
691 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
692 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
693 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
694 SNDRV_PCM_RATE_KNOT),
695 .rate_min = 8000,
696 .rate_max = 48000,
697 .channels_min = 2,
698 .channels_max = 2,
699 .buffer_bytes_max = 64*1024,
700 .period_bytes_min = 64,
701 .period_bytes_max = DMA_BUF_SIZE,
702 .periods_min = 2,
703 .periods_max = 255,
704 .fifo_size = 0,
705};
706
707static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
708{
709 .info = (SNDRV_PCM_INFO_INTERLEAVED |
710 SNDRV_PCM_INFO_BLOCK_TRANSFER |
711 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
712 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
713 .formats = SNDRV_PCM_FMTBIT_S16_LE,
714 .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
715 SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
716 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
717 SNDRV_PCM_RATE_KNOT),
718 .rate_min = 8000,
719 .rate_max = 48000,
720 .channels_min = 2,
721 .channels_max = 2,
722 .buffer_bytes_max = 64*1024,
723 .period_bytes_min = 64,
724 .period_bytes_max = DMA_BUF_SIZE,
725 .periods_min = 2,
726 .periods_max = 255,
727 .fifo_size = 0,
728};
729
730static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
731{
732 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
733 struct snd_pcm_runtime *runtime = substream->runtime;
734 int stream_id = substream->pstr->stream;
735 int err;
736
737 chip->s[stream_id].stream = substream;
738
739 if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
740 runtime->hw = snd_sa11xx_uda1341_playback;
741 else
742 runtime->hw = snd_sa11xx_uda1341_capture;
743 if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
744 return err;
745 if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
746 return err;
747
748 return 0;
749}
750
751static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
752{
753 struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
754
755 chip->s[substream->pstr->stream].stream = NULL;
756 return 0;
757}
758
759/* {{{ HW params & free */
760
761static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
762 struct snd_pcm_hw_params *hw_params)
763{
764
765 return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
766}
767
768static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
769{
770 return snd_pcm_lib_free_pages(substream);
771}
772
773/* }}} */
774
775static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
776 .open = snd_card_sa11xx_uda1341_open,
777 .close = snd_card_sa11xx_uda1341_close,
778 .ioctl = snd_pcm_lib_ioctl,
779 .hw_params = snd_sa11xx_uda1341_hw_params,
780 .hw_free = snd_sa11xx_uda1341_hw_free,
781 .prepare = snd_sa11xx_uda1341_prepare,
782 .trigger = snd_sa11xx_uda1341_trigger,
783 .pointer = snd_sa11xx_uda1341_pointer,
784};
785
786static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
787 .open = snd_card_sa11xx_uda1341_open,
788 .close = snd_card_sa11xx_uda1341_close,
789 .ioctl = snd_pcm_lib_ioctl,
790 .hw_params = snd_sa11xx_uda1341_hw_params,
791 .hw_free = snd_sa11xx_uda1341_hw_free,
792 .prepare = snd_sa11xx_uda1341_prepare,
793 .trigger = snd_sa11xx_uda1341_trigger,
794 .pointer = snd_sa11xx_uda1341_pointer,
795};
796
797static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
798{
799 struct snd_pcm *pcm;
800 int err;
801
802 if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
803 return err;
804
805 /*
806 * this sets up our initial buffers and sets the dma_type to isa.
807 * isa works but I'm not sure why (or if) it's the right choice
808 * this may be too large, trying it for now
809 */
810 snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
811 snd_dma_isa_data(),
812 64*1024, 64*1024);
813
814 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
815 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
816 pcm->private_data = sa11xx_uda1341;
817 pcm->info_flags = 0;
818 strcpy(pcm->name, "UDA1341 PCM");
819
820 sa11xx_uda1341_audio_init(sa11xx_uda1341);
821
822 /* setup DMA controller */
823 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
824 audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
825
826 sa11xx_uda1341->pcm = pcm;
827
828 return 0;
829}
830
831/* }}} */
832
833/* {{{ module init & exit */
834
835#ifdef CONFIG_PM
836
837static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
838 pm_message_t state)
839{
840 struct snd_card *card = platform_get_drvdata(devptr);
841 struct sa11xx_uda1341 *chip = card->private_data;
842
843 snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
844 snd_pcm_suspend_all(chip->pcm);
845#ifdef HH_VERSION
846 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
847 sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
848#else
849 //FIXME
850#endif
851 l3_command(chip->uda1341, CMD_SUSPEND, NULL);
852 sa11xx_uda1341_audio_shutdown(chip);
853
854 return 0;
855}
856
857static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
858{
859 struct snd_card *card = platform_get_drvdata(devptr);
860 struct sa11xx_uda1341 *chip = card->private_data;
861
862 sa11xx_uda1341_audio_init(chip);
863 l3_command(chip->uda1341, CMD_RESUME, NULL);
864#ifdef HH_VERSION
865 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
866 sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
867#else
868 //FIXME
869#endif
870 snd_power_change_state(card, SNDRV_CTL_POWER_D0);
871 return 0;
872}
873#endif /* COMFIG_PM */
874
875void snd_sa11xx_uda1341_free(struct snd_card *card)
876{
877 struct sa11xx_uda1341 *chip = card->private_data;
878
879 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
880 audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
881}
882
883static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr)
884{
885 int err;
886 struct snd_card *card;
887 struct sa11xx_uda1341 *chip;
888
889 /* register the soundcard */
890 err = snd_card_create(-1, id, THIS_MODULE,
891 sizeof(struct sa11xx_uda1341), &card);
892 if (err < 0)
893 return err;
894
895 chip = card->private_data;
896 spin_lock_init(&chip->s[0].dma_lock);
897 spin_lock_init(&chip->s[1].dma_lock);
898
899 card->private_free = snd_sa11xx_uda1341_free;
900 chip->card = card;
901 chip->samplerate = AUDIO_RATE_DEFAULT;
902
903 // mixer
904 if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
905 goto nodev;
906
907 // PCM
908 if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
909 goto nodev;
910
911 strcpy(card->driver, "UDA1341");
912 strcpy(card->shortname, "H3600 UDA1341TS");
913 sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
914
915 snd_card_set_dev(card, &devptr->dev);
916
917 if ((err = snd_card_register(card)) == 0) {
918 printk( KERN_INFO "iPAQ audio support initialized\n" );
919 platform_set_drvdata(devptr, card);
920 return 0;
921 }
922
923 nodev:
924 snd_card_free(card);
925 return err;
926}
927
928static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
929{
930 snd_card_free(platform_get_drvdata(devptr));
931 platform_set_drvdata(devptr, NULL);
932 return 0;
933}
934
935#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341"
936
937static struct platform_driver sa11xx_uda1341_driver = {
938 .probe = sa11xx_uda1341_probe,
939 .remove = __devexit_p(sa11xx_uda1341_remove),
940#ifdef CONFIG_PM
941 .suspend = snd_sa11xx_uda1341_suspend,
942 .resume = snd_sa11xx_uda1341_resume,
943#endif
944 .driver = {
945 .name = SA11XX_UDA1341_DRIVER,
946 },
947};
948
949static int __init sa11xx_uda1341_init(void)
950{
951 int err;
952
953 if (!machine_is_h3xxx())
954 return -ENODEV;
955 if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
956 return err;
957 device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
958 if (!IS_ERR(device)) {
959 if (platform_get_drvdata(device))
960 return 0;
961 platform_device_unregister(device);
962 err = -ENODEV;
963 } else
964 err = PTR_ERR(device);
965 platform_driver_unregister(&sa11xx_uda1341_driver);
966 return err;
967}
968
969static void __exit sa11xx_uda1341_exit(void)
970{
971 platform_device_unregister(device);
972 platform_driver_unregister(&sa11xx_uda1341_driver);
973}
974
975module_init(sa11xx_uda1341_init);
976module_exit(sa11xx_uda1341_exit);
977
978/* }}} */
979
980/*
981 * Local variables:
982 * indent-tabs-mode: t
983 * End:
984 */
generated by cgit v1.2.2 (git 2.25.0) at 2024-11-28 06:42:43 -0500