Merge branch 'topic/asoc' into for-linus

4 files changed, 581 insertions, 3 deletions

diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 5998ab366e83..ad8a10fe6298 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -116,6 +116,16 @@ config SND_SOC_ZYLONITE
           Say Y if you want to add support for SoC audio on the
           Marvell Zylonite reference platform.
 
+config SND_PXA2XX_SOC_MAGICIAN
+        tristate "SoC Audio support for HTC Magician"
+        depends on SND_PXA2XX_SOC && MACH_MAGICIAN
+        select SND_PXA2XX_SOC_I2S
+        select SND_PXA_SOC_SSP
+        select SND_SOC_UDA1380
+        help
+          Say Y if you want to add support for SoC audio on the
+          HTC Magician.
+
 config SND_PXA2XX_SOC_MIOA701
         tristate "SoC Audio support for MIO A701"
         depends on SND_PXA2XX_SOC && MACH_MIOA701
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 8ed881c5e5cc..4b90c3ccae45 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -20,6 +20,7 @@ snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
 snd-soc-zylonite-objs := zylonite.o
+snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
@@ -31,5 +32,6 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
 obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
new file mode 100644
index 000000000000..f7c4544f7859
--- /dev/null
+++ b/sound/soc/pxa/magician.c
@@ -0,0 +1,560 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/pxa-regs.h>
+#include <mach/hardware.h>
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC       0
+#define MAGICIAN_MIC_EXT   1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_codec *codec)
+{
+        if (magician_spk_switch)
+                snd_soc_dapm_enable_pin(codec, "Speaker");
+        else
+                snd_soc_dapm_disable_pin(codec, "Speaker");
+        if (magician_hp_switch)
+                snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+        else
+                snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+
+        switch (magician_in_sel) {
+        case MAGICIAN_MIC:
+                snd_soc_dapm_disable_pin(codec, "Headset Mic");
+                snd_soc_dapm_enable_pin(codec, "Call Mic");
+                break;
+        case MAGICIAN_MIC_EXT:
+                snd_soc_dapm_disable_pin(codec, "Call Mic");
+                snd_soc_dapm_enable_pin(codec, "Headset Mic");
+                break;
+        }
+
+        snd_soc_dapm_sync(codec);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+        struct snd_soc_pcm_runtime *rtd = substream->private_data;
+        struct snd_soc_codec *codec = rtd->socdev->card->codec;
+
+        /* check the jack status at stream startup */
+        magician_ext_control(codec);
+
+        return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_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 *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        unsigned int acps, acds, width, rate;
+        unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+        int ret = 0;
+
+        rate = params_rate(params);
+        width = snd_pcm_format_physical_width(params_format(params));
+
+        /*
+         * rate = SSPSCLK / (2 * width(16 or 32))
+         * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+         */
+        switch (params_rate(params)) {
+        case 8000:
+                /* off by a factor of 2: bug in the PXA27x audio clock? */
+                acps = 32842000;
+                switch (width) {
+                case 16:
+                        /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_16;
+                        break;
+                case 32:
+                        /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_8;
+                }
+                break;
+        case 11025:
+                acps = 5622000;
+                switch (width) {
+                case 16:
+                        /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_4;
+                        break;
+                case 32:
+                        /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                }
+                break;
+        case 22050:
+                acps = 5622000;
+                switch (width) {
+                case 16:
+                        /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                        break;
+                case 32:
+                        /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_1;
+                }
+                break;
+        case 44100:
+                acps = 5622000;
+                switch (width) {
+                case 16:
+                        /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                        break;
+                case 32:
+                        /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_1;
+                }
+                break;
+        case 48000:
+                acps = 12235000;
+                switch (width) {
+                case 16:
+                        /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                        break;
+                case 32:
+                        /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_1;
+                }
+                break;
+        case 96000:
+                acps = 12235000;
+                switch (width) {
+                case 16:
+                        /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_1;
+                        break;
+                case 32:
+                        /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+                        acds = PXA_SSP_CLK_AUDIO_DIV_2;
+                        div4 = PXA_SSP_CLK_SCDB_1;
+                        break;
+                }
+                break;
+        }
+
+        /* set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+                        SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+                        SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1);
+        if (ret < 0)
+                return ret;
+
+        /* set audio clock as clock source */
+        ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+                        SND_SOC_CLOCK_OUT);
+        if (ret < 0)
+                return ret;
+
+        /* set the SSP audio system clock ACDS divider */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai,
+                        PXA_SSP_AUDIO_DIV_ACDS, acds);
+        if (ret < 0)
+                return ret;
+
+        /* set the SSP audio system clock SCDB divider4 */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai,
+                        PXA_SSP_AUDIO_DIV_SCDB, div4);
+        if (ret < 0)
+                return ret;
+
+        /* set SSP audio pll clock */
+        ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_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 *codec_dai = rtd->dai->codec_dai;
+        struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+        int ret = 0;
+
+        /* set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai,
+                        SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+                        SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai,
+                        SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+                        SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* set the I2S system clock as output */
+        ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+                        SND_SOC_CLOCK_OUT);
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+        .startup = magician_startup,
+        .hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+        .startup = magician_startup,
+        .hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.integer.value[0] = magician_hp_switch;
+        return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+        if (magician_hp_switch == ucontrol->value.integer.value[0])
+                return 0;
+
+        magician_hp_switch = ucontrol->value.integer.value[0];
+        magician_ext_control(codec);
+        return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.integer.value[0] = magician_spk_switch;
+        return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+        if (magician_spk_switch == ucontrol->value.integer.value[0])
+                return 0;
+
+        magician_spk_switch = ucontrol->value.integer.value[0];
+        magician_ext_control(codec);
+        return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.integer.value[0] = magician_in_sel;
+        return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
+{
+        if (magician_in_sel == ucontrol->value.integer.value[0])
+                return 0;
+
+        magician_in_sel = ucontrol->value.integer.value[0];
+
+        switch (magician_in_sel) {
+        case MAGICIAN_MIC:
+                gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+                break;
+        case MAGICIAN_MIC_EXT:
+                gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+        }
+
+        return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+                                struct snd_kcontrol *k, int event)
+{
+        gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+        return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+                                struct snd_kcontrol *k, int event)
+{
+        gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+        return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+                                struct snd_kcontrol *k, int event)
+{
+        gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+        return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+        SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+        SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+        SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+        SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+        /* Headphone connected to VOUTL, VOUTR */
+        {"Headphone Jack", NULL, "VOUTL"},
+        {"Headphone Jack", NULL, "VOUTR"},
+
+        /* Speaker connected to VOUTL, VOUTR */
+        {"Speaker", NULL, "VOUTL"},
+        {"Speaker", NULL, "VOUTR"},
+
+        /* Mics are connected to VINM */
+        {"VINM", NULL, "Headset Mic"},
+        {"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+        SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+        SOC_SINGLE_BOOL_EXT("Headphone Switch",
+                        (unsigned long)&magician_hp_switch,
+                        magician_get_hp, magician_set_hp),
+        SOC_SINGLE_BOOL_EXT("Speaker Switch",
+                        (unsigned long)&magician_spk_switch,
+                        magician_get_spk, magician_set_spk),
+        SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+                        magician_get_input, magician_set_input),
+};
+
+/*
+ * Logic for a uda1380 as connected on a HTC Magician
+ */
+static int magician_uda1380_init(struct snd_soc_codec *codec)
+{
+        int err;
+
+        /* NC codec pins */
+        snd_soc_dapm_nc_pin(codec, "VOUTLHP");
+        snd_soc_dapm_nc_pin(codec, "VOUTRHP");
+
+        /* FIXME: is anything connected here? */
+        snd_soc_dapm_nc_pin(codec, "VINL");
+        snd_soc_dapm_nc_pin(codec, "VINR");
+
+        /* Add magician specific controls */
+        err = snd_soc_add_controls(codec, uda1380_magician_controls,
+                                ARRAY_SIZE(uda1380_magician_controls));
+        if (err < 0)
+                return err;
+
+        /* Add magician specific widgets */
+        snd_soc_dapm_new_controls(codec, uda1380_dapm_widgets,
+                                  ARRAY_SIZE(uda1380_dapm_widgets));
+
+        /* Set up magician specific audio path interconnects */
+        snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+        snd_soc_dapm_sync(codec);
+        return 0;
+}
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+        .name = "uda1380",
+        .stream_name = "UDA1380 Playback",
+        .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1],
+        .codec_dai = &uda1380_dai[UDA1380_DAI_PLAYBACK],
+        .init = magician_uda1380_init,
+        .ops = &magician_playback_ops,
+},
+{
+        .name = "uda1380",
+        .stream_name = "UDA1380 Capture",
+        .cpu_dai = &pxa_i2s_dai,
+        .codec_dai = &uda1380_dai[UDA1380_DAI_CAPTURE],
+        .ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+        .name = "Magician",
+        .dai_link = magician_dai,
+        .num_links = ARRAY_SIZE(magician_dai),
+        .platform = &pxa2xx_soc_platform,
+};
+
+/* magician audio private data */
+static struct uda1380_setup_data magician_uda1380_setup = {
+        .i2c_address = 0x18,
+        .dac_clk = UDA1380_DAC_CLK_WSPLL,
+};
+
+/* magician audio subsystem */
+static struct snd_soc_device magician_snd_devdata = {
+        .card = &snd_soc_card_magician,
+        .codec_dev = &soc_codec_dev_uda1380,
+        .codec_data = &magician_uda1380_setup,
+};
+
+static struct platform_device *magician_snd_device;
+
+static int __init magician_init(void)
+{
+        int ret;
+
+        if (!machine_is_magician())
+                return -ENODEV;
+
+        ret = gpio_request(EGPIO_MAGICIAN_CODEC_POWER, "CODEC_POWER");
+        if (ret)
+                goto err_request_power;
+        ret = gpio_request(EGPIO_MAGICIAN_CODEC_RESET, "CODEC_RESET");
+        if (ret)
+                goto err_request_reset;
+        ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+        if (ret)
+                goto err_request_spk;
+        ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+        if (ret)
+                goto err_request_ep;
+        ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+        if (ret)
+                goto err_request_mic;
+        ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+        if (ret)
+                goto err_request_in_sel0;
+        ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+        if (ret)
+                goto err_request_in_sel1;
+
+        gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 1);
+        gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+        /* we may need to have the clock running here - pH5 */
+        gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 1);
+        udelay(5);
+        gpio_set_value(EGPIO_MAGICIAN_CODEC_RESET, 0);
+
+        magician_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!magician_snd_device) {
+                ret = -ENOMEM;
+                goto err_pdev;
+        }
+
+        platform_set_drvdata(magician_snd_device, &magician_snd_devdata);
+        magician_snd_devdata.dev = &magician_snd_device->dev;
+        ret = platform_device_add(magician_snd_device);
+        if (ret) {
+                platform_device_put(magician_snd_device);
+                goto err_pdev;
+        }
+
+        return 0;
+
+err_pdev:
+        gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+        gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+        gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+        gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+        gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+        gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+err_request_reset:
+        gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+err_request_power:
+        return ret;
+}
+
+static void __exit magician_exit(void)
+{
+        platform_device_unregister(magician_snd_device);
+
+        gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+        gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+        gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+        gpio_set_value(EGPIO_MAGICIAN_CODEC_POWER, 0);
+
+        gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+        gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+        gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+        gpio_free(EGPIO_MAGICIAN_EP_POWER);
+        gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+        gpio_free(EGPIO_MAGICIAN_CODEC_RESET);
+        gpio_free(EGPIO_MAGICIAN_CODEC_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 7acd3febf8b0..308a657928d2 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -627,12 +627,18 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
         u32 sscr0;
         u32 sspsp;
         int width = snd_pcm_format_physical_width(params_format(params));
+        int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf;
 
         /* select correct DMA params */
         if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
                 dma = 1; /* capture DMA offset is 1,3 */
-        if (chn == 2)
-                dma += 2; /* stereo DMA offset is 2, mono is 0 */
+        /* Network mode with one active slot (ttsa == 1) can be used
+         * to force 16-bit frame width on the wire (for S16_LE), even
+         * with two channels. Use 16-bit DMA transfers for this case.
+         */
+        if (((chn == 2) && (ttsa != 1)) || (width == 32))
+                dma += 2; /* 32-bit DMA offset is 2, 16-bit is 0 */
+
         cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
 
         dev_dbg(&ssp->pdev->dev, "pxa_ssp_hw_params: dma %d\n", dma);
@@ -712,7 +718,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
         /* When we use a network mode, we always require TDM slots
          * - complain loudly and fail if they've not been set up yet.
          */
-        if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+        if ((sscr0 & SSCR0_MOD) && !ttsa) {
                 dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
                 return -EINVAL;
         }