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authorLinus Torvalds <torvalds@linux-foundation.org>2013-07-03 19:52:22 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2013-07-03 19:52:22 -0700
commit1286da8bc009cb2aee7f285e94623fc974c0c983 (patch)
tree51ec0a79c3de63fa809b831ae0cbb5b85e44482f /sound/soc/davinci/davinci-sffsdr.c
parent9e220385c4eb8b7e66174a60ea0e15b6b296f228 (diff)
parent1ba65ae4bdbd43265c51ee4c30ff21a48124b6d8 (diff)
Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "A relative calm release at this time with a flat diffstat. The only significant change in the ALSA core side is the support for more than 32 card instances, configurable via kconfig. Other than that, in both ASoC and other parts, mostly some improvements and fixes on the driver side. - hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes - hda: Haswell HDMI audio fixes, runtime PM improvements - hda: Intel BayTrail support, ALC5505 DSP support - es1968: MediaForte M56VAP support - usb-audio: Improved support for Yamaha/Roland devices - usb-audio: M2Tech hiFace, Audio Advantage Micro II support - hdspm: wordclock fixes - ASoC: Pending fixes for WM8962 - ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500 - ASoC: Generalisation of the Bluetooth and HDMI stub drivers - ASoC: SSM2518 and RT5640 codec drivers. - ASoC: Tegra CPUs with RT5640 machine driver - ASoC: AC'97 refactoring bug fixes - ASoC: ADAU1701 driver fixes - Clean up of *_set_drvdata() in a wide range of drivers" * tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits) ALSA: vmaster: Fix the regression of missing vmaster hook call ALSA: hda - Add Dell SSID to support Headset Mic recording ASoC: adau1701: remove control_data assignment ASoC: adau1701: more direct regmap usage ASoC: ac97: fixup multi-platform AC'97 module build failure ASoC: pxa2xx: fixup multi-platform AC'97 build failures ASoC: tegra20-ac97: Remove unused variable ASoC: tegra20-ac97: Remove duplicate error message ALSA: usb-audio: Add Audio Advantage Micro II ASoC: tas5086: fix Mid-Z implementation ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size ALSA: Replace the magic number 44 with const ALSA: hda - Fix the max length of control name in generic parser ALSA: hda - Guess what, it's two more Dell headset mic quirks ALSA: hda - Yet another Dell headset mic quirk ALSA: hda - Add support for ALC5505 DSP power-save mode ASoC: mfld: Remove unused variable ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE ALSA: usb-audio: claim autodetected PCM interfaces all at once ALSA: usb-audio: remove superfluous Roland quirks ...
Diffstat (limited to 'sound/soc/davinci/davinci-sffsdr.c')
-rw-r--r--sound/soc/davinci/davinci-sffsdr.c181
1 files changed, 0 insertions, 181 deletions
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
deleted file mode 100644
index 5be65aae7e0..00000000000
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ /dev/null
@@ -1,181 +0,0 @@
-/*
- * ASoC driver for Lyrtech SFFSDR board.
- *
- * Author: Hugo Villeneuve
- * Copyright (C) 2008 Lyrtech inc
- *
- * Based on ASoC driver for TI DAVINCI EVM platform, original copyright follow:
- * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/gpio.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/dma.h>
-#include <asm/mach-types.h>
-#ifdef CONFIG_SFFSDR_FPGA
-#include <asm/plat-sffsdr/sffsdr-fpga.h>
-#endif
-
-#include <mach/edma.h>
-
-#include "../codecs/pcm3008.h"
-#include "davinci-pcm.h"
-#include "davinci-i2s.h"
-
-/*
- * CLKX and CLKR are the inputs for the Sample Rate Generator.
- * FSX and FSR are outputs, driven by the sample Rate Generator.
- */
-#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
- SND_SOC_DAIFMT_CBM_CFS | \
- SND_SOC_DAIFMT_IB_NF)
-
-static int sffsdr_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int fs;
- int ret = 0;
-
- /* Fsref can be 32000, 44100 or 48000. */
- fs = params_rate(params);
-
-#ifndef CONFIG_SFFSDR_FPGA
- /* Without the FPGA module, the Fs is fixed at 44100 Hz */
- if (fs != 44100) {
- pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n");
- return -EINVAL;
- }
-#endif
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
- if (ret < 0)
- return ret;
-
- pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
-
-#ifndef CONFIG_SFFSDR_FPGA
- return 0;
-#else
- return sffsdr_fpga_set_codec_fs(fs);
-#endif
-}
-
-static struct snd_soc_ops sffsdr_ops = {
- .hw_params = sffsdr_hw_params,
-};
-
-/* davinci-sffsdr digital audio interface glue - connects codec <--> CPU */
-static struct snd_soc_dai_link sffsdr_dai = {
- .name = "PCM3008", /* Codec name */
- .stream_name = "PCM3008 HiFi",
- .cpu_dai_name = "davinci-mcbsp",
- .codec_dai_name = "pcm3008-hifi",
- .codec_name = "pcm3008-codec",
- .platform_name = "davinci-mcbsp",
- .ops = &sffsdr_ops,
-};
-
-/* davinci-sffsdr audio machine driver */
-static struct snd_soc_card snd_soc_sffsdr = {
- .name = "DaVinci SFFSDR",
- .owner = THIS_MODULE,
- .dai_link = &sffsdr_dai,
- .num_links = 1,
-};
-
-/* sffsdr audio private data */
-static struct pcm3008_setup_data sffsdr_pcm3008_setup = {
- .dem0_pin = GPIO(45),
- .dem1_pin = GPIO(46),
- .pdad_pin = GPIO(47),
- .pdda_pin = GPIO(38),
-};
-
-struct platform_device pcm3008_codec = {
- .name = "pcm3008-codec",
- .id = 0,
- .dev = {
- .platform_data = &sffsdr_pcm3008_setup,
- },
-};
-
-static struct resource sffsdr_snd_resources[] = {
- {
- .start = DAVINCI_MCBSP_BASE,
- .end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
- .flags = IORESOURCE_MEM,
- },
-};
-
-static struct evm_snd_platform_data sffsdr_snd_data = {
- .tx_dma_ch = DAVINCI_DMA_MCBSP_TX,
- .rx_dma_ch = DAVINCI_DMA_MCBSP_RX,
-};
-
-static struct platform_device *sffsdr_snd_device;
-
-static int __init sffsdr_init(void)
-{
- int ret;
-
- if (!machine_is_sffsdr())
- return -EINVAL;
-
- platform_device_register(&pcm3008_codec);
-
- sffsdr_snd_device = platform_device_alloc("soc-audio", 0);
- if (!sffsdr_snd_device) {
- printk(KERN_ERR "platform device allocation failed\n");
- return -ENOMEM;
- }
-
- platform_set_drvdata(sffsdr_snd_device, &snd_soc_sffsdr);
- platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data,
- sizeof(sffsdr_snd_data));
-
- ret = platform_device_add_resources(sffsdr_snd_device,
- sffsdr_snd_resources,
- ARRAY_SIZE(sffsdr_snd_resources));
- if (ret) {
- printk(KERN_ERR "platform device add resources failed\n");
- goto error;
- }
-
- ret = platform_device_add(sffsdr_snd_device);
- if (ret)
- goto error;
-
- return ret;
-
-error:
- platform_device_put(sffsdr_snd_device);
- return ret;
-}
-
-static void __exit sffsdr_exit(void)
-{
- platform_device_unregister(sffsdr_snd_device);
- platform_device_unregister(&pcm3008_codec);
-}
-
-module_init(sffsdr_init);
-module_exit(sffsdr_exit);
-
-MODULE_AUTHOR("Hugo Villeneuve");
-MODULE_DESCRIPTION("Lyrtech SFFSDR ASoC driver");
-MODULE_LICENSE("GPL");