From patchwork Wed Mar 5 08:34:34 2014 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: rjying X-Patchwork-Id: 3771361 Return-Path: X-Original-To: patchwork-alsa-devel@patchwork.kernel.org Delivered-To: patchwork-parsemail@patchwork2.web.kernel.org Received: from mail.kernel.org (mail.kernel.org [198.145.19.201]) by patchwork2.web.kernel.org (Postfix) with ESMTP id D0C98BF13A for ; Wed, 5 Mar 2014 08:36:12 +0000 (UTC) Received: from mail.kernel.org (localhost [127.0.0.1]) by mail.kernel.org (Postfix) with ESMTP id 212682024F for ; Wed, 5 Mar 2014 08:36:11 +0000 (UTC) Received: from alsa0.perex.cz (alsa0.perex.cz [77.48.224.243]) by mail.kernel.org (Postfix) with ESMTP id 1A73520237 for ; Wed, 5 Mar 2014 08:36:09 +0000 (UTC) Received: by alsa0.perex.cz (Postfix, from userid 1000) id 0DFFE265010; Wed, 5 Mar 2014 09:36:07 +0100 (CET) X-Spam-Checker-Version: SpamAssassin 3.3.1 (2010-03-16) on mail.kernel.org X-Spam-Level: X-Spam-Status: No, score=-1.8 required=5.0 tests=BAYES_00, DKIM_ADSP_CUSTOM_MED, DKIM_SIGNED, FREEMAIL_FROM, NO_DNS_FOR_FROM, T_DKIM_INVALID, UNPARSEABLE_RELAY autolearn=no version=3.3.1 Received: from alsa0.perex.cz (localhost [IPv6:::1]) by alsa0.perex.cz (Postfix) with ESMTP id 37B0026516A; Wed, 5 Mar 2014 09:35:27 +0100 (CET) X-Original-To: alsa-devel@alsa-project.org Delivered-To: alsa-devel@alsa-project.org Received: by alsa0.perex.cz (Postfix, from userid 1000) id F1883265192; Wed, 5 Mar 2014 09:35:25 +0100 (CET) Received: from mail-pa0-f45.google.com (mail-pa0-f45.google.com [209.85.220.45]) by alsa0.perex.cz (Postfix) with ESMTP id 67BA826500A for ; Wed, 5 Mar 2014 09:35:19 +0100 (CET) Received: by mail-pa0-f45.google.com with SMTP id kl14so798756pab.4 for ; Wed, 05 Mar 2014 00:35:18 -0800 (PST) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20120113; h=from:to:cc:subject:date:message-id:in-reply-to:references; bh=4j8t0TO7PI0j/jSAonb6H1+RbdX5XK+oGHGv9NlXVH0=; b=0/OeL88xPafADjZCHXcHsBGw+LwYxcigdSDiePoYAC5llGvB1sqaDWUsKZq9jhzyJn BF7S/upa60pxK2uNbo4WH8m5X24lPNCQ+tO5WxCYhUJnKVZDF0SIo5/1qsCecQpOXg6Y C5l9bhVcAn4RXcijiCDvQnxL8PR0t9LGgMbCyRX22YOim6zKaYOsZ7PvZWFv1sF0lt/r SJ7FY6V/dLqf6422mwkGdakGIptJRqp5/JSByLVG563iHfgaCxGzb4oRQ57KLPkH1e0P aSAYet62dV8RwVIu7eihmUoaaFyhn+LD5K4ZJ/giy/bAa5Pijq+MlX3fGhPuPkFRusEI VOOQ== X-Received: by 10.68.230.200 with SMTP id ta8mr4171859pbc.10.1394008518472; Wed, 05 Mar 2014 00:35:18 -0800 (PST) Received: from localhost.localdomain ([101.83.57.75]) by mx.google.com with ESMTPSA id db3sm5444984pbb.10.2014.03.05.00.35.10 for (version=TLSv1 cipher=ECDHE-RSA-RC4-SHA bits=128/128); Wed, 05 Mar 2014 00:35:17 -0800 (PST) From: RongJun Ying To: Liam Girdwood , Mark Brown , rjying@gmail.com Date: Wed, 5 Mar 2014 16:34:34 +0800 Message-Id: <1394008480-29135-2-git-send-email-rongjun.ying@csr.com> X-Mailer: git-send-email 1.7.5.4 In-Reply-To: <1394008480-29135-1-git-send-email-rongjun.ying@csr.com> References: <1394008480-29135-1-git-send-email-rongjun.ying@csr.com> Cc: Takashi Iwai , Rongjun Ying , alsa-devel@alsa-project.org, workgroup.linux@csr.com Subject: [alsa-devel] [PATCH v5-resend 1/7] ASoC: sirf: Add SiRF internal audio codec driver X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.14 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , MIME-Version: 1.0 Errors-To: alsa-devel-bounces@alsa-project.org Sender: alsa-devel-bounces@alsa-project.org X-Virus-Scanned: ClamAV using ClamSMTP From: Rongjun Ying SiRF internal audio codec is integrated in SiRF atlas6 and prima2 SoC. Features include: 1. Stereo DAC and ADC with 16-bit resolution amd 48KHz sample rate 2. Support headphone and/or speaker output 3. Integrate headphone and speaker output amp 4. Support LINE and MIC input 5. Support single ended and differential input mode Signed-off-by: Rongjun Ying --v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. --- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 533 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ 5 files changed, 631 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt new file mode 100644 index 0000000..062f5ec --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt @@ -0,0 +1,17 @@ +SiRF internal audio CODEC + +Required properties: + + - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec" + + - reg : the register address of the device. + + - clocks: the clock of SiRF internal audio codec + +Example: + +audiocodec: audiocodec@b0040000 { + compatible = "sirf,atlas6-audio-codec"; + reg = <0xb0040000 0x10000>; + clocks = <&clks 27>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f2383eb..e19b64f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2518 if I2C @@ -383,6 +384,10 @@ config SND_SOC_SIGMADSP tristate select CRC32 +config SND_SOC_SIRF_AUDIO_CODEC + tristate "SiRF SoC internal audio codec" + select REGMAP_MMIO + config SND_SOC_SN95031 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6af7a55..4c9b70a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -61,6 +61,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-si476x-objs := si476x.o +snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c new file mode 100644 index 0000000..90e3a22 --- /dev/null +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -0,0 +1,533 @@ +/* + * SiRF audio codec driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sirf-audio-codec.h" + +struct sirf_audio_codec { + struct clk *clk; + struct regmap *regmap; + u32 reg_ctrl0, reg_ctrl1; +}; + +static const char * const input_mode_mux[] = {"Single-ended", + "Differential"}; + +static const struct soc_enum input_mode_mux_enum = + SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux); + +static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control = + SOC_DAPM_ENUM("Route", input_mode_mux_enum); + +static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0); +static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0); +static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6, + 0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0), + 0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0), +); + +static struct snd_kcontrol_new volume_controls_atlas6[] = { + SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10, + 0x3F, 0, capture_vol_tlv_atlas6), +}; + +static struct snd_kcontrol_new volume_controls_prima2[] = { + SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10, + 0x1F, 0, capture_vol_tlv_prima2), +}; + +static struct snd_kcontrol_new left_input_path_controls[] = { + SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0), +}; + +static struct snd_kcontrol_new right_input_path_controls[] = { + SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0), +}; + +static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0); + +static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0); + +static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0); + +static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0); + +/* After enable adc, Delay 200ms to avoid pop noise */ +static int adc_enable_delay_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(200); + break; + default: + break; + } + + return 0; +} + +static void enable_and_reset_codec(struct regmap *regmap, + u32 codec_enable_bits, u32 codec_reset_bits) +{ + regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, + codec_enable_bits | codec_reset_bits, + codec_enable_bits | ~codec_reset_bits); + msleep(20); + regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, + codec_reset_bits, codec_reset_bits); +} + +static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ +#define ATLAS6_CODEC_ENABLE_BITS (1 << 29) +#define ATLAS6_CODEC_RESET_BITS (1 << 28) + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + enable_and_reset_codec(sirf_audio_codec->regmap, + ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, + ~ATLAS6_CODEC_ENABLE_BITS); + break; + default: + break; + } + + return 0; +} + +static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ +#define PRIMA2_CODEC_ENABLE_BITS (1 << 27) +#define PRIMA2_CODEC_RESET_BITS (1 << 26) + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + enable_and_reset_codec(sirf_audio_codec->regmap, + PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, + ~PRIMA2_CODEC_ENABLE_BITS); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1, + 25, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1, + 26, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1, + 27, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1, + 23, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1, + 24, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1, + 25, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget = + SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0, + atlas6_codec_enable_and_reset_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD); + +static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget = + SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0, + prima2_codec_enable_and_reset_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD); + +static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0), + SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0), + SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0, + NULL, 0), + + SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &left_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &right_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + + SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0, + adc_enable_delay_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0, + adc_enable_delay_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0, + &left_input_path_controls[0], + ARRAY_SIZE(left_input_path_controls)), + SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0, + &right_input_path_controls[0], + ARRAY_SIZE(right_input_path_controls)), + + SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0, + &sirf_audio_codec_input_mode_control), + SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0), + SND_SOC_DAPM_INPUT("MICIN1"), + SND_SOC_DAPM_INPUT("MICIN2"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + + SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0, + 30, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route sirf_audio_codec_map[] = { + {"SPKOUT", NULL, "Speaker Driver"}, + {"Speaker Driver", NULL, "Speaker amp driver"}, + {"Speaker amp driver", NULL, "Left dac to speaker lineout"}, + {"Speaker amp driver", NULL, "Right dac to speaker lineout"}, + {"Left dac to speaker lineout", "Switch", "DAC left"}, + {"Right dac to speaker lineout", "Switch", "DAC right"}, + {"HPOUTL", NULL, "HP Left Driver"}, + {"HPOUTR", NULL, "HP Right Driver"}, + {"HP Left Driver", NULL, "HP amp left driver"}, + {"HP Right Driver", NULL, "HP amp right driver"}, + {"HP amp left driver", NULL, "Right dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"HP amp left driver", NULL, "Left dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"Right dac to hp left amp", "Switch", "DAC left"}, + {"Right dac to hp right amp", "Switch", "DAC right"}, + {"Left dac to hp left amp", "Switch", "DAC left"}, + {"Left dac to hp right amp", "Switch", "DAC right"}, + {"DAC left", NULL, "codecclk"}, + {"DAC right", NULL, "codecclk"}, + {"DAC left", NULL, "Playback"}, + {"DAC right", NULL, "Playback"}, + {"DAC left", NULL, "HSL Phase Opposite"}, + {"DAC right", NULL, "HSL Phase Opposite"}, + + {"Capture", NULL, "ADC left"}, + {"Capture", NULL, "ADC right"}, + {"ADC left", NULL, "codecclk"}, + {"ADC right", NULL, "codecclk"}, + {"ADC left", NULL, "Left PGA mixer"}, + {"ADC right", NULL, "Right PGA mixer"}, + {"Left PGA mixer", "Line Left Switch", "LINEIN2"}, + {"Right PGA mixer", "Line Right Switch", "LINEIN1"}, + {"Left PGA mixer", "Mic Left Switch", "MICIN2"}, + {"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"}, + {"Mic input mode mux", "Single-ended", "MICIN1"}, + {"Mic input mode mux", "Differential", "MICIN1"}, +}; + +static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, + int cmd, + struct snd_soc_dai *dai) +{ + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + u32 val = 0; + + /* + * This is a workaround, When stop playback, + * need disable HP amp, avoid the current noise. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (playback) + val = IC_HSLEN | IC_HSREN; + break; + default: + return -EINVAL; + } + + if (playback) + snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, + IC_HSLEN | IC_HSREN, val); + return 0; +} + +struct snd_soc_dai_ops sirf_audio_codec_dai_ops = { + .trigger = sirf_audio_codec_trigger, +}; + +struct snd_soc_dai_driver sirf_audio_codec_dai = { + .name = "sirf-audio-codec", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &sirf_audio_codec_dai_ops, +}; + +static int sirf_audio_codec_probe(struct snd_soc_codec *codec) +{ + int ret; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); + + pm_runtime_enable(codec->dev); + codec->control_data = sirf_audio_codec->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { + snd_soc_dapm_new_controls(dapm, + prima2_output_driver_dapm_widgets, + ARRAY_SIZE(prima2_output_driver_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, + &prima2_codec_clock_dapm_widget, 1); + return snd_soc_add_codec_controls(codec, + volume_controls_prima2, + ARRAY_SIZE(volume_controls_prima2)); + } + if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) { + snd_soc_dapm_new_controls(dapm, + atlas6_output_driver_dapm_widgets, + ARRAY_SIZE(atlas6_output_driver_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, + &atlas6_codec_clock_dapm_widget, 1); + return snd_soc_add_codec_controls(codec, + volume_controls_atlas6, + ARRAY_SIZE(volume_controls_atlas6)); + } + + return -EINVAL; +} + +static int sirf_audio_codec_remove(struct snd_soc_codec *codec) +{ + pm_runtime_disable(codec->dev); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = { + .probe = sirf_audio_codec_probe, + .remove = sirf_audio_codec_remove, + .dapm_widgets = sirf_audio_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets), + .dapm_routes = sirf_audio_codec_map, + .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map), + .idle_bias_off = true, +}; + +static const struct of_device_id sirf_audio_codec_of_match[] = { + { .compatible = "sirf,prima2-audio-codec" }, + { .compatible = "sirf,atlas6-audio-codec" }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match); + +static const struct regmap_config sirf_audio_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AUDIO_IC_CODEC_CTRL3, + .cache_type = REGCACHE_NONE, +}; + +static int sirf_audio_codec_driver_probe(struct platform_device *pdev) +{ + int ret; + struct sirf_audio_codec *sirf_audio_codec; + void __iomem *base; + struct resource *mem_res; + const struct of_device_id *match; + + match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); + + sirf_audio_codec = devm_kzalloc(&pdev->dev, + sizeof(struct sirf_audio_codec), GFP_KERNEL); + if (!sirf_audio_codec) + return -ENOMEM; + + platform_set_drvdata(pdev, sirf_audio_codec); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (base == NULL) + return -ENOMEM; + + sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sirf_audio_codec_regmap_config); + if (IS_ERR(sirf_audio_codec->regmap)) + return PTR_ERR(sirf_audio_codec->regmap); + + sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(sirf_audio_codec->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + return PTR_ERR(sirf_audio_codec->clk); + } + + ret = clk_prepare_enable(sirf_audio_codec->clk); + if (ret) { + dev_err(&pdev->dev, "Enable clock failed.\n"); + return ret; + } + + ret = snd_soc_register_codec(&(pdev->dev), + &soc_codec_device_sirf_audio_codec, + &sirf_audio_codec_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio Codec dai failed.\n"); + goto err_clk_put; + } + + /* + * Always open charge pump, if not, when the charge pump closed the + * adc will not stable + */ + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + IC_CPFREQ, IC_CPFREQ); + + if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec")) + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN); + return 0; + +err_clk_put: + clk_disable_unprepare(sirf_audio_codec->clk); + return ret; +} + +static int sirf_audio_codec_driver_remove(struct platform_device *pdev) +{ + struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev); + + clk_disable_unprepare(sirf_audio_codec->clk); + snd_soc_unregister_codec(&(pdev->dev)); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int sirf_audio_codec_suspend(struct device *dev) +{ + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev); + + regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + &sirf_audio_codec->reg_ctrl0); + regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1, + &sirf_audio_codec->reg_ctrl1); + clk_disable_unprepare(sirf_audio_codec->clk); + + return 0; +} + +static int sirf_audio_codec_resume(struct device *dev) +{ + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(sirf_audio_codec->clk); + if (ret) + return ret; + + regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + sirf_audio_codec->reg_ctrl0); + regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1, + sirf_audio_codec->reg_ctrl1); + + return 0; +} +#endif + +static const struct dev_pm_ops sirf_audio_codec_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume) +}; + +static struct platform_driver sirf_audio_codec_driver = { + .driver = { + .name = "sirf-audio-codec", + .owner = THIS_MODULE, + .of_match_table = sirf_audio_codec_of_match, + .pm = &sirf_audio_codec_pm_ops, + }, + .probe = sirf_audio_codec_driver_probe, + .remove = sirf_audio_codec_driver_remove, +}; + +module_platform_driver(sirf_audio_codec_driver); + +MODULE_DESCRIPTION("SiRF audio codec driver"); +MODULE_AUTHOR("RongJun Ying "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h new file mode 100644 index 0000000..d4c187b --- /dev/null +++ b/sound/soc/codecs/sirf-audio-codec.h @@ -0,0 +1,75 @@ +/* + * SiRF inner codec controllers define + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef _SIRF_AUDIO_CODEC_H +#define _SIRF_AUDIO_CODEC_H + + +#define AUDIO_IC_CODEC_PWR (0x00E0) +#define AUDIO_IC_CODEC_CTRL0 (0x00E4) +#define AUDIO_IC_CODEC_CTRL1 (0x00E8) +#define AUDIO_IC_CODEC_CTRL2 (0x00EC) +#define AUDIO_IC_CODEC_CTRL3 (0x00F0) + +#define MICBIASEN (1 << 3) + +#define IC_RDACEN (1 << 0) +#define IC_LDACEN (1 << 1) +#define IC_HSREN (1 << 2) +#define IC_HSLEN (1 << 3) +#define IC_SPEN (1 << 4) +#define IC_CPEN (1 << 5) + +#define IC_HPRSELR (1 << 6) +#define IC_HPLSELR (1 << 7) +#define IC_HPRSELL (1 << 8) +#define IC_HPLSELL (1 << 9) +#define IC_SPSELR (1 << 10) +#define IC_SPSELL (1 << 11) + +#define IC_MONOR (1 << 12) +#define IC_MONOL (1 << 13) + +#define IC_RXOSRSEL (1 << 28) +#define IC_CPFREQ (1 << 29) +#define IC_HSINVEN (1 << 30) + +#define IC_MICINREN (1 << 0) +#define IC_MICINLEN (1 << 1) +#define IC_MICIN1SEL (1 << 2) +#define IC_MICIN2SEL (1 << 3) +#define IC_MICDIFSEL (1 << 4) +#define IC_LINEIN1SEL (1 << 5) +#define IC_LINEIN2SEL (1 << 6) +#define IC_RADCEN (1 << 7) +#define IC_LADCEN (1 << 8) +#define IC_ALM (1 << 9) + +#define IC_DIGMICEN (1 << 22) +#define IC_DIGMICFREQ (1 << 23) +#define IC_ADC14B_12 (1 << 24) +#define IC_FIRDAC_HSL_EN (1 << 25) +#define IC_FIRDAC_HSR_EN (1 << 26) +#define IC_FIRDAC_LOUT_EN (1 << 27) +#define IC_POR (1 << 28) +#define IC_CODEC_CLK_EN (1 << 29) +#define IC_HP_3DB_BOOST (1 << 30) + +#define IC_ADC_LEFT_GAIN_SHIFT 16 +#define IC_ADC_RIGHT_GAIN_SHIFT 10 +#define IC_ADC_GAIN_MASK 0x3F +#define IC_MIC_MAX_GAIN 0x39 + +#define IC_RXPGAR_MASK 0x3F +#define IC_RXPGAR_SHIFT 14 +#define IC_RXPGAL_MASK 0x3F +#define IC_RXPGAL_SHIFT 21 +#define IC_RXPGAR 0x7B +#define IC_RXPGAL 0x7B + +#endif /*__SIRF_AUDIO_CODEC_H*/