initial commit

based on nebkat's device tree

5 files changed, 3565 insertions, 0 deletions

diff --git a/audio/Android.mk b/audio/Android.mk
new file mode 100644
index 0000000..4655db0
--- /dev/null
+++ b/audio/Android.mk
@@ -0,0 +1,33 @@
+# Copyright (C) 2011 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#      http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := audio.primary.$(TARGET_BOOTLOADER_BOARD_NAME)
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := audio_hw.c ril_interface.c
+
+LOCAL_C_INCLUDES += \
+	external/tinyalsa/include \
+	external/expat/lib \
+	$(call include-path-for, audio-utils) \
+	$(call include-path-for, audio-effects)
+
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioutils libdl libexpat
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/audio/audio_hw.c b/audio/audio_hw.c
new file mode 100755
index 0000000..42c6122
--- /dev/null
+++ b/audio/audio_hw.c
@@ -0,0 +1,3037 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ * Copyright (C) 2012 Wolfson Microelectronics plc
+ * Copyright (C) 2012 The CyanogenMod Project
+ *               Daniel Hillenbrand <codeworkx@cyanogenmod.com>
+ *               Guillaume "XpLoDWilD" Lesniak <xplodgui@gmail.com>
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_primary"
+#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/time.h>
+#include <stdlib.h>
+#include <expat.h>
+
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <hardware/audio.h>
+
+#include <tinyalsa/asoundlib.h>
+#include <audio_utils/resampler.h>
+#include <audio_utils/echo_reference.h>
+#include <hardware/audio_effect.h>
+#include <audio_effects/effect_aec.h>
+
+#include "audio_hw.h"
+#include "ril_interface.h"
+
+struct pcm_config pcm_config_mm = {
+    .channels = 2,
+    .rate = MM_FULL_POWER_SAMPLING_RATE,
+    .period_size = DEEP_BUFFER_LONG_PERIOD_SIZE,
+    .period_count = PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+};
+
+struct pcm_config pcm_config_tones = {
+    .channels = 2,
+    .rate = MM_FULL_POWER_SAMPLING_RATE,
+    .period_size = SHORT_PERIOD_SIZE,
+    .period_count = PLAYBACK_SHORT_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 0,
+    .avail_min = 0,
+};
+
+struct pcm_config pcm_config_capture = {
+    .channels = 2,
+    .rate = DEFAULT_IN_SAMPLING_RATE,
+    .period_size = CAPTURE_PERIOD_SIZE,
+    .period_count = CAPTURE_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+};
+
+struct pcm_config pcm_config_vx = {
+    .channels = 2,
+    .rate = VX_NB_SAMPLING_RATE,
+    .period_size = 160,
+    .period_count = 2,
+    .format = PCM_FORMAT_S16_LE,
+};
+
+#define MIN(x, y) ((x) > (y) ? (y) : (x))
+
+struct m0_audio_device {
+    struct audio_hw_device hw_device;
+
+    pthread_mutex_t lock;       /* see note below on mutex acquisition order */
+    struct m0_dev_cfg *dev_cfgs;
+    int num_dev_cfgs;
+    struct mixer *mixer;
+    struct mixer_ctls mixer_ctls;
+    audio_mode_t mode;
+    int active_devices;
+    int devices;
+    struct pcm *pcm_modem_dl;
+    struct pcm *pcm_modem_ul;
+    int in_call;
+    float voice_volume;
+    struct m0_stream_in *active_input;
+    struct m0_stream_out *outputs[OUTPUT_TOTAL];
+    bool mic_mute;
+    int tty_mode;
+    struct echo_reference_itfe *echo_reference;
+    bool bluetooth_nrec;
+    int wb_amr;
+    bool screen_off;
+
+    /* RIL */
+    struct ril_handle ril;
+};
+
+struct m0_stream_out {
+    struct audio_stream_out stream;
+
+    pthread_mutex_t lock;       /* see note below on mutex acquisition order */
+    struct pcm_config config[PCM_TOTAL];
+    struct pcm *pcm[PCM_TOTAL];
+    struct resampler_itfe *resampler;
+    char *buffer;
+    size_t buffer_frames;
+    int standby;
+    struct echo_reference_itfe *echo_reference;
+    int write_threshold;
+    bool use_long_periods;
+    audio_channel_mask_t channel_mask;
+    audio_channel_mask_t sup_channel_masks[3];
+
+    struct m0_audio_device *dev;
+};
+
+#define MAX_PREPROCESSORS 3 /* maximum one AGC + one NS + one AEC per input stream */
+
+struct effect_info_s {
+    effect_handle_t effect_itfe;
+    size_t num_channel_configs;
+    channel_config_t* channel_configs;
+};
+
+#define NUM_IN_AUX_CNL_CONFIGS 2
+channel_config_t in_aux_cnl_configs[NUM_IN_AUX_CNL_CONFIGS] = {
+    { AUDIO_CHANNEL_IN_FRONT , AUDIO_CHANNEL_IN_BACK},
+    { AUDIO_CHANNEL_IN_STEREO , AUDIO_CHANNEL_IN_RIGHT}
+};
+
+struct m0_stream_in {
+    struct audio_stream_in stream;
+
+    pthread_mutex_t lock;       /* see note below on mutex acquisition order */
+    struct pcm_config config;
+    struct pcm *pcm;
+    int device;
+    struct resampler_itfe *resampler;
+    struct resampler_buffer_provider buf_provider;
+    unsigned int requested_rate;
+    int standby;
+    int source;
+    struct echo_reference_itfe *echo_reference;
+    bool need_echo_reference;
+
+    int16_t *read_buf;
+    size_t read_buf_size;
+    size_t read_buf_frames;
+
+    int16_t *proc_buf_in;
+    int16_t *proc_buf_out;
+    size_t proc_buf_size;
+    size_t proc_buf_frames;
+
+    int16_t *ref_buf;
+    size_t ref_buf_size;
+    size_t ref_buf_frames;
+
+    int read_status;
+
+    int num_preprocessors;
+    struct effect_info_s preprocessors[MAX_PREPROCESSORS];
+
+    bool aux_channels_changed;
+    uint32_t main_channels;
+    uint32_t aux_channels;
+    struct m0_audio_device *dev;
+};
+
+struct m0_dev_cfg {
+    int mask;
+
+    struct route_setting *on;
+    unsigned int on_len;
+
+    struct route_setting *off;
+    unsigned int off_len;
+};
+
+/**
+ * NOTE: when multiple mutexes have to be acquired, always respect the following order:
+ *        hw device > in stream > out stream
+ */
+
+static void select_output_device(struct m0_audio_device *adev);
+static void select_input_device(struct m0_audio_device *adev);
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume);
+static int do_input_standby(struct m0_stream_in *in);
+static int do_output_standby(struct m0_stream_out *out);
+static void in_update_aux_channels(struct m0_stream_in *in, effect_handle_t effect);
+
+/* The enable flag when 0 makes the assumption that enums are disabled by
+ * "Off" and integers/booleans by 0 */
+static int set_bigroute_by_array(struct mixer *mixer, struct route_setting *route,
+                              int enable)
+{
+    struct mixer_ctl *ctl;
+    unsigned int i, j, ret;
+
+    /* Go through the route array and set each value */
+    i = 0;
+    while (route[i].ctl_name) {
+        ctl = mixer_get_ctl_by_name(mixer, route[i].ctl_name);
+        if (!ctl) {
+        ALOGE("Unknown control '%s'\n", route[i].ctl_name);
+            return -EINVAL;
+        }
+
+        if (route[i].strval) {
+            if (enable) {
+                ret = mixer_ctl_set_enum_by_string(ctl, route[i].strval);
+                if (ret != 0) {
+                    ALOGE("Failed to set '%s' to '%s'\n", route[i].ctl_name, route[i].strval);
+                } else {
+                    ALOGV("Set '%s' to '%s'\n", route[i].ctl_name, route[i].strval);
+                }
+            } else {
+                ret = mixer_ctl_set_enum_by_string(ctl, "Off");
+                if (ret != 0) {
+                    ALOGE("Failed to set '%s' to '%s'\n", route[i].ctl_name, route[i].strval);
+                } else {
+                    ALOGV("Set '%s' to '%s'\n", route[i].ctl_name, "Off");
+                }
+            }
+        } else {
+            /* This ensures multiple (i.e. stereo) values are set jointly */
+            for (j = 0; j < mixer_ctl_get_num_values(ctl); j++) {
+                if (enable) {
+                    ret = mixer_ctl_set_value(ctl, j, route[i].intval);
+                    if (ret != 0) {
+                        ALOGE("Failed to set '%s' to '%d'\n", route[i].ctl_name, route[i].intval);
+                    } else {
+                        ALOGV("Set '%s' to '%d'\n", route[i].ctl_name, route[i].intval);
+                    }
+                } else {
+                    ret = mixer_ctl_set_value(ctl, j, 0);
+                    if (ret != 0) {
+                        ALOGE("Failed to set '%s' to '%d'\n", route[i].ctl_name, route[i].intval);
+                    } else {
+                        ALOGV("Set '%s' to '%d'\n", route[i].ctl_name, 0);
+                    }
+                }
+            }
+        }
+        i++;
+    }
+
+    return 0;
+}
+
+/* The enable flag when 0 makes the assumption that enums are disabled by
+ * "Off" and integers/booleans by 0 */
+static int set_route_by_array(struct mixer *mixer, struct route_setting *route,
+                  unsigned int len)
+{
+    struct mixer_ctl *ctl;
+    unsigned int i, j, ret;
+
+    /* Go through the route array and set each value */
+    for (i = 0; i < len; i++) {
+        ctl = mixer_get_ctl_by_name(mixer, route[i].ctl_name);
+        if (!ctl) {
+        ALOGE("Unknown control '%s'\n", route[i].ctl_name);
+            return -EINVAL;
+        }
+
+        if (route[i].strval) {
+        ret = mixer_ctl_set_enum_by_string(ctl, route[i].strval);
+        if (ret != 0) {
+        ALOGE("Failed to set '%s' to '%s'\n",
+             route[i].ctl_name, route[i].strval);
+        } else {
+        ALOGV("Set '%s' to '%s'\n",
+             route[i].ctl_name, route[i].strval);
+        }
+
+        } else {
+            /* This ensures multiple (i.e. stereo) values are set jointly */
+            for (j = 0; j < mixer_ctl_get_num_values(ctl); j++) {
+        ret = mixer_ctl_set_value(ctl, j, route[i].intval);
+        if (ret != 0) {
+            ALOGE("Failed to set '%s'.%d to %d\n",
+             route[i].ctl_name, j, route[i].intval);
+        } else {
+            ALOGV("Set '%s'.%d to %d\n",
+             route[i].ctl_name, j, route[i].intval);
+        }
+        }
+        }
+    }
+
+    return 0;
+}
+
+/* Must be called with lock */
+void select_devices(struct m0_audio_device *adev)
+{
+    int i;
+
+    if (adev->active_devices == adev->devices)
+    return;
+
+    ALOGV("Changing devices %x => %x\n", adev->active_devices, adev->devices);
+
+    /* Turn on new devices first so we don't glitch due to powerdown... */
+    for (i = 0; i < adev->num_dev_cfgs; i++)
+    if ((adev->devices & adev->dev_cfgs[i].mask) &&
+        !(adev->active_devices & adev->dev_cfgs[i].mask))
+        set_route_by_array(adev->mixer, adev->dev_cfgs[i].on,
+                   adev->dev_cfgs[i].on_len);
+
+    /* ...then disable old ones. */
+    for (i = 0; i < adev->num_dev_cfgs; i++)
+    if (!(adev->devices & adev->dev_cfgs[i].mask) &&
+        (adev->active_devices & adev->dev_cfgs[i].mask))
+        set_route_by_array(adev->mixer, adev->dev_cfgs[i].off,
+                   adev->dev_cfgs[i].off_len);
+
+    adev->active_devices = adev->devices;
+}
+
+static int start_call(struct m0_audio_device *adev)
+{
+    ALOGE("Opening modem PCMs");
+    int bt_on;
+
+    bt_on = adev->devices & AUDIO_DEVICE_OUT_ALL_SCO;
+    pcm_config_vx.rate = adev->wb_amr ? VX_WB_SAMPLING_RATE : VX_NB_SAMPLING_RATE;
+
+    /* Open modem PCM channels */
+    if (adev->pcm_modem_dl == NULL) {
+        if (bt_on)
+            adev->pcm_modem_dl = pcm_open(CARD_DEFAULT, PORT_BT, PCM_OUT, &pcm_config_vx);
+        else
+            adev->pcm_modem_dl = pcm_open(CARD_DEFAULT, PORT_MODEM, PCM_OUT, &pcm_config_vx);
+        if (!pcm_is_ready(adev->pcm_modem_dl)) {
+            ALOGE("cannot open PCM modem DL stream: %s", pcm_get_error(adev->pcm_modem_dl));
+            goto err_open_dl;
+        }
+    }
+
+    if (adev->pcm_modem_ul == NULL) {
+        adev->pcm_modem_ul = pcm_open(CARD_DEFAULT, PORT_MODEM, PCM_IN, &pcm_config_vx);
+        if (!pcm_is_ready(adev->pcm_modem_ul)) {
+            ALOGE("cannot open PCM modem UL stream: %s", pcm_get_error(adev->pcm_modem_ul));
+            goto err_open_ul;
+        }
+    }
+
+    pcm_start(adev->pcm_modem_dl);
+    pcm_start(adev->pcm_modem_ul);
+
+    return 0;
+
+err_open_ul:
+    pcm_close(adev->pcm_modem_ul);
+    adev->pcm_modem_ul = NULL;
+err_open_dl:
+    pcm_close(adev->pcm_modem_dl);
+    adev->pcm_modem_dl = NULL;
+
+    return -ENOMEM;
+}
+
+static void end_call(struct m0_audio_device *adev)
+{
+    ALOGE("Closing modem PCMs");
+    pcm_stop(adev->pcm_modem_dl);
+    pcm_stop(adev->pcm_modem_ul);
+    pcm_close(adev->pcm_modem_dl);
+    pcm_close(adev->pcm_modem_ul);
+    adev->pcm_modem_dl = NULL;
+    adev->pcm_modem_ul = NULL;
+
+    /* re-enable +30db boost on mics */
+    mixer_ctl_set_value(adev->mixer_ctls.mixinl_in1l_volume, 0, 1);
+    mixer_ctl_set_value(adev->mixer_ctls.mixinl_in2l_volume, 0, 1);
+}
+
+static void set_eq_filter(struct m0_audio_device *adev)
+{
+}
+
+void audio_set_wb_amr_callback(void *data, int enable)
+{
+    struct m0_audio_device *adev = (struct m0_audio_device *)data;
+
+    pthread_mutex_lock(&adev->lock);
+    if (adev->wb_amr != enable) {
+        adev->wb_amr = enable;
+
+        /* reopen the modem PCMs at the new rate */
+        if (adev->in_call) {
+            end_call(adev);
+            select_output_device(adev);
+            start_call(adev);
+        }
+    }
+    pthread_mutex_unlock(&adev->lock);
+}
+
+static void set_incall_device(struct m0_audio_device *adev)
+{
+    int device_type;
+
+    switch(adev->devices & AUDIO_DEVICE_OUT_ALL) {
+        case AUDIO_DEVICE_OUT_EARPIECE:
+            device_type = SOUND_AUDIO_PATH_HANDSET;
+            break;
+        case AUDIO_DEVICE_OUT_SPEAKER:
+        case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+        case AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET:
+            device_type = SOUND_AUDIO_PATH_SPEAKER;
+            break;
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+            device_type = SOUND_AUDIO_PATH_HEADSET;
+            break;
+        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+            device_type = SOUND_AUDIO_PATH_HEADPHONE;
+            break;
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+        case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+            if (adev->bluetooth_nrec) {
+                device_type = SOUND_AUDIO_PATH_BLUETOOTH;
+            } else {
+                device_type = SOUND_AUDIO_PATH_BLUETOOTH_NO_NR;
+            }
+            break;
+        default:
+            device_type = SOUND_AUDIO_PATH_HANDSET;
+            break;
+    }
+
+    /* if output device isn't supported, open modem side to handset by default */
+    ALOGE("%s: ril_set_call_audio_path(%d)", __func__, device_type);
+    ril_set_call_audio_path(&adev->ril, device_type);
+}
+
+static void set_input_volumes(struct m0_audio_device *adev, int main_mic_on,
+                              int headset_mic_on, int sub_mic_on)
+{
+}
+
+static void set_output_volumes(struct m0_audio_device *adev, bool tty_volume)
+{
+}
+
+static void force_all_standby(struct m0_audio_device *adev)
+{
+    struct m0_stream_in *in;
+    struct m0_stream_out *out;
+
+    /* only needed for low latency output streams as other streams are not used
+     * for voice use cases */
+    if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL &&
+            !adev->outputs[OUTPUT_LOW_LATENCY]->standby) {
+        out = adev->outputs[OUTPUT_LOW_LATENCY];
+        pthread_mutex_lock(&out->lock);
+        do_output_standby(out);
+        pthread_mutex_unlock(&out->lock);
+    }
+
+    if (adev->active_input) {
+        in = adev->active_input;
+        pthread_mutex_lock(&in->lock);
+        do_input_standby(in);
+        pthread_mutex_unlock(&in->lock);
+    }
+}
+
+static void select_mode(struct m0_audio_device *adev)
+{
+    if (adev->mode == AUDIO_MODE_IN_CALL) {
+        ALOGE("Entering IN_CALL state, in_call=%d", adev->in_call);
+        if (!adev->in_call) {
+            force_all_standby(adev);
+            /* force earpiece route for in call state if speaker is the
+            only currently selected route. This prevents having to tear
+            down the modem PCMs to change route from speaker to earpiece
+            after the ringtone is played, but doesn't cause a route
+            change if a headset or bt device is already connected. If
+            speaker is not the only thing active, just remove it from
+            the route. We'll assume it'll never be used initally during
+            a call. This works because we're sure that the audio policy
+            manager will update the output device after the audio mode
+            change, even if the device selection did not change. */
+            if ((adev->devices & AUDIO_DEVICE_OUT_ALL) == AUDIO_DEVICE_OUT_SPEAKER)
+                adev->devices = AUDIO_DEVICE_OUT_EARPIECE |
+                                AUDIO_DEVICE_IN_BUILTIN_MIC;
+            else
+                adev->devices &= ~AUDIO_DEVICE_OUT_SPEAKER;
+            select_output_device(adev);
+            start_call(adev);
+            ril_set_call_clock_sync(&adev->ril, SOUND_CLOCK_START);
+            adev_set_voice_volume(&adev->hw_device, adev->voice_volume);
+            adev->in_call = 1;
+        }
+    } else {
+        ALOGE("Leaving IN_CALL state, in_call=%d, mode=%d",
+             adev->in_call, adev->mode);
+        if (adev->in_call) {
+            adev->in_call = 0;
+            end_call(adev);
+            force_all_standby(adev);
+            select_output_device(adev);
+            select_input_device(adev);
+        }
+    }
+}
+
+static void select_output_device(struct m0_audio_device *adev)
+{
+    int headset_on;
+    int headphone_on;
+    int speaker_on;
+    int earpiece_on;
+    int bt_on;
+    bool tty_volume = false;
+    unsigned int channel;
+
+    headset_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
+    headphone_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+    speaker_on = adev->devices & AUDIO_DEVICE_OUT_SPEAKER;
+    earpiece_on = adev->devices & AUDIO_DEVICE_OUT_EARPIECE;
+    bt_on = adev->devices & AUDIO_DEVICE_OUT_ALL_SCO;
+
+    switch(adev->devices & AUDIO_DEVICE_OUT_ALL) {
+        case AUDIO_DEVICE_OUT_SPEAKER:
+            ALOGD("%s: AUDIO_DEVICE_OUT_SPEAKER", __func__);
+            break;
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+            ALOGD("%s: AUDIO_DEVICE_OUT_WIRED_HEADSET", __func__);
+            break;
+        case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+            ALOGD("%s: AUDIO_DEVICE_OUT_WIRED_HEADPHONE", __func__);
+            break;
+        case AUDIO_DEVICE_OUT_EARPIECE:
+            ALOGD("%s: AUDIO_DEVICE_OUT_EARPIECE", __func__);
+            break;
+        case AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET:
+            ALOGD("%s: AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET", __func__);
+            break;
+        case AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET:
+            ALOGD("%s: AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET", __func__);
+            break;
+        case AUDIO_DEVICE_OUT_ALL_SCO:
+            ALOGD("%s: AUDIO_DEVICE_OUT_ALL_SCO", __func__);
+            break;
+        default:
+            ALOGD("%s: AUDIO_DEVICE_OUT_ALL", __func__);
+            break;
+    }
+
+    select_devices(adev);
+
+    set_eq_filter(adev);
+
+    if (adev->mode == AUDIO_MODE_IN_CALL) {
+        if (!bt_on) {
+            /* force tx path according to TTY mode when in call */
+            switch(adev->tty_mode) {
+                case TTY_MODE_FULL:
+                case TTY_MODE_HCO:
+                    /* tx path from headset mic */
+                    headphone_on = 0;
+                    headset_on = 1;
+                    speaker_on = 0;
+                    earpiece_on = 0;
+                    break;
+                case TTY_MODE_VCO:
+                    /* tx path from device sub mic */
+                    headphone_on = 0;
+                    headset_on = 0;
+                    speaker_on = 1;
+                    earpiece_on = 0;
+                    break;
+                case TTY_MODE_OFF:
+                default:
+                    break;
+            }
+        }
+
+        if (headset_on || headphone_on || speaker_on || earpiece_on) {
+            ALOGD("%s: set voicecall route: voicecall_default", __func__);
+            set_bigroute_by_array(adev->mixer, voicecall_default, 1);
+        } else {
+            ALOGD("%s: set voicecall route: voicecall_default_disable", __func__);
+            set_bigroute_by_array(adev->mixer, voicecall_default_disable, 1);
+        }
+
+        if (speaker_on || earpiece_on || headphone_on) {
+            ALOGD("%s: set voicecall route: default_input", __func__);
+            set_bigroute_by_array(adev->mixer, default_input, 1);
+        } else {
+            ALOGD("%s: set voicecall route: default_input_disable", __func__);
+            set_bigroute_by_array(adev->mixer, default_input_disable, 1);
+        }
+
+        if (headset_on) {
+            ALOGD("%s: set voicecall route: headset_input", __func__);
+            set_bigroute_by_array(adev->mixer, headset_input, 1);
+        } else {
+            ALOGD("%s: set voicecall route: headset_input_disable", __func__);
+            set_bigroute_by_array(adev->mixer, headset_input_disable, 1);
+        }
+
+        if (bt_on) {
+            // bt uses a different port (PORT_BT) for playback, reopen the pcms
+            end_call(adev);
+            start_call(adev);
+            ALOGD("%s: set voicecall route: bt_input", __func__);
+            set_bigroute_by_array(adev->mixer, bt_input, 1);
+            ALOGD("%s: set voicecall route: bt_output", __func__);
+            set_bigroute_by_array(adev->mixer, bt_output, 1);
+        } else {
+            ALOGD("%s: set voicecall route: bt_disable", __func__);
+            set_bigroute_by_array(adev->mixer, bt_disable, 1);
+        }
+
+        set_incall_device(adev);
+    }
+}
+
+static void select_input_device(struct m0_audio_device *adev)
+{
+    switch(adev->devices & AUDIO_DEVICE_IN_ALL) {
+        case AUDIO_DEVICE_IN_BUILTIN_MIC:
+            ALOGD("%s: AUDIO_DEVICE_IN_BUILTIN_MIC", __func__);
+            break;
+        case AUDIO_DEVICE_IN_BACK_MIC:
+            ALOGD("%s: AUDIO_DEVICE_IN_BACK_MIC", __func__);
+            break;
+        case AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET:
+            ALOGD("%s: AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET", __func__);
+            break;
+        case AUDIO_DEVICE_IN_WIRED_HEADSET:
+            ALOGD("%s: AUDIO_DEVICE_IN_WIRED_HEADSET", __func__);
+            break;
+        default:
+            break;
+    }
+
+    select_devices(adev);
+}
+
+/* must be called with hw device and output stream mutexes locked */
+static int start_output_stream_low_latency(struct m0_stream_out *out)
+{
+    struct m0_audio_device *adev = out->dev;
+    unsigned int flags = PCM_OUT;
+    int i;
+    bool success = true;
+
+    if (adev->mode != AUDIO_MODE_IN_CALL) {
+        select_output_device(adev);
+    }
+
+    /* default to low power: will be corrected in out_write if necessary before first write to
+     * tinyalsa.
+     */
+
+    if (adev->devices & (AUDIO_DEVICE_OUT_ALL &
+                         ~(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET | AUDIO_DEVICE_OUT_AUX_DIGITAL))) {
+        /* Something not a dock in use */
+        out->config[PCM_NORMAL] = pcm_config_tones;
+        out->config[PCM_NORMAL].rate = MM_FULL_POWER_SAMPLING_RATE;
+        out->pcm[PCM_NORMAL] = pcm_open(CARD_DEFAULT, PORT_PLAYBACK,
+                                            flags, &out->config[PCM_NORMAL]);
+    }
+
+    if (adev->devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
+        /* SPDIF output in use */
+        out->config[PCM_SPDIF] = pcm_config_tones;
+        out->config[PCM_SPDIF].rate = MM_FULL_POWER_SAMPLING_RATE;
+        out->pcm[PCM_SPDIF] = pcm_open(CARD_DEFAULT, PORT_PLAYBACK,
+                                           flags, &out->config[PCM_SPDIF]);
+    }
+
+    /* Close any PCMs that could not be opened properly and return an error */
+    for (i = 0; i < PCM_TOTAL; i++) {
+        if (out->pcm[i] && !pcm_is_ready(out->pcm[i])) {
+            ALOGE("%s: cannot open pcm_out driver %d: %s", __func__ , i, pcm_get_error(out->pcm[i]));
+            pcm_close(out->pcm[i]);
+            out->pcm[i] = NULL;
+            success = false;
+        }
+    }
+
+    if (success) {
+        out->buffer_frames = pcm_config_tones.period_size * 2;
+        if (out->buffer == NULL)
+            out->buffer = malloc(out->buffer_frames * audio_stream_frame_size(&out->stream.common));
+
+        if (adev->echo_reference != NULL)
+            out->echo_reference = adev->echo_reference;
+        out->resampler->reset(out->resampler);
+
+        return 0;
+    }
+
+    return -ENOMEM;
+}
+
+/* must be called with hw device and output stream mutexes locked */
+static int start_output_stream_deep_buffer(struct m0_stream_out *out)
+{
+    struct m0_audio_device *adev = out->dev;
+
+    if (adev->mode != AUDIO_MODE_IN_CALL) {
+        select_output_device(adev);
+    }
+
+    out->write_threshold = PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT * DEEP_BUFFER_LONG_PERIOD_SIZE;
+    out->use_long_periods = true;
+
+    out->config[PCM_NORMAL] = pcm_config_mm;
+    out->config[PCM_NORMAL].rate = MM_FULL_POWER_SAMPLING_RATE;
+    out->pcm[PCM_NORMAL] = pcm_open(CARD_DEFAULT, PORT_PLAYBACK,
+                                        PCM_OUT | PCM_MMAP | PCM_NOIRQ, &out->config[PCM_NORMAL]);
+    if (out->pcm[PCM_NORMAL] && !pcm_is_ready(out->pcm[PCM_NORMAL])) {
+        ALOGE("%s: cannot open pcm_out driver: %s", __func__, pcm_get_error(out->pcm[PCM_NORMAL]));
+        pcm_close(out->pcm[PCM_NORMAL]);
+        out->pcm[PCM_NORMAL] = NULL;
+        return -ENOMEM;
+    }
+    out->buffer_frames = DEEP_BUFFER_SHORT_PERIOD_SIZE * 2;
+    if (out->buffer == NULL)
+        out->buffer = malloc(PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT * DEEP_BUFFER_LONG_PERIOD_SIZE);
+
+    return 0;
+}
+
+static int check_input_parameters(uint32_t sample_rate, audio_format_t format, int channel_count)
+{
+    if (format != AUDIO_FORMAT_PCM_16_BIT)
+        return -EINVAL;
+
+    if ((channel_count < 1) || (channel_count > 2))
+        return -EINVAL;
+
+    switch(sample_rate) {
+    case 8000:
+    case 11025:
+    case 16000:
+    case 22050:
+    case 24000:
+    case 32000:
+    case 44100:
+    case 48000:
+        break;
+    default:
+        return -EINVAL;
+    }
+
+    return 0;
+}
+
+static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format, int channel_count)
+{
+    size_t size;
+    size_t device_rate;
+
+    if (check_input_parameters(sample_rate, format, channel_count) != 0)
+        return 0;
+
+    /* take resampling into account and return the closest majoring
+    multiple of 16 frames, as audioflinger expects audio buffers to
+    be a multiple of 16 frames */
+    size = (pcm_config_capture.period_size * sample_rate) / pcm_config_capture.rate;
+    size = ((size + 15) / 16) * 16;
+
+    return size * channel_count * sizeof(short);
+}
+
+static void add_echo_reference(struct m0_stream_out *out,
+                               struct echo_reference_itfe *reference)
+{
+    pthread_mutex_lock(&out->lock);
+    out->echo_reference = reference;
+    pthread_mutex_unlock(&out->lock);
+}
+
+static void remove_echo_reference(struct m0_stream_out *out,
+                                  struct echo_reference_itfe *reference)
+{
+    pthread_mutex_lock(&out->lock);
+    if (out->echo_reference == reference) {
+        /* stop writing to echo reference */
+        reference->write(reference, NULL);
+        out->echo_reference = NULL;
+    }
+    pthread_mutex_unlock(&out->lock);
+}
+
+static void put_echo_reference(struct m0_audio_device *adev,
+                          struct echo_reference_itfe *reference)
+{
+    if (adev->echo_reference != NULL &&
+            reference == adev->echo_reference) {
+        /* echo reference is taken from the low latency output stream used
+         * for voice use cases */
+        if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL &&
+                !adev->outputs[OUTPUT_LOW_LATENCY]->standby)
+            remove_echo_reference(adev->outputs[OUTPUT_LOW_LATENCY], reference);
+        release_echo_reference(reference);
+        adev->echo_reference = NULL;
+    }
+}
+
+static struct echo_reference_itfe *get_echo_reference(struct m0_audio_device *adev,
+                                               audio_format_t format,
+                                               uint32_t channel_count,
+                                               uint32_t sampling_rate)
+{
+    put_echo_reference(adev, adev->echo_reference);
+    /* echo reference is taken from the low latency output stream used
+     * for voice use cases */
+    if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL &&
+            !adev->outputs[OUTPUT_LOW_LATENCY]->standby) {
+        struct audio_stream *stream =
+                &adev->outputs[OUTPUT_LOW_LATENCY]->stream.common;
+        uint32_t wr_channel_count = popcount(stream->get_channels(stream));
+        uint32_t wr_sampling_rate = stream->get_sample_rate(stream);
+
+        int status = create_echo_reference(AUDIO_FORMAT_PCM_16_BIT,
+                                           channel_count,
+                                           sampling_rate,
+                                           AUDIO_FORMAT_PCM_16_BIT,
+                                           wr_channel_count,
+                                           wr_sampling_rate,
+                                           &adev->echo_reference);
+        if (status == 0)
+            add_echo_reference(adev->outputs[OUTPUT_LOW_LATENCY],
+                               adev->echo_reference);
+    }
+    return adev->echo_reference;
+}
+
+static int get_playback_delay(struct m0_stream_out *out,
+                       size_t frames,
+                       struct echo_reference_buffer *buffer)
+{
+    size_t kernel_frames;
+    int status;
+    int primary_pcm = 0;
+
+    /* Find the first active PCM to act as primary */
+    while ((primary_pcm < PCM_TOTAL) && !out->pcm[primary_pcm])
+        primary_pcm++;
+
+    status = pcm_get_htimestamp(out->pcm[primary_pcm], &kernel_frames, &buffer->time_stamp);
+    if (status < 0) {
+        buffer->time_stamp.tv_sec  = 0;
+        buffer->time_stamp.tv_nsec = 0;
+        buffer->delay_ns           = 0;
+        ALOGV("%s: pcm_get_htimestamp error,"
+                "setting playbackTimestamp to 0", __func__);
+        return status;
+    }
+
+    kernel_frames = pcm_get_buffer_size(out->pcm[primary_pcm]) - kernel_frames;
+
+    /* adjust render time stamp with delay added by current driver buffer.
+     * Add the duration of current frame as we want the render time of the last
+     * sample being written. */
+    buffer->delay_ns = (long)(((int64_t)(kernel_frames + frames)* 1000000000)/
+                            MM_FULL_POWER_SAMPLING_RATE);
+
+    return 0;
+}
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    return DEFAULT_OUT_SAMPLING_RATE;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return 0;
+}
+
+static size_t out_get_buffer_size_low_latency(const struct audio_stream *stream)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+
+    /* take resampling into account and return the closest majoring
+    multiple of 16 frames, as audioflinger expects audio buffers to
+    be a multiple of 16 frames. Note: we use the default rate here
+    from pcm_config_tones.rate. */
+    size_t size = (SHORT_PERIOD_SIZE * DEFAULT_OUT_SAMPLING_RATE) / pcm_config_tones.rate;
+    size = ((size + 15) / 16) * 16;
+    return size * audio_stream_frame_size((struct audio_stream *)stream);
+}
+
+static size_t out_get_buffer_size_deep_buffer(const struct audio_stream *stream)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+
+    /* take resampling into account and return the closest majoring
+    multiple of 16 frames, as audioflinger expects audio buffers to
+    be a multiple of 16 frames. Note: we use the default rate here
+    from pcm_config_mm.rate. */
+    size_t size = (DEEP_BUFFER_SHORT_PERIOD_SIZE * DEFAULT_OUT_SAMPLING_RATE) /
+                        pcm_config_mm.rate;
+    size = ((size + 15) / 16) * 16;
+    return size * audio_stream_frame_size((struct audio_stream *)stream);
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+
+    return out->channel_mask;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return 0;
+}
+
+/* must be called with hw device and output stream mutexes locked */
+static int do_output_standby(struct m0_stream_out *out)
+{
+    struct m0_audio_device *adev = out->dev;
+    int i;
+    bool all_outputs_in_standby = true;
+
+    if (!out->standby) {
+        out->standby = 1;
+
+        for (i = 0; i < PCM_TOTAL; i++) {
+            if (out->pcm[i]) {
+                pcm_close(out->pcm[i]);
+                out->pcm[i] = NULL;
+            }
+        }
+
+        for (i = 0; i < OUTPUT_TOTAL; i++) {
+            if (adev->outputs[i] != NULL && !adev->outputs[i]->standby) {
+                all_outputs_in_standby = false;
+                break;
+            }
+        }
+
+        /* force standby on low latency output stream so that it can reuse HDMI driver if
+         * necessary when restarted */
+        if (out == adev->outputs[OUTPUT_HDMI]) {
+            if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL &&
+                    !adev->outputs[OUTPUT_LOW_LATENCY]->standby) {
+                struct m0_stream_out *ll_out = adev->outputs[OUTPUT_LOW_LATENCY];
+                pthread_mutex_lock(&ll_out->lock);
+                do_output_standby(ll_out);
+                pthread_mutex_unlock(&ll_out->lock);
+            }
+        }
+
+        /* stop writing to echo reference */
+        if (out->echo_reference != NULL) {
+            out->echo_reference->write(out->echo_reference, NULL);
+            out->echo_reference = NULL;
+        }
+    }
+    return 0;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+    int status;
+
+    pthread_mutex_lock(&out->dev->lock);
+    pthread_mutex_lock(&out->lock);
+    status = do_output_standby(out);
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&out->dev->lock);
+    return status;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+    struct m0_audio_device *adev = out->dev;
+    struct m0_stream_in *in;
+    struct str_parms *parms;
+    char *str;
+    char value[32];
+    int ret, val = 0;
+    bool force_input_standby = false;
+
+    parms = str_parms_create_str(kvpairs);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        val = atoi(value);
+        pthread_mutex_lock(&adev->lock);
+        pthread_mutex_lock(&out->lock);
+        if (((adev->devices & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) {
+            /* this is needed only when changing device on low latency output
+             * as other output streams are not used for voice use cases nor
+             * handle duplication to HDMI or SPDIF */
+            if (out == adev->outputs[OUTPUT_LOW_LATENCY] && !out->standby) {
+                /* a change in output device may change the microphone selection */
+                if (adev->active_input &&
+                        adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
+                    force_input_standby = true;
+                }
+                /* force standby if moving to/from HDMI/SPDIF or if the output
+                 * device changes when in HDMI/SPDIF mode */
+                /* FIXME also force standby when in call as some audio path switches do not work
+                 * while in call and an output stream is active (e.g BT SCO => earpiece) */
+
+                /* FIXME workaround for audio being dropped when switching path without forcing standby
+                 * (several hundred ms of audio can be lost: e.g beginning of a ringtone. We must understand
+                 * the root cause in audio HAL, driver or ABE.
+                if (((val & AUDIO_DEVICE_OUT_AUX_DIGITAL) ^
+                        (adev->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)) ||
+                        ((val & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) ^
+                        (adev->devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) ||
+                        (adev->devices & (AUDIO_DEVICE_OUT_AUX_DIGITAL |
+                                         AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)))
+                */
+                if (((val & AUDIO_DEVICE_OUT_AUX_DIGITAL) ^
+                        (adev->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)) ||
+                        ((val & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) ^
+                        (adev->devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) ||
+                        (adev->devices & (AUDIO_DEVICE_OUT_AUX_DIGITAL |
+                                         AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) ||
+                        ((val & AUDIO_DEVICE_OUT_SPEAKER) ^
+                        (adev->devices & AUDIO_DEVICE_OUT_SPEAKER)) ||
+                        (adev->mode == AUDIO_MODE_IN_CALL))
+                    do_output_standby(out);
+            }
+            if (out != adev->outputs[OUTPUT_HDMI]) {
+                adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
+                adev->devices |= val;
+                select_output_device(adev);
+            }
+        }
+        pthread_mutex_unlock(&out->lock);
+        if (force_input_standby) {
+            in = adev->active_input;
+            pthread_mutex_lock(&in->lock);
+            do_input_standby(in);
+            pthread_mutex_unlock(&in->lock);
+        }
+        pthread_mutex_unlock(&adev->lock);
+    }
+
+    str_parms_destroy(parms);
+    return ret;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+
+    struct str_parms *query = str_parms_create_str(keys);
+    char *str;
+    char value[256];
+    struct str_parms *reply = str_parms_create();
+    size_t i, j;
+    int ret;
+    bool first = true;
+
+    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
+    if (ret >= 0) {
+        value[0] = '\0';
+        i = 0;
+        while (out->sup_channel_masks[i] != 0) {
+            for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
+                if (out_channels_name_to_enum_table[j].value == out->sup_channel_masks[i]) {
+                    if (!first) {
+                        strcat(value, "|");
+                    }
+                    strcat(value, out_channels_name_to_enum_table[j].name);
+                    first = false;
+                    break;
+                }
+            }
+            i++;
+        }
+        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
+        str = strdup(str_parms_to_str(reply));
+    } else {
+        str = strdup(keys);
+    }
+    str_parms_destroy(query);
+    str_parms_destroy(reply);
+    return str;
+}
+
+static uint32_t out_get_latency_low_latency(const struct audio_stream_out *stream)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+
+    /*  Note: we use the default rate here from pcm_config_mm.rate */
+    return (SHORT_PERIOD_SIZE * PLAYBACK_SHORT_PERIOD_COUNT * 1000) / pcm_config_tones.rate;
+}
+
+static uint32_t out_get_latency_deep_buffer(const struct audio_stream_out *stream)
+{
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+
+    /*  Note: we use the default rate here from pcm_config_mm.rate */
+    return (DEEP_BUFFER_LONG_PERIOD_SIZE * PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT * 1000) /
+                    pcm_config_mm.rate;
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    return -ENOSYS;
+}
+
+static ssize_t out_write_low_latency(struct audio_stream_out *stream, const void* buffer,
+                         size_t bytes)
+{
+    int ret;
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+    struct m0_audio_device *adev = out->dev;
+    size_t frame_size = audio_stream_frame_size(&out->stream.common);
+    size_t in_frames = bytes / frame_size;
+    size_t out_frames = in_frames;
+    bool force_input_standby = false;
+    struct m0_stream_in *in;
+    int i;
+
+    /* acquiring hw device mutex systematically is useful if a low priority thread is waiting
+     * on the output stream mutex - e.g. executing select_mode() while holding the hw device
+     * mutex
+     */
+    pthread_mutex_lock(&adev->lock);
+    pthread_mutex_lock(&out->lock);
+    if (out->standby) {
+        ret = start_output_stream_low_latency(out);
+        if (ret != 0) {
+            pthread_mutex_unlock(&adev->lock);
+            goto exit;
+        }
+        out->standby = 0;
+        /* a change in output device may change the microphone selection */
+        if (adev->active_input &&
+                adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION)
+            force_input_standby = true;
+    }
+    pthread_mutex_unlock(&adev->lock);
+
+    for (i = 0; i < PCM_TOTAL; i++) {
+        /* only use resampler if required */
+        if (out->pcm[i] && (out->config[i].rate != DEFAULT_OUT_SAMPLING_RATE)) {
+            out_frames = out->buffer_frames;
+            out->resampler->resample_from_input(out->resampler,
+                                                (int16_t *)buffer,
+                                                &in_frames,
+                                                (int16_t *)out->buffer,
+                                                &out_frames);
+            break;
+        }
+    }
+
+    if (out->echo_reference != NULL) {
+        struct echo_reference_buffer b;
+        b.raw = (void *)buffer;
+        b.frame_count = in_frames;
+
+        get_playback_delay(out, out_frames, &b);
+        out->echo_reference->write(out->echo_reference, &b);
+    }
+
+    /* Write to all active PCMs */
+    for (i = 0; i < PCM_TOTAL; i++) {
+        if (out->pcm[i]) {
+            if (out->config[i].rate == DEFAULT_OUT_SAMPLING_RATE) {
+                /* PCM uses native sample rate */
+                ret = PCM_WRITE(out->pcm[i], (void *)buffer, bytes);
+            } else {
+                /* PCM needs resampler */
+                ret = PCM_WRITE(out->pcm[i], (void *)out->buffer, out_frames * frame_size);
+            }
+            if (ret)
+                break;
+        }
+    }
+
+exit:
+    pthread_mutex_unlock(&out->lock);
+
+    if (ret != 0) {
+        usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
+               out_get_sample_rate(&stream->common));
+    }
+
+    if (force_input_standby) {
+        pthread_mutex_lock(&adev->lock);
+        if (adev->active_input) {
+            in = adev->active_input;
+            pthread_mutex_lock(&in->lock);
+            do_input_standby(in);
+            pthread_mutex_unlock(&in->lock);
+        }
+        pthread_mutex_unlock(&adev->lock);
+    }
+
+    return bytes;
+}
+
+static ssize_t out_write_deep_buffer(struct audio_stream_out *stream, const void* buffer,
+                         size_t bytes)
+{
+    int ret;
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+    struct m0_audio_device *adev = out->dev;
+    size_t frame_size = audio_stream_frame_size(&out->stream.common);
+    size_t in_frames = bytes / frame_size;
+    size_t out_frames;
+    bool use_long_periods;
+    int kernel_frames;
+    void *buf;
+
+    /* acquiring hw device mutex systematically is useful if a low priority thread is waiting
+     * on the output stream mutex - e.g. executing select_mode() while holding the hw device
+     * mutex
+     */
+    pthread_mutex_lock(&adev->lock);
+    pthread_mutex_lock(&out->lock);
+    if (out->standby) {
+        ret = start_output_stream_deep_buffer(out);
+        if (ret != 0) {
+            pthread_mutex_unlock(&adev->lock);
+            goto exit;
+        }
+        out->standby = 0;
+    }
+    use_long_periods = adev->screen_off && !adev->active_input;
+    pthread_mutex_unlock(&adev->lock);
+
+    if (use_long_periods != out->use_long_periods) {
+        size_t period_size;
+        size_t period_count;
+
+        if (use_long_periods) {
+            period_size = DEEP_BUFFER_LONG_PERIOD_SIZE;
+            period_count = PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT;
+        } else {
+            period_size = DEEP_BUFFER_SHORT_PERIOD_SIZE;
+            period_count = PLAYBACK_DEEP_BUFFER_SHORT_PERIOD_COUNT;
+        }
+        out->write_threshold = period_size * period_count;
+        pcm_set_avail_min(out->pcm[PCM_NORMAL], period_size);
+        out->use_long_periods = use_long_periods;
+    }
+
+    /* only use resampler if required */
+    if (out->config[PCM_NORMAL].rate != DEFAULT_OUT_SAMPLING_RATE) {
+        out_frames = out->buffer_frames;
+        out->resampler->resample_from_input(out->resampler,
+                                            (int16_t *)buffer,
+                                            &in_frames,
+                                            (int16_t *)out->buffer,
+                                            &out_frames);
+        buf = (void *)out->buffer;
+    } else {
+        out_frames = in_frames;
+        buf = (void *)buffer;
+    }
+
+    /* do not allow more than out->write_threshold frames in kernel pcm driver buffer */
+    do {
+        struct timespec time_stamp;
+
+        if (pcm_get_htimestamp(out->pcm[PCM_NORMAL],
+                               (unsigned int *)&kernel_frames, &time_stamp) < 0)
+            break;
+        kernel_frames = pcm_get_buffer_size(out->pcm[PCM_NORMAL]) - kernel_frames;
+
+        if (kernel_frames > out->write_threshold) {
+            unsigned long time = (unsigned long)
+                    (((int64_t)(kernel_frames - out->write_threshold) * 1000000) /
+                            MM_FULL_POWER_SAMPLING_RATE);
+            if (time < MIN_WRITE_SLEEP_US)
+                time = MIN_WRITE_SLEEP_US;
+            usleep(time);
+        }
+    } while (kernel_frames > out->write_threshold);
+
+    ret = pcm_mmap_write(out->pcm[PCM_NORMAL], buf, out_frames * frame_size);
+
+exit:
+    pthread_mutex_unlock(&out->lock);
+
+    if (ret != 0) {
+        usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
+               out_get_sample_rate(&stream->common));
+    }
+
+    return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    return -EINVAL;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+/** audio_stream_in implementation **/
+
+/* must be called with hw device and input stream mutexes locked */
+static int start_input_stream(struct m0_stream_in *in)
+{
+    int ret = 0;
+    struct m0_audio_device *adev = in->dev;
+
+    adev->active_input = in;
+
+    if (adev->mode != AUDIO_MODE_IN_CALL) {
+        adev->devices &= ~AUDIO_DEVICE_IN_ALL;
+        adev->devices |= in->device;
+        select_input_device(adev);
+    }
+
+    if (in->aux_channels_changed)
+    {
+        in->aux_channels_changed = false;
+        in->config.channels = popcount(in->main_channels | in->aux_channels);
+
+        if (in->resampler) {
+            /* release and recreate the resampler with the new number of channel of the input */
+            release_resampler(in->resampler);
+            in->resampler = NULL;
+            ret = create_resampler(in->config.rate,
+                               in->requested_rate,
+                               in->config.channels,
+                               RESAMPLER_QUALITY_DEFAULT,
+                               &in->buf_provider,
+                               &in->resampler);
+        }
+        ALOGV("%s: New channel configuration, "
+                "main_channels = [%04x], aux_channels = [%04x], config.channels = %d",
+                __func__, in->main_channels, in->aux_channels, in->config.channels);
+    }
+
+    if (in->need_echo_reference && in->echo_reference == NULL)
+        in->echo_reference = get_echo_reference(adev,
+                                        AUDIO_FORMAT_PCM_16_BIT,
+                                        in->config.channels,
+                                        in->requested_rate);
+
+    /* this assumes routing is done previously */
+    in->pcm = pcm_open(CARD_DEFAULT, PORT_CAPTURE, PCM_IN, &in->config);
+    if (!pcm_is_ready(in->pcm)) {
+        ALOGE("cannot open pcm_in driver: %s", pcm_get_error(in->pcm));
+        pcm_close(in->pcm);
+        adev->active_input = NULL;
+        return -ENOMEM;
+    }
+
+    /* force read and proc buf reallocation case of frame size or channel count change */
+    in->read_buf_frames = 0;
+    in->read_buf_size = 0;
+    in->proc_buf_frames = 0;
+    in->proc_buf_size = 0;
+    /* if no supported sample rate is available, use the resampler */
+    if (in->resampler) {
+        in->resampler->reset(in->resampler);
+    }
+    return 0;
+}
+
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+
+    return in->requested_rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return 0;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+
+    return get_input_buffer_size(in->requested_rate,
+                                 AUDIO_FORMAT_PCM_16_BIT,
+                                 popcount(in->main_channels));
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+
+    return in->main_channels;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return 0;
+}
+
+/* must be called with hw device and input stream mutexes locked */
+static int do_input_standby(struct m0_stream_in *in)
+{
+    struct m0_audio_device *adev = in->dev;
+
+    if (!in->standby) {
+        pcm_close(in->pcm);
+        in->pcm = NULL;
+
+        adev->active_input = 0;
+        if (adev->mode != AUDIO_MODE_IN_CALL) {
+            adev->devices &= ~AUDIO_DEVICE_IN_ALL;
+            select_input_device(adev);
+        }
+
+        if (in->echo_reference != NULL) {
+            /* stop reading from echo reference */
+            in->echo_reference->read(in->echo_reference, NULL);
+            put_echo_reference(adev, in->echo_reference);
+            in->echo_reference = NULL;
+        }
+
+        in->standby = 1;
+    }
+    return 0;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+    int status;
+
+    pthread_mutex_lock(&in->dev->lock);
+    pthread_mutex_lock(&in->lock);
+    status = do_input_standby(in);
+    pthread_mutex_unlock(&in->lock);
+    pthread_mutex_unlock(&in->dev->lock);
+    return status;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+    struct m0_audio_device *adev = in->dev;
+    struct str_parms *parms;
+    char *str;
+    char value[32];
+    int ret, val = 0;
+    bool do_standby = false;
+
+    parms = str_parms_create_str(kvpairs);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
+
+    pthread_mutex_lock(&adev->lock);
+    pthread_mutex_lock(&in->lock);
+    if (ret >= 0) {
+        val = atoi(value);
+        /* no audio source uses val == 0 */
+        if ((in->source != val) && (val != 0)) {
+            in->source = val;
+            do_standby = true;
+        }
+    }
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        val = atoi(value);
+        if ((in->device != val) && (val != 0)) {
+            in->device = val;
+            do_standby = true;
+            /* make sure new device selection is incompatible with multi-mic pre processing
+             * configuration */
+            in_update_aux_channels(in, NULL);
+        }
+    }
+
+    if (do_standby)
+        do_input_standby(in);
+    pthread_mutex_unlock(&in->lock);
+    pthread_mutex_unlock(&adev->lock);
+
+    str_parms_destroy(parms);
+    return ret;
+}
+
+static char * in_get_parameters(const struct audio_stream *stream,
+                                const char *keys)
+{
+    return strdup("");
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain)
+{
+    return 0;
+}
+
+static void get_capture_delay(struct m0_stream_in *in,
+                       size_t frames,
+                       struct echo_reference_buffer *buffer)
+{
+
+    /* read frames available in kernel driver buffer */
+    size_t kernel_frames;
+    struct timespec tstamp;
+    long buf_delay;
+    long rsmp_delay;
+    long kernel_delay;
+    long delay_ns;
+
+    if (pcm_get_htimestamp(in->pcm, &kernel_frames, &tstamp) < 0) {
+        buffer->time_stamp.tv_sec  = 0;
+        buffer->time_stamp.tv_nsec = 0;
+        buffer->delay_ns           = 0;
+        ALOGW("%s: pcm_htimestamp error", __func__);
+        return;
+    }
+
+    /* read frames available in audio HAL input buffer
+     * add number of frames being read as we want the capture time of first sample
+     * in current buffer */
+    /* frames in in->buffer are at driver sampling rate while frames in in->proc_buf are
+     * at requested sampling rate */
+    buf_delay = (long)(((int64_t)(in->read_buf_frames) * 1000000000) / in->config.rate +
+                       ((int64_t)(in->proc_buf_frames) * 1000000000) /
+                           in->requested_rate);
+
+    /* add delay introduced by resampler */
+    rsmp_delay = 0;
+    if (in->resampler) {
+        rsmp_delay = in->resampler->delay_ns(in->resampler);
+    }
+
+    kernel_delay = (long)(((int64_t)kernel_frames * 1000000000) / in->config.rate);
+
+    delay_ns = kernel_delay + buf_delay + rsmp_delay;
+
+    buffer->time_stamp = tstamp;
+    buffer->delay_ns   = delay_ns;
+    ALOGV("%s: time_stamp = [%ld].[%ld], delay_ns: [%d],"
+         " kernel_delay:[%ld], buf_delay:[%ld], rsmp_delay:[%ld], kernel_frames:[%d], "
+         "in->read_buf_frames:[%d], in->proc_buf_frames:[%d], frames:[%d]",
+         __func__, buffer->time_stamp.tv_sec , buffer->time_stamp.tv_nsec, buffer->delay_ns,
+         kernel_delay, buf_delay, rsmp_delay, kernel_frames,
+         in->read_buf_frames, in->proc_buf_frames, frames);
+
+}
+
+static int32_t update_echo_reference(struct m0_stream_in *in, size_t frames)
+{
+    struct echo_reference_buffer b;
+    b.delay_ns = 0;
+
+    ALOGV("%s: frames = [%d], in->ref_frames_in = [%d],  "
+          "b.frame_count = [%d]",
+         __func__, frames, in->ref_buf_frames, frames - in->ref_buf_frames);
+    if (in->ref_buf_frames < frames) {
+        if (in->ref_buf_size < frames) {
+            in->ref_buf_size = frames;
+            in->ref_buf = (int16_t *)realloc(in->ref_buf, pcm_frames_to_bytes(in->pcm, frames));
+            ALOG_ASSERT((in->ref_buf != NULL),
+                        "%s failed to reallocate ref_buf", __func__);
+            ALOGV("%s: ref_buf %p extended to %d bytes",
+                      __func__, in->ref_buf, pcm_frames_to_bytes(in->pcm, frames));
+        }
+        b.frame_count = frames - in->ref_buf_frames;
+        b.raw = (void *)(in->ref_buf + in->ref_buf_frames * in->config.channels);
+
+        get_capture_delay(in, frames, &b);
+
+        if (in->echo_reference->read(in->echo_reference, &b) == 0)
+        {
+            in->ref_buf_frames += b.frame_count;
+            ALOGD("%s: in->ref_buf_frames:[%d], "
+                    "in->ref_buf_size:[%d], frames:[%d], b.frame_count:[%d]",
+                 __func__, in->ref_buf_frames, in->ref_buf_size, frames, b.frame_count);
+        }
+    } else
+        ALOGW("%s: NOT enough frames to read ref buffer", __func__);
+    return b.delay_ns;
+}
+
+static int set_preprocessor_param(effect_handle_t handle,
+                           effect_param_t *param)
+{
+    uint32_t size = sizeof(int);
+    uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
+                        param->vsize;
+
+    int status = (*handle)->command(handle,
+                                   EFFECT_CMD_SET_PARAM,
+                                   sizeof (effect_param_t) + psize,
+                                   param,
+                                   &size,
+                                   &param->status);
+    if (status == 0)
+        status = param->status;
+
+    return status;
+}
+
+static int set_preprocessor_echo_delay(effect_handle_t handle,
+                                     int32_t delay_us)
+{
+    uint32_t buf[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
+    effect_param_t *param = (effect_param_t *)buf;
+
+    param->psize = sizeof(uint32_t);
+    param->vsize = sizeof(uint32_t);
+    *(uint32_t *)param->data = AEC_PARAM_ECHO_DELAY;
+    *((int32_t *)param->data + 1) = delay_us;
+
+    return set_preprocessor_param(handle, param);
+}
+
+static void push_echo_reference(struct m0_stream_in *in, size_t frames)
+{
+    /* read frames from echo reference buffer and update echo delay
+     * in->ref_buf_frames is updated with frames available in in->ref_buf */
+    int32_t delay_us = update_echo_reference(in, frames)/1000;
+    int i;
+    audio_buffer_t buf;
+
+    if (in->ref_buf_frames < frames)
+        frames = in->ref_buf_frames;
+
+    buf.frameCount = frames;
+    buf.raw = in->ref_buf;
+
+    for (i = 0; i < in->num_preprocessors; i++) {
+        if ((*in->preprocessors[i].effect_itfe)->process_reverse == NULL)
+            continue;
+
+        (*in->preprocessors[i].effect_itfe)->process_reverse(in->preprocessors[i].effect_itfe,
+                                               &buf,
+                                               NULL);
+        set_preprocessor_echo_delay(in->preprocessors[i].effect_itfe, delay_us);
+    }
+
+    in->ref_buf_frames -= buf.frameCount;
+    if (in->ref_buf_frames) {
+        memcpy(in->ref_buf,
+               in->ref_buf + buf.frameCount * in->config.channels,
+               in->ref_buf_frames * in->config.channels * sizeof(int16_t));
+    }
+}
+
+static int get_next_buffer(struct resampler_buffer_provider *buffer_provider,
+                                   struct resampler_buffer* buffer)
+{
+    struct m0_stream_in *in;
+
+    if (buffer_provider == NULL || buffer == NULL)
+        return -EINVAL;
+
+    in = (struct m0_stream_in *)((char *)buffer_provider -
+                                   offsetof(struct m0_stream_in, buf_provider));
+
+    if (in->pcm == NULL) {
+        buffer->raw = NULL;
+        buffer->frame_count = 0;
+        in->read_status = -ENODEV;
+        return -ENODEV;
+    }
+
+    if (in->read_buf_frames == 0) {
+        size_t size_in_bytes = pcm_frames_to_bytes(in->pcm, in->config.period_size);
+        if (in->read_buf_size < in->config.period_size) {
+            in->read_buf_size = in->config.period_size;
+            in->read_buf = (int16_t *) realloc(in->read_buf, size_in_bytes);
+            ALOG_ASSERT((in->read_buf != NULL),
+                        "%s failed to reallocate read_buf", __func__);
+            ALOGV("%s: read_buf %p extended to %d bytes",
+                  __func__, in->read_buf, size_in_bytes);
+        }
+
+        in->read_status = pcm_read(in->pcm, (void*)in->read_buf, size_in_bytes);
+
+        if (in->read_status != 0) {
+            ALOGE("%s: pcm_read error %d", __func__, in->read_status);
+            buffer->raw = NULL;
+            buffer->frame_count = 0;
+            return in->read_status;
+        }
+        in->read_buf_frames = in->config.period_size;
+    }
+
+    buffer->frame_count = (buffer->frame_count > in->read_buf_frames) ?
+                                in->read_buf_frames : buffer->frame_count;
+    buffer->i16 = in->read_buf + (in->config.period_size - in->read_buf_frames) *
+                                                in->config.channels;
+
+    return in->read_status;
+
+}
+
+static void release_buffer(struct resampler_buffer_provider *buffer_provider,
+                                  struct resampler_buffer* buffer)
+{
+    struct m0_stream_in *in;
+
+    if (buffer_provider == NULL || buffer == NULL)
+        return;
+
+    in = (struct m0_stream_in *)((char *)buffer_provider -
+                                   offsetof(struct m0_stream_in, buf_provider));
+
+    in->read_buf_frames -= buffer->frame_count;
+}
+
+/* read_frames() reads frames from kernel driver, down samples to capture rate
+ * if necessary and output the number of frames requested to the buffer specified */
+static ssize_t read_frames(struct m0_stream_in *in, void *buffer, ssize_t frames)
+{
+    ssize_t frames_wr = 0;
+
+    while (frames_wr < frames) {
+        size_t frames_rd = frames - frames_wr;
+        if (in->resampler != NULL) {
+            in->resampler->resample_from_provider(in->resampler,
+                                                  (int16_t *)((char *)buffer +
+                                                      pcm_frames_to_bytes(in->pcm ,frames_wr)),
+                                                  &frames_rd);
+
+        } else {
+            struct resampler_buffer buf = {
+                    { raw : NULL, },
+                    frame_count : frames_rd,
+            };
+            get_next_buffer(&in->buf_provider, &buf);
+            if (buf.raw != NULL) {
+                memcpy((char *)buffer +
+                            pcm_frames_to_bytes(in->pcm, frames_wr),
+                        buf.raw,
+                        pcm_frames_to_bytes(in->pcm, buf.frame_count));
+                frames_rd = buf.frame_count;
+            }
+            release_buffer(&in->buf_provider, &buf);
+        }
+        /* in->read_status is updated by getNextBuffer() also called by
+         * in->resampler->resample_from_provider() */
+        if (in->read_status != 0)
+            return in->read_status;
+
+        frames_wr += frames_rd;
+    }
+    return frames_wr;
+}
+
+/* process_frames() reads frames from kernel driver (via read_frames()),
+ * calls the active audio pre processings and output the number of frames requested
+ * to the buffer specified */
+static ssize_t process_frames(struct m0_stream_in *in, void* buffer, ssize_t frames)
+{
+    ssize_t frames_wr = 0;
+    audio_buffer_t in_buf;
+    audio_buffer_t out_buf;
+    int i;
+    bool has_aux_channels = (~in->main_channels & in->aux_channels);
+    void *proc_buf_out;
+
+    if (has_aux_channels)
+        proc_buf_out = in->proc_buf_out;
+    else
+        proc_buf_out = buffer;
+
+    /* since all the processing below is done in frames and using the config.channels
+     * as the number of channels, no changes is required in case aux_channels are present */
+    while (frames_wr < frames) {
+        /* first reload enough frames at the end of process input buffer */
+        if (in->proc_buf_frames < (size_t)frames) {
+            ssize_t frames_rd;
+
+            if (in->proc_buf_size < (size_t)frames) {
+                size_t size_in_bytes = pcm_frames_to_bytes(in->pcm, frames);
+
+                in->proc_buf_size = (size_t)frames;
+                in->proc_buf_in = (int16_t *)realloc(in->proc_buf_in, size_in_bytes);
+                ALOG_ASSERT((in->proc_buf_in != NULL),
+                            "%s failed to reallocate proc_buf_in", __func__);
+                if (has_aux_channels) {
+                    in->proc_buf_out = (int16_t *)realloc(in->proc_buf_out, size_in_bytes);
+                    ALOG_ASSERT((in->proc_buf_out != NULL),
+                                "%s failed to reallocate proc_buf_out", __func__);
+                    proc_buf_out = in->proc_buf_out;
+                }
+                ALOGV("process_frames(): proc_buf_in %p extended to %d bytes",
+                     in->proc_buf_in, size_in_bytes);
+            }
+            frames_rd = read_frames(in,
+                                    in->proc_buf_in +
+                                        in->proc_buf_frames * in->config.channels,
+                                    frames - in->proc_buf_frames);
+            if (frames_rd < 0) {
+                frames_wr = frames_rd;
+                break;
+            }
+            in->proc_buf_frames += frames_rd;
+        }
+
+        if (in->echo_reference != NULL)
+            push_echo_reference(in, in->proc_buf_frames);
+
+         /* in_buf.frameCount and out_buf.frameCount indicate respectively
+          * the maximum number of frames to be consumed and produced by process() */
+        in_buf.frameCount = in->proc_buf_frames;
+        in_buf.s16 = in->proc_buf_in;
+        out_buf.frameCount = frames - frames_wr;
+        out_buf.s16 = (int16_t *)proc_buf_out + frames_wr * in->config.channels;
+
+        /* FIXME: this works because of current pre processing library implementation that
+         * does the actual process only when the last enabled effect process is called.
+         * The generic solution is to have an output buffer for each effect and pass it as
+         * input to the next.
+         */
+        for (i = 0; i < in->num_preprocessors; i++) {
+            (*in->preprocessors[i].effect_itfe)->process(in->preprocessors[i].effect_itfe,
+                                               &in_buf,
+                                               &out_buf);
+        }
+
+        /* process() has updated the number of frames consumed and produced in
+         * in_buf.frameCount and out_buf.frameCount respectively
+         * move remaining frames to the beginning of in->proc_buf_in */
+        in->proc_buf_frames -= in_buf.frameCount;
+
+        if (in->proc_buf_frames) {
+            memcpy(in->proc_buf_in,
+                   in->proc_buf_in + in_buf.frameCount * in->config.channels,
+                   in->proc_buf_frames * in->config.channels * sizeof(int16_t));
+        }
+
+        /* if not enough frames were passed to process(), read more and retry. */
+        if (out_buf.frameCount == 0) {
+            ALOGW("No frames produced by preproc");
+            continue;
+        }
+
+        if ((frames_wr + (ssize_t)out_buf.frameCount) <= frames) {
+            frames_wr += out_buf.frameCount;
+        } else {
+            /* The effect does not comply to the API. In theory, we should never end up here! */
+            ALOGE("%s: preprocessing produced too many frames: %d + %d  > %d !", __func__,
+                  (unsigned int)frames_wr, out_buf.frameCount, (unsigned int)frames);
+            frames_wr = frames;
+        }
+    }
+
+    /* Remove aux_channels that have been added on top of main_channels
+     * Assumption is made that the channels are interleaved and that the main
+     * channels are first. */
+    if (has_aux_channels)
+    {
+        size_t src_channels = in->config.channels;
+        size_t dst_channels = popcount(in->main_channels);
+        int16_t* src_buffer = (int16_t *)proc_buf_out;
+        int16_t* dst_buffer = (int16_t *)buffer;
+
+        if (dst_channels == 1) {
+            for (i = frames_wr; i > 0; i--)
+            {
+                *dst_buffer++ = *src_buffer;
+                src_buffer += src_channels;
+            }
+        } else {
+            for (i = frames_wr; i > 0; i--)
+            {
+                memcpy(dst_buffer, src_buffer, dst_channels*sizeof(int16_t));
+                dst_buffer += dst_channels;
+                src_buffer += src_channels;
+            }
+        }
+    }
+
+    return frames_wr;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
+                       size_t bytes)
+{
+    int ret = 0;
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+    struct m0_audio_device *adev = in->dev;
+    size_t frames_rq = bytes / audio_stream_frame_size(&stream->common);
+
+    /* acquiring hw device mutex systematically is useful if a low priority thread is waiting
+     * on the input stream mutex - e.g. executing select_mode() while holding the hw device
+     * mutex
+     */
+    pthread_mutex_lock(&adev->lock);
+    pthread_mutex_lock(&in->lock);
+    if (in->standby) {
+        ret = start_input_stream(in);
+        if (ret == 0)
+            in->standby = 0;
+    }
+    pthread_mutex_unlock(&adev->lock);
+
+    if (ret < 0)
+        goto exit;
+
+    if (in->num_preprocessors != 0)
+        ret = process_frames(in, buffer, frames_rq);
+    else if (in->resampler != NULL)
+        ret = read_frames(in, buffer, frames_rq);
+    else
+        ret = pcm_read(in->pcm, buffer, bytes);
+
+    if (ret > 0)
+        ret = 0;
+
+    if (ret == 0 && adev->mic_mute)
+        memset(buffer, 0, bytes);
+
+exit:
+    if (ret < 0)
+        usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
+               in_get_sample_rate(&stream->common));
+
+    pthread_mutex_unlock(&in->lock);
+    return bytes;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+    return 0;
+}
+
+#define GET_COMMAND_STATUS(status, fct_status, cmd_status) \
+            do {                                           \
+                if (fct_status != 0)                       \
+                    status = fct_status;                   \
+                else if (cmd_status != 0)                  \
+                    status = cmd_status;                   \
+            } while(0)
+
+static int in_configure_reverse(struct m0_stream_in *in)
+{
+    int32_t cmd_status;
+    uint32_t size = sizeof(int);
+    effect_config_t config;
+    int32_t status = 0;
+    int32_t fct_status = 0;
+    int i;
+
+    if (in->num_preprocessors > 0) {
+        config.inputCfg.channels = in->main_channels;
+        config.outputCfg.channels = in->main_channels;
+        config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+        config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+        config.inputCfg.samplingRate = in->requested_rate;
+        config.outputCfg.samplingRate = in->requested_rate;
+        config.inputCfg.mask =
+                ( EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT );
+        config.outputCfg.mask =
+                ( EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT );
+
+        for (i = 0; i < in->num_preprocessors; i++)
+        {
+            if ((*in->preprocessors[i].effect_itfe)->process_reverse == NULL)
+                continue;
+            fct_status = (*(in->preprocessors[i].effect_itfe))->command(
+                                                        in->preprocessors[i].effect_itfe,
+                                                        EFFECT_CMD_SET_CONFIG_REVERSE,
+                                                        sizeof(effect_config_t),
+                                                        &config,
+                                                        &size,
+                                                        &cmd_status);
+            GET_COMMAND_STATUS(status, fct_status, cmd_status);
+        }
+    }
+    return status;
+}
+
+#define MAX_NUM_CHANNEL_CONFIGS 10
+
+static void in_read_audio_effect_channel_configs(struct m0_stream_in *in,
+                                                 struct effect_info_s *effect_info)
+{
+    /* size and format of the cmd are defined in hardware/audio_effect.h */
+    effect_handle_t effect = effect_info->effect_itfe;
+    uint32_t cmd_size = 2 * sizeof(uint32_t);
+    uint32_t cmd[] = { EFFECT_FEATURE_AUX_CHANNELS, MAX_NUM_CHANNEL_CONFIGS };
+    /* reply = status + number of configs (n) + n x channel_config_t */
+    uint32_t reply_size =
+            2 * sizeof(uint32_t) + (MAX_NUM_CHANNEL_CONFIGS * sizeof(channel_config_t));
+    int32_t reply[reply_size];
+    int32_t cmd_status;
+
+    ALOG_ASSERT((effect_info->num_channel_configs == 0),
+                "in_read_audio_effect_channel_configs() num_channel_configs not cleared");
+    ALOG_ASSERT((effect_info->channel_configs == NULL),
+                "in_read_audio_effect_channel_configs() channel_configs not cleared");
+
+    /* if this command is not supported, then the effect is supposed to return -EINVAL.
+     * This error will be interpreted as if the effect supports the main_channels but does not
+     * support any aux_channels */
+    cmd_status = (*effect)->command(effect,
+                                EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS,
+                                cmd_size,
+                                (void*)&cmd,
+                                &reply_size,
+                                (void*)&reply);
+
+    if (cmd_status != 0) {
+        ALOGI("%s: fx->command returned %d", __func__, cmd_status);
+        return;
+    }
+
+    if (reply[0] != 0) {
+        ALOGW("%s: "
+                "command EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS error %d num configs %d",
+                __func__, reply[0], (reply[0] == -ENOMEM) ? reply[1] : MAX_NUM_CHANNEL_CONFIGS);
+        return;
+    }
+
+    /* the feature is not supported */
+    ALOGI("in_read_audio_effect_channel_configs()(): "
+            "Feature supported and adding %d channel configs to the list", reply[1]);
+    effect_info->num_channel_configs = reply[1];
+    effect_info->channel_configs =
+            (channel_config_t *) malloc(sizeof(channel_config_t) * reply[1]); /* n x configs */
+    memcpy(effect_info->channel_configs, (reply + 2), sizeof(channel_config_t) * reply[1]);
+}
+
+
+static uint32_t in_get_aux_channels(struct m0_stream_in *in)
+{
+    int i;
+    channel_config_t new_chcfg = {0, 0};
+
+    if (in->num_preprocessors == 0)
+        return 0;
+
+    /* do not enable dual mic configurations when capturing from other microphones than
+     * main or sub */
+    if (!(in->device & (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC)))
+        return 0;
+
+    /* retain most complex aux channels configuration compatible with requested main channels and
+     * supported by audio driver and all pre processors */
+    for (i = 0; i < NUM_IN_AUX_CNL_CONFIGS; i++) {
+        channel_config_t *cur_chcfg = &in_aux_cnl_configs[i];
+        if (cur_chcfg->main_channels == in->main_channels) {
+            size_t match_cnt;
+            size_t idx_preproc;
+            for (idx_preproc = 0, match_cnt = 0;
+                 /* no need to continue if at least one preprocessor doesn't match */
+                 idx_preproc < (size_t)in->num_preprocessors && match_cnt == idx_preproc;
+                 idx_preproc++) {
+                struct effect_info_s *effect_info = &in->preprocessors[idx_preproc];
+                size_t idx_chcfg;
+
+                for (idx_chcfg = 0; idx_chcfg < effect_info->num_channel_configs; idx_chcfg++) {
+                    if (memcmp(effect_info->channel_configs + idx_chcfg,
+                               cur_chcfg,
+                               sizeof(channel_config_t)) == 0) {
+                        match_cnt++;
+                        break;
+                    }
+                }
+            }
+            /* if all preprocessors match, we have a candidate */
+            if (match_cnt == (size_t)in->num_preprocessors) {
+                /* retain most complex aux channels configuration */
+                if (popcount(cur_chcfg->aux_channels) > popcount(new_chcfg.aux_channels)) {
+                    new_chcfg = *cur_chcfg;
+                }
+            }
+        }
+    }
+
+    ALOGI("in_get_aux_channels(): return %04x", new_chcfg.aux_channels);
+
+    return new_chcfg.aux_channels;
+}
+
+static int in_configure_effect_channels(effect_handle_t effect,
+                                        channel_config_t *channel_config)
+{
+    int status = 0;
+    int fct_status;
+    int32_t cmd_status;
+    uint32_t reply_size;
+    effect_config_t config;
+    uint32_t cmd[(sizeof(uint32_t) + sizeof(channel_config_t) - 1) / sizeof(uint32_t) + 1];
+
+    ALOGI("in_configure_effect_channels(): configure effect with channels: [%04x][%04x]",
+            channel_config->main_channels,
+            channel_config->aux_channels);
+
+    config.inputCfg.mask = EFFECT_CONFIG_CHANNELS;
+    config.outputCfg.mask = EFFECT_CONFIG_CHANNELS;
+    reply_size = sizeof(effect_config_t);
+    fct_status = (*effect)->command(effect,
+                                EFFECT_CMD_GET_CONFIG,
+                                0,
+                                NULL,
+                                &reply_size,
+                                &config);
+    if (fct_status != 0) {
+        ALOGE("in_configure_effect_channels(): EFFECT_CMD_GET_CONFIG failed");
+        return fct_status;
+    }
+
+    config.inputCfg.channels = channel_config->main_channels | channel_config->aux_channels;
+    config.outputCfg.channels = config.inputCfg.channels;
+    reply_size = sizeof(uint32_t);
+    fct_status = (*effect)->command(effect,
+                                    EFFECT_CMD_SET_CONFIG,
+                                    sizeof(effect_config_t),
+                                    &config,
+                                    &reply_size,
+                                    &cmd_status);
+    GET_COMMAND_STATUS(status, fct_status, cmd_status);
+
+    cmd[0] = EFFECT_FEATURE_AUX_CHANNELS;
+    memcpy(cmd + 1, channel_config, sizeof(channel_config_t));
+    reply_size = sizeof(uint32_t);
+    fct_status = (*effect)->command(effect,
+                                EFFECT_CMD_SET_FEATURE_CONFIG,
+                                sizeof(cmd), //sizeof(uint32_t) + sizeof(channel_config_t),
+                                cmd,
+                                &reply_size,
+                                &cmd_status);
+    GET_COMMAND_STATUS(status, fct_status, cmd_status);
+
+    /* some implementations need to be re-enabled after a config change */
+    reply_size = sizeof(uint32_t);
+    fct_status = (*effect)->command(effect,
+                                  EFFECT_CMD_ENABLE,
+                                  0,
+                                  NULL,
+                                  &reply_size,
+                                  &cmd_status);
+    GET_COMMAND_STATUS(status, fct_status, cmd_status);
+
+    return status;
+}
+
+static int in_reconfigure_channels(struct m0_stream_in *in,
+                                   effect_handle_t effect,
+                                   channel_config_t *channel_config,
+                                   bool config_changed) {
+
+    int status = 0;
+
+    ALOGI("%s: config_changed %d effect %p",
+          __func__, config_changed, effect);
+
+    /* if config changed, reconfigure all previously added effects */
+    if (config_changed) {
+        int i;
+        for (i = 0; i < in->num_preprocessors; i++)
+        {
+            int cur_status = in_configure_effect_channels(in->preprocessors[i].effect_itfe,
+                                                  channel_config);
+            if (cur_status != 0) {
+                ALOGI("%s: error %d configuring effect "
+                        "%d with channels: [%04x][%04x]",
+                        __func__,
+                        cur_status,
+                        i,
+                        channel_config->main_channels,
+                        channel_config->aux_channels);
+                status = cur_status;
+            }
+        }
+    } else if (effect != NULL && channel_config->aux_channels) {
+        /* if aux channels config did not change but aux channels are present,
+         * we still need to configure the effect being added */
+        status = in_configure_effect_channels(effect, channel_config);
+    }
+    return status;
+}
+
+static void in_update_aux_channels(struct m0_stream_in *in,
+                                   effect_handle_t effect)
+{
+    uint32_t aux_channels;
+    channel_config_t channel_config;
+    int status;
+
+    aux_channels = in_get_aux_channels(in);
+
+    channel_config.main_channels = in->main_channels;
+    channel_config.aux_channels = aux_channels;
+    status = in_reconfigure_channels(in,
+                                     effect,
+                                     &channel_config,
+                                     (aux_channels != in->aux_channels));
+
+    if (status != 0) {
+        ALOGI("%s: in_reconfigure_channels error %d", __func__, status);
+        /* resetting aux channels configuration */
+        aux_channels = 0;
+        channel_config.aux_channels = 0;
+        in_reconfigure_channels(in, effect, &channel_config, true);
+    }
+    if (in->aux_channels != aux_channels) {
+        in->aux_channels_changed = true;
+        in->aux_channels = aux_channels;
+        do_input_standby(in);
+    }
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream,
+                               effect_handle_t effect)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+    int status;
+    effect_descriptor_t desc;
+
+    pthread_mutex_lock(&in->dev->lock);
+    pthread_mutex_lock(&in->lock);
+    if (in->num_preprocessors >= MAX_PREPROCESSORS) {
+        status = -ENOSYS;
+        goto exit;
+    }
+
+    status = (*effect)->get_descriptor(effect, &desc);
+    if (status != 0)
+        goto exit;
+
+    in->preprocessors[in->num_preprocessors].effect_itfe = effect;
+    /* add the supported channel of the effect in the channel_configs */
+    in_read_audio_effect_channel_configs(in, &in->preprocessors[in->num_preprocessors]);
+
+    in->num_preprocessors++;
+
+    /* check compatibility between main channel supported and possible auxiliary channels */
+    in_update_aux_channels(in, effect);
+
+    ALOGV("%s: effect type: %08x", __func__, desc.type.timeLow);
+
+    if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) {
+        in->need_echo_reference = true;
+        do_input_standby(in);
+        in_configure_reverse(in);
+    }
+
+exit:
+
+    ALOGW_IF(status != 0, "%s: error %d", __func__, status);
+    pthread_mutex_unlock(&in->lock);
+    pthread_mutex_unlock(&in->dev->lock);
+    return status;
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream,
+                                  effect_handle_t effect)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+    int i;
+    int status = -EINVAL;
+    effect_descriptor_t desc;
+
+    pthread_mutex_lock(&in->dev->lock);
+    pthread_mutex_lock(&in->lock);
+    if (in->num_preprocessors <= 0) {
+        status = -ENOSYS;
+        goto exit;
+    }
+
+    for (i = 0; i < in->num_preprocessors; i++) {
+        if (status == 0) { /* status == 0 means an effect was removed from a previous slot */
+            in->preprocessors[i - 1].effect_itfe = in->preprocessors[i].effect_itfe;
+            in->preprocessors[i - 1].channel_configs = in->preprocessors[i].channel_configs;
+            in->preprocessors[i - 1].num_channel_configs = in->preprocessors[i].num_channel_configs;
+            ALOGI("in_remove_audio_effect moving fx from %d to %d", i, i - 1);
+            continue;
+        }
+        if (in->preprocessors[i].effect_itfe == effect) {
+            ALOGI("in_remove_audio_effect found fx at index %d", i);
+            free(in->preprocessors[i].channel_configs);
+            status = 0;
+        }
+    }
+
+    if (status != 0)
+        goto exit;
+
+    in->num_preprocessors--;
+    /* if we remove one effect, at least the last preproc should be reset */
+    in->preprocessors[in->num_preprocessors].num_channel_configs = 0;
+    in->preprocessors[in->num_preprocessors].effect_itfe = NULL;
+    in->preprocessors[in->num_preprocessors].channel_configs = NULL;
+
+
+    /* check compatibility between main channel supported and possible auxiliary channels */
+    in_update_aux_channels(in, NULL);
+
+    status = (*effect)->get_descriptor(effect, &desc);
+    if (status != 0)
+        goto exit;
+
+    ALOGI("%s: effect type: %08x", __func__, desc.type.timeLow);
+
+    if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) {
+        in->need_echo_reference = false;
+        do_input_standby(in);
+    }
+
+exit:
+
+    ALOGW_IF(status != 0, "%s: error %d", __func__, status);
+    pthread_mutex_unlock(&in->lock);
+    pthread_mutex_unlock(&in->dev->lock);
+    return status;
+}
+
+static int adev_open_output_stream(struct audio_hw_device *dev,
+                                   audio_io_handle_t handle,
+                                   audio_devices_t devices,
+                                   audio_output_flags_t flags,
+                                   struct audio_config *config,
+                                   struct audio_stream_out **stream_out)
+{
+    struct m0_audio_device *ladev = (struct m0_audio_device *)dev;
+    struct m0_stream_out *out;
+    int ret;
+    int output_type;
+    *stream_out = NULL;
+
+    out = (struct m0_stream_out *)calloc(1, sizeof(struct m0_stream_out));
+    if (!out)
+        return -ENOMEM;
+
+    out->sup_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
+    out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+
+    if (ladev->outputs[OUTPUT_DEEP_BUF] != NULL) {
+        ret = -ENOSYS;
+        goto err_open;
+    }
+    output_type = OUTPUT_DEEP_BUF;
+    out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    out->stream.common.get_buffer_size = out_get_buffer_size_deep_buffer;
+    out->stream.common.get_sample_rate = out_get_sample_rate;
+    out->stream.get_latency = out_get_latency_deep_buffer;
+    out->stream.write = out_write_deep_buffer;
+
+    ret = create_resampler(DEFAULT_OUT_SAMPLING_RATE,
+                           MM_FULL_POWER_SAMPLING_RATE,
+                           2,
+                           RESAMPLER_QUALITY_DEFAULT,
+                           NULL,
+                           &out->resampler);
+    if (ret != 0)
+        goto err_open;
+
+    out->stream.common.set_sample_rate = out_set_sample_rate;
+    out->stream.common.get_channels = out_get_channels;
+    out->stream.common.get_format = out_get_format;
+    out->stream.common.set_format = out_set_format;
+    out->stream.common.standby = out_standby;
+    out->stream.common.dump = out_dump;
+    out->stream.common.set_parameters = out_set_parameters;
+    out->stream.common.get_parameters = out_get_parameters;
+    out->stream.common.add_audio_effect = out_add_audio_effect;
+    out->stream.common.remove_audio_effect = out_remove_audio_effect;
+    out->stream.set_volume = out_set_volume;
+    out->stream.get_render_position = out_get_render_position;
+
+    out->dev = ladev;
+    out->standby = 1;
+
+    /* FIXME: when we support multiple output devices, we will want to
+     * do the following:
+     * adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
+     * adev->devices |= out->device;
+     * select_output_device(adev);
+     * This is because out_set_parameters() with a route is not
+     * guaranteed to be called after an output stream is opened. */
+
+    config->format = out->stream.common.get_format(&out->stream.common);
+    config->channel_mask = out->stream.common.get_channels(&out->stream.common);
+    config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common);
+
+    *stream_out = &out->stream;
+    ladev->outputs[output_type] = out;
+
+    return 0;
+
+err_open:
+    free(out);
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct m0_audio_device *ladev = (struct m0_audio_device *)dev;
+    struct m0_stream_out *out = (struct m0_stream_out *)stream;
+    int i;
+
+    out_standby(&stream->common);
+    for (i = 0; i < OUTPUT_TOTAL; i++) {
+        if (ladev->outputs[i] == out) {
+            ladev->outputs[i] = NULL;
+            break;
+        }
+    }
+
+    if (out->buffer)
+        free(out->buffer);
+    if (out->resampler)
+        release_resampler(out->resampler);
+    free(stream);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    struct m0_audio_device *adev = (struct m0_audio_device *)dev;
+    struct str_parms *parms;
+    char *str;
+    char value[32];
+    int ret;
+
+    parms = str_parms_create_str(kvpairs);
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_TTY_MODE, value, sizeof(value));
+    if (ret >= 0) {
+        int tty_mode;
+
+        if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_OFF) == 0)
+            tty_mode = TTY_MODE_OFF;
+        else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_VCO) == 0)
+            tty_mode = TTY_MODE_VCO;
+        else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_HCO) == 0)
+            tty_mode = TTY_MODE_HCO;
+        else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_FULL) == 0)
+            tty_mode = TTY_MODE_FULL;
+        else
+            return -EINVAL;
+
+        pthread_mutex_lock(&adev->lock);
+        if (tty_mode != adev->tty_mode) {
+            adev->tty_mode = tty_mode;
+            if (adev->mode == AUDIO_MODE_IN_CALL)
+                select_output_device(adev);
+        }
+        pthread_mutex_unlock(&adev->lock);
+    }
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value));
+    if (ret >= 0) {
+        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
+            adev->bluetooth_nrec = true;
+        else
+            adev->bluetooth_nrec = false;
+    }
+
+    ret = str_parms_get_str(parms, "screen_off", value, sizeof(value));
+    if (ret >= 0) {
+        if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
+            adev->screen_off = false;
+        else
+            adev->screen_off = true;
+    }
+
+    str_parms_destroy(parms);
+    return ret;
+}
+
+static char * adev_get_parameters(const struct audio_hw_device *dev,
+                                  const char *keys)
+{
+    return strdup("");
+}
+
+static int adev_init_check(const struct audio_hw_device *dev)
+{
+    return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    struct m0_audio_device *adev = (struct m0_audio_device *)dev;
+
+    adev->voice_volume = volume;
+
+    if (adev->mode == AUDIO_MODE_IN_CALL)
+        ril_set_call_volume(&adev->ril, SOUND_TYPE_VOICE, volume);
+
+    return 0;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
+{
+    struct m0_audio_device *adev = (struct m0_audio_device *)dev;
+
+    pthread_mutex_lock(&adev->lock);
+    if (adev->mode != mode) {
+        adev->mode = mode;
+        select_mode(adev);
+    }
+    pthread_mutex_unlock(&adev->lock);
+
+    return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    struct m0_audio_device *adev = (struct m0_audio_device *)dev;
+
+    adev->mic_mute = state;
+
+    return 0;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    struct m0_audio_device *adev = (struct m0_audio_device *)dev;
+
+    *state = adev->mic_mute;
+
+    return 0;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         const struct audio_config *config)
+{
+    size_t size;
+    int channel_count = popcount(config->channel_mask);
+    if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
+        return 0;
+
+    return get_input_buffer_size(config->sample_rate, config->format, channel_count);
+}
+
+static int adev_open_input_stream(struct audio_hw_device *dev,
+                                  audio_io_handle_t handle,
+                                  audio_devices_t devices,
+                                  struct audio_config *config,
+                                  struct audio_stream_in **stream_in)
+{
+    struct m0_audio_device *ladev = (struct m0_audio_device *)dev;
+    struct m0_stream_in *in;
+    int ret;
+    int channel_count = popcount(config->channel_mask);
+
+    *stream_in = NULL;
+
+    if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
+        return -EINVAL;
+
+    in = (struct m0_stream_in *)calloc(1, sizeof(struct m0_stream_in));
+    if (!in)
+        return -ENOMEM;
+
+    in->stream.common.get_sample_rate = in_get_sample_rate;
+    in->stream.common.set_sample_rate = in_set_sample_rate;
+    in->stream.common.get_buffer_size = in_get_buffer_size;
+    in->stream.common.get_channels = in_get_channels;
+    in->stream.common.get_format = in_get_format;
+    in->stream.common.set_format = in_set_format;
+    in->stream.common.standby = in_standby;
+    in->stream.common.dump = in_dump;
+    in->stream.common.set_parameters = in_set_parameters;
+    in->stream.common.get_parameters = in_get_parameters;
+    in->stream.common.add_audio_effect = in_add_audio_effect;
+    in->stream.common.remove_audio_effect = in_remove_audio_effect;
+    in->stream.set_gain = in_set_gain;
+    in->stream.read = in_read;
+    in->stream.get_input_frames_lost = in_get_input_frames_lost;
+
+    in->requested_rate = config->sample_rate;
+
+    memcpy(&in->config, &pcm_config_capture, sizeof(pcm_config_capture));
+    in->config.channels = channel_count;
+
+    in->main_channels = config->channel_mask;
+
+    /* initialisation of preprocessor structure array is implicit with the calloc.
+     * same for in->aux_channels and in->aux_channels_changed */
+
+    if (in->requested_rate != in->config.rate) {
+        in->buf_provider.get_next_buffer = get_next_buffer;
+        in->buf_provider.release_buffer = release_buffer;
+
+        ret = create_resampler(in->config.rate,
+                               in->requested_rate,
+                               in->config.channels,
+                               RESAMPLER_QUALITY_DEFAULT,
+                               &in->buf_provider,
+                               &in->resampler);
+        if (ret != 0) {
+            ret = -EINVAL;
+            goto err;
+        }
+    }
+
+    in->dev = ladev;
+    in->standby = 1;
+    in->device = devices;
+
+    *stream_in = &in->stream;
+    return 0;
+
+err:
+    if (in->resampler)
+        release_resampler(in->resampler);
+
+    free(in);
+    return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                                   struct audio_stream_in *stream)
+{
+    struct m0_stream_in *in = (struct m0_stream_in *)stream;
+    int i;
+
+    in_standby(&stream->common);
+
+    for (i = 0; i < in->num_preprocessors; i++) {
+        free(in->preprocessors[i].channel_configs);
+    }
+
+    free(in->read_buf);
+    if (in->resampler) {
+        release_resampler(in->resampler);
+    }
+    if (in->proc_buf_in)
+        free(in->proc_buf_in);
+    if (in->proc_buf_out)
+        free(in->proc_buf_out);
+    if (in->ref_buf)
+        free(in->ref_buf);
+
+    free(stream);
+    return;
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+    return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+    struct m0_audio_device *adev = (struct m0_audio_device *)device;
+
+    /* RIL */
+    ril_close(&adev->ril);
+
+    mixer_close(adev->mixer);
+    free(device);
+    return 0;
+}
+
+static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev)
+{
+    return (/* OUT */
+            AUDIO_DEVICE_OUT_EARPIECE |
+            AUDIO_DEVICE_OUT_SPEAKER |
+            AUDIO_DEVICE_OUT_WIRED_HEADSET |
+            AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
+            AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
+            AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET |
+            AUDIO_DEVICE_OUT_ALL_SCO |
+            AUDIO_DEVICE_OUT_DEFAULT |
+            /* IN */
+            AUDIO_DEVICE_IN_BUILTIN_MIC |
+            AUDIO_DEVICE_IN_WIRED_HEADSET |
+            AUDIO_DEVICE_IN_ALL_SCO |
+            AUDIO_DEVICE_IN_DEFAULT);
+}
+
+struct config_parse_state {
+    struct m0_audio_device *adev;
+    struct m0_dev_cfg *dev;
+    bool on;
+
+    struct route_setting *path;
+    unsigned int path_len;
+};
+
+static const struct {
+    int mask;
+    const char *name;
+} dev_names[] = {
+    { AUDIO_DEVICE_OUT_SPEAKER, "speaker" },
+    { AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "headphone" },
+    { AUDIO_DEVICE_OUT_EARPIECE, "earpiece" },
+    { AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET, "analog-dock" },
+    { AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, "digital-dock" },
+    { AUDIO_DEVICE_OUT_ALL_SCO, "sco-out" },
+
+    { AUDIO_DEVICE_IN_BUILTIN_MIC, "builtin-mic" },
+    { AUDIO_DEVICE_IN_BACK_MIC, "back-mic" },
+    { AUDIO_DEVICE_IN_WIRED_HEADSET, "headset-in" },
+    { AUDIO_DEVICE_IN_ALL_SCO, "sco-in" },
+};
+
+static void adev_config_start(void *data, const XML_Char *elem,
+                  const XML_Char **attr)
+{
+    struct config_parse_state *s = data;
+    struct m0_dev_cfg *dev_cfg;
+    const XML_Char *name = NULL;
+    const XML_Char *val = NULL;
+    unsigned int i, j;
+
+    for (i = 0; attr[i]; i += 2) {
+    if (strcmp(attr[i], "name") == 0)
+        name = attr[i + 1];
+
+    if (strcmp(attr[i], "val") == 0)
+        val = attr[i + 1];
+    }
+
+    if (strcmp(elem, "device") == 0) {
+    if (!name) {
+        ALOGE("Unnamed device\n");
+        return;
+    }
+
+    for (i = 0; i < sizeof(dev_names) / sizeof(dev_names[0]); i++) {
+        if (strcmp(dev_names[i].name, name) == 0) {
+        ALOGI("Allocating device %s\n", name);
+        dev_cfg = realloc(s->adev->dev_cfgs,
+                  (s->adev->num_dev_cfgs + 1)
+                  * sizeof(*dev_cfg));
+        if (!dev_cfg) {
+            ALOGE("Unable to allocate dev_cfg\n");
+            return;
+        }
+
+        s->dev = &dev_cfg[s->adev->num_dev_cfgs];
+        memset(s->dev, 0, sizeof(*s->dev));
+        s->dev->mask = dev_names[i].mask;
+
+        s->adev->dev_cfgs = dev_cfg;
+        s->adev->num_dev_cfgs++;
+        }
+    }
+
+    } else if (strcmp(elem, "path") == 0) {
+    if (s->path_len)
+        ALOGW("Nested paths\n");
+
+    /* If this a path for a device it must have a role */
+    if (s->dev) {
+        /* Need to refactor a bit... */
+        if (strcmp(name, "on") == 0) {
+        s->on = true;
+        } else if (strcmp(name, "off") == 0) {
+        s->on = false;
+        } else {
+        ALOGW("Unknown path name %s\n", name);
+        }
+    }
+
+    } else if (strcmp(elem, "ctl") == 0) {
+    struct route_setting *r;
+
+    if (!name) {
+        ALOGE("Unnamed control\n");
+        return;
+    }
+
+    if (!val) {
+        ALOGE("No value specified for %s\n", name);
+        return;
+    }
+
+    ALOGV("Parsing control %s => %s\n", name, val);
+
+    r = realloc(s->path, sizeof(*r) * (s->path_len + 1));
+    if (!r) {
+        ALOGE("Out of memory handling %s => %s\n", name, val);
+        return;
+    }
+
+    r[s->path_len].ctl_name = strdup(name);
+    r[s->path_len].strval = NULL;
+
+    /* This can be fooled but it'll do */
+    r[s->path_len].intval = atoi(val);
+    if (!r[s->path_len].intval && strcmp(val, "0") != 0)
+        r[s->path_len].strval = strdup(val);
+
+    s->path = r;
+    s->path_len++;
+    }
+}
+
+static void adev_config_end(void *data, const XML_Char *name)
+{
+    struct config_parse_state *s = data;
+    unsigned int i;
+
+    if (strcmp(name, "path") == 0) {
+    if (!s->path_len)
+        ALOGW("Empty path\n");
+
+    if (!s->dev) {
+        ALOGV("Applying %d element default route\n", s->path_len);
+
+        set_route_by_array(s->adev->mixer, s->path, s->path_len);
+
+        for (i = 0; i < s->path_len; i++) {
+        free(s->path[i].ctl_name);
+        free(s->path[i].strval);
+        }
+
+        free(s->path);
+
+        /* Refactor! */
+    } else if (s->on) {
+        ALOGV("%d element on sequence\n", s->path_len);
+        s->dev->on = s->path;
+        s->dev->on_len = s->path_len;
+
+    } else {
+        ALOGV("%d element off sequence\n", s->path_len);
+
+        /* Apply it, we'll reenable anything that's wanted later */
+        set_route_by_array(s->adev->mixer, s->path, s->path_len);
+
+        s->dev->off = s->path;
+        s->dev->off_len = s->path_len;
+    }
+
+    s->path_len = 0;
+    s->path = NULL;
+
+    } else if (strcmp(name, "device") == 0) {
+    s->dev = NULL;
+    }
+}
+
+static int adev_config_parse(struct m0_audio_device *adev)
+{
+    struct config_parse_state s;
+    FILE *f;
+    XML_Parser p;
+    char property[PROPERTY_VALUE_MAX];
+    char file[80];
+    int ret = 0;
+    bool eof = false;
+    int len;
+
+    property_get("ro.product.device", property, "tiny_hw");
+    snprintf(file, sizeof(file), "/system/etc/sound/%s", property);
+
+    ALOGV("Reading configuration from %s\n", file);
+    f = fopen(file, "r");
+    if (!f) {
+    ALOGE("Failed to open %s\n", file);
+    return -ENODEV;
+    }
+
+    p = XML_ParserCreate(NULL);
+    if (!p) {
+    ALOGE("Failed to create XML parser\n");
+    ret = -ENOMEM;
+    goto out;
+    }
+
+    memset(&s, 0, sizeof(s));
+    s.adev = adev;
+    XML_SetUserData(p, &s);
+
+    XML_SetElementHandler(p, adev_config_start, adev_config_end);
+
+    while (!eof) {
+    len = fread(file, 1, sizeof(file), f);
+    if (ferror(f)) {
+        ALOGE("I/O error reading config\n");
+        ret = -EIO;
+        goto out_parser;
+    }
+    eof = feof(f);
+
+    if (XML_Parse(p, file, len, eof) == XML_STATUS_ERROR) {
+        ALOGE("Parse error at line %u:\n%s\n",
+         (unsigned int)XML_GetCurrentLineNumber(p),
+         XML_ErrorString(XML_GetErrorCode(p)));
+        ret = -EINVAL;
+        goto out_parser;
+    }
+    }
+
+ out_parser:
+    XML_ParserFree(p);
+ out:
+    fclose(f);
+
+    return ret;
+}
+
+static int adev_open(const hw_module_t* module, const char* name,
+                     hw_device_t** device)
+{
+    struct m0_audio_device *adev;
+    int ret;
+
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    adev = calloc(1, sizeof(struct m0_audio_device));
+    if (!adev)
+        return -ENOMEM;
+
+    adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
+    adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_1_0;
+    adev->hw_device.common.module = (struct hw_module_t *) module;
+    adev->hw_device.common.close = adev_close;
+
+    adev->hw_device.get_supported_devices = adev_get_supported_devices;
+    adev->hw_device.init_check = adev_init_check;
+    adev->hw_device.set_voice_volume = adev_set_voice_volume;
+    adev->hw_device.set_master_volume = adev_set_master_volume;
+    adev->hw_device.set_mode = adev_set_mode;
+    adev->hw_device.set_mic_mute = adev_set_mic_mute;
+    adev->hw_device.get_mic_mute = adev_get_mic_mute;
+    adev->hw_device.set_parameters = adev_set_parameters;
+    adev->hw_device.get_parameters = adev_get_parameters;
+    adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
+    adev->hw_device.open_output_stream = adev_open_output_stream;
+    adev->hw_device.close_output_stream = adev_close_output_stream;
+    adev->hw_device.open_input_stream = adev_open_input_stream;
+    adev->hw_device.close_input_stream = adev_close_input_stream;
+    adev->hw_device.dump = adev_dump;
+
+    adev->mixer = mixer_open(CARD_DEFAULT);
+    if (!adev->mixer) {
+        free(adev);
+        ALOGE("Unable to open the mixer, aborting.");
+        return -EINVAL;
+    }
+
+    /* +30db boost for mics */
+    adev->mixer_ctls.mixinl_in1l_volume = mixer_get_ctl_by_name(adev->mixer, "MIXINL IN1L Volume");
+    adev->mixer_ctls.mixinl_in2l_volume = mixer_get_ctl_by_name(adev->mixer, "MIXINL IN2L Volume");
+
+    ret = adev_config_parse(adev);
+    if (ret != 0)
+        goto err_mixer;
+
+    /* Set the default route before the PCM stream is opened */
+    pthread_mutex_lock(&adev->lock);
+    adev->mode = AUDIO_MODE_NORMAL;
+    adev->devices = AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_IN_BUILTIN_MIC;
+    select_devices(adev);
+
+    adev->pcm_modem_dl = NULL;
+    adev->pcm_modem_ul = NULL;
+    adev->voice_volume = 1.0f;
+    adev->tty_mode = TTY_MODE_OFF;
+    adev->bluetooth_nrec = true;
+    adev->wb_amr = 0;
+
+    /* RIL */
+    ril_open(&adev->ril);
+    pthread_mutex_unlock(&adev->lock);
+    /* register callback for wideband AMR setting */
+    ril_register_set_wb_amr_callback(audio_set_wb_amr_callback, (void *)adev);
+
+    *device = &adev->hw_device.common;
+
+    return 0;
+
+err_mixer:
+    mixer_close(adev->mixer);
+err:
+    return -EINVAL;
+}
+
+static struct hw_module_methods_t hal_module_methods = {
+    .open = adev_open,
+};
+
+struct audio_module HAL_MODULE_INFO_SYM = {
+    .common = {
+        .tag = HARDWARE_MODULE_TAG,
+        .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
+        .hal_api_version = HARDWARE_HAL_API_VERSION,
+        .id = AUDIO_HARDWARE_MODULE_ID,
+        .name = "M0 audio HW HAL",
+        .author = "The CyanogenMod Project",
+        .methods = &hal_module_methods,
+    },
+};
diff --git a/audio/audio_hw.h b/audio/audio_hw.h
new file mode 100644
index 0000000..e7b9ace
--- /dev/null
+++ b/audio/audio_hw.h
@@ -0,0 +1,240 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ * Copyright (C) 2012 Wolfson Microelectronics plc
+ * Copyright (C) 2012 The CyanogenMod Project
+ *               Daniel Hillenbrand <codeworkx@cyanogenmod.com>
+ *               Guillaume "XpLoDWilD" Lesniak <xplodgui@gmail.com>
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/* ALSA cards for WM1811 */
+#define CARD_DEFAULT  0
+
+#define PORT_PLAYBACK 0
+#define PORT_MODEM    1
+#define PORT_BT       2
+#define PORT_CAPTURE  3
+
+#define PCM_WRITE pcm_write
+
+#define PLAYBACK_PERIOD_SIZE  880
+#define PLAYBACK_PERIOD_COUNT 8
+#define PLAYBACK_SHORT_PERIOD_COUNT 2
+
+#define CAPTURE_PERIOD_SIZE   1056
+#define CAPTURE_PERIOD_COUNT  2
+
+#define SHORT_PERIOD_SIZE 192
+
+//
+// deep buffer
+//
+/* screen on */
+#define DEEP_BUFFER_SHORT_PERIOD_SIZE 1056
+#define PLAYBACK_DEEP_BUFFER_SHORT_PERIOD_COUNT 4
+/* screen off */
+#define DEEP_BUFFER_LONG_PERIOD_SIZE 880
+#define PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT 8
+
+
+/* minimum sleep time in out_write() when write threshold is not reached */
+#define MIN_WRITE_SLEEP_US 5000
+
+#define RESAMPLER_BUFFER_FRAMES (PLAYBACK_PERIOD_SIZE * 2)
+#define RESAMPLER_BUFFER_SIZE (4 * RESAMPLER_BUFFER_FRAMES)
+
+#define DEFAULT_OUT_SAMPLING_RATE 44100
+#define MM_LOW_POWER_SAMPLING_RATE 44100
+#define MM_FULL_POWER_SAMPLING_RATE 44100
+#define DEFAULT_IN_SAMPLING_RATE 44100
+
+/* sampling rate when using VX port for narrow band */
+#define VX_NB_SAMPLING_RATE 8000
+/* sampling rate when using VX port for wide band */
+#define VX_WB_SAMPLING_RATE 16000
+
+/* product-specific defines */
+#define PRODUCT_DEVICE_PROPERTY "ro.product.device"
+#define PRODUCT_NAME_PROPERTY   "ro.product.name"
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+#define STRING_TO_ENUM(string) { #string, string }
+
+struct string_to_enum {
+    const char *name;
+    uint32_t value;
+};
+
+const struct string_to_enum out_channels_name_to_enum_table[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+enum pcm_type {
+    PCM_NORMAL = 0,
+    PCM_SPDIF,
+    PCM_HDMI,
+    PCM_TOTAL,
+};
+
+enum output_type {
+    OUTPUT_DEEP_BUF,      // deep PCM buffers output stream
+    OUTPUT_LOW_LATENCY,   // low latency output stream
+    OUTPUT_HDMI,
+    OUTPUT_TOTAL
+};
+
+enum tty_modes {
+    TTY_MODE_OFF,
+    TTY_MODE_VCO,
+    TTY_MODE_HCO,
+    TTY_MODE_FULL
+};
+
+struct mixer_ctls
+{
+    struct mixer_ctl *mixinl_in1l_volume;
+    struct mixer_ctl *mixinl_in2l_volume;
+};
+
+struct route_setting
+{
+    char *ctl_name;
+    int intval;
+    char *strval;
+};
+
+struct route_setting voicecall_default[] = {
+    { .ctl_name = "AIF2 Mode", .intval = 0, },
+    { .ctl_name = "MainMicBias Mode", .intval = 3, },
+    { .ctl_name = "SubMicBias Mode", .intval = 3, },
+    { .ctl_name = "DAC1L Mixer AIF1.1 Switch", .intval = 1, },
+    { .ctl_name = "DAC1R Mixer AIF1.1 Switch", .intval = 1, },
+    { .ctl_name = "DAC1L Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = "DAC1R Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = "AIF2DAC Mux", .strval = "AIF2DACDAT", },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting voicecall_default_disable[] = {
+    { .ctl_name = "AIF2 Mode", .intval = 0, },
+    { .ctl_name = "DAC1L Mixer AIF2 Switch", .intval = 0, },
+    { .ctl_name = "DAC1R Mixer AIF2 Switch", .intval = 0, },
+    { .ctl_name = "AIF2DAC Mux", .strval = "AIF3DACDAT", },
+    { .ctl_name = "MainMicBias Mode", .intval = 2, },
+    { .ctl_name = "SubMicBias Mode", .intval = 2, },
+    { .ctl_name = "Main Mic Switch", .intval = 0, },
+    { .ctl_name = "MIXINL IN1L Switch", .intval = 0, },
+    { .ctl_name = "Sub Mic Switch", .intval = 0, },
+    { .ctl_name = "MIXINR IN1R Switch", .intval = 0, },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting default_input[] = {
+    { .ctl_name = "Main Mic Switch", .intval = 1, },
+    { .ctl_name = "MainMicBias Mode", .intval = 1, },
+    { .ctl_name = "IN2L Volume", .intval = 30, },
+    { .ctl_name = "MIXINL IN2L Switch", .intval = 1, },
+    { .ctl_name = "MIXINL IN2L Volume", .intval = 1, },
+    { .ctl_name = "AIF1ADC1 HPF Mode", .intval = 0, },
+    { .ctl_name = "AIF1ADC1 HPF Switch", .intval = 1, },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting default_input_disable[] = {
+    { .ctl_name = "Main Mic Switch", .intval = 0, },
+    { .ctl_name = "MainMicBias Mode", .intval = 0, },
+    { .ctl_name = "IN2L Volume", .intval = 22, },
+    { .ctl_name = "MIXINL IN2L Switch", .intval = 0, },
+    { .ctl_name = "AIF1ADC1 HPF Switch", .intval = 0, },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting headset_input[] = {
+    { .ctl_name = "Headset Mic Switch", .intval = 1, },
+    { .ctl_name = "IN2L Volume", .intval = 18, },
+    { .ctl_name = "MIXINL IN2L Switch", .intval = 1, },
+    { .ctl_name = "MIXINL IN2L Volume", .intval = 0, },
+    { .ctl_name = "AIF1ADC1 HPF Mode", .intval = 1, },
+    { .ctl_name = "AIF1ADC1 HPF Switch", .intval = 1, },
+    { .ctl_name = "AIF1ADC1 Volume", .intval = 96, },
+    { .ctl_name = "AIF1ADCL Source", .intval = 0, },
+    { .ctl_name = "AIF1ADCR Source", .intval = 0, },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting headset_input_disable[] = {
+    { .ctl_name = "Headset Mic Switch", .intval = 0, },
+    { .ctl_name = "MIXINL IN1L Switch", .intval = 0, },
+    { .ctl_name = "AIF1ADC1 HPF Mode", .intval = 0, },
+    { .ctl_name = "AIF1ADC1 HPF Switch", .intval = 0, },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting bt_output[] = {
+    { .ctl_name = "DAC1L Mixer AIF1.1 Switch", .intval = 1, },
+    { .ctl_name = "DAC1R Mixer AIF1.1 Switch", .intval = 1, },
+    { .ctl_name = "AIF3ADC Mux", .intval = 1, },
+    { .ctl_name = "AIF2DAC2L Mixer AIF1.1 Switch", .intval = 1, },
+    { .ctl_name = "AIF2DAC2R Mixer AIF1.1 Switch", .intval = 1, },
+    { .ctl_name = "AIF2DAC Volume", .intval = 96, },
+    { .ctl_name = "DAC2 Volume", .intval = 96, },
+    { .ctl_name = "AIF2ADC Volume", .intval = 96, },
+    { .ctl_name = "Speaker Mixer Volume", .intval = 1, },
+    { .ctl_name = "MIXINL IN2L Volume", .intval = 1, },
+    { .ctl_name = "IN1L Volume", .intval = 25, },
+    { .ctl_name = "IN1R Volume", .intval = 25, },
+    { .ctl_name = "Speaker Boost Volume", .intval = 4, },
+    { .ctl_name = "LINEOUT1N Switch", .intval = 0, },
+    { .ctl_name = "LINEOUT1P Switch", .intval = 0, },
+    { .ctl_name = "AIF2DACR Source", .intval = 0, },
+    { .ctl_name = "AIF1ADC1 HPF Switch", .intval = 0, },
+    { .ctl_name = "AIF2ADC HPF Mode", .intval = 1, },
+    { .ctl_name = "AIF2ADC HPF Switch", .intval = 1, },
+    { .ctl_name = "AIF2DAC Mux", .strval = "AIF2DACDAT", },
+    { .ctl_name = "AIF2DAC2R Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = "AIF2DAC2L Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting bt_input[] = {
+    { .ctl_name = "AIF2ADC Mux", .intval = 1, },
+    { .ctl_name = "AIF1ADCL Source", .intval = 0, },
+    { .ctl_name = "AIF1ADCR Source", .intval = 1, },
+    { .ctl_name = "AIF2ADCL Source", .intval = 0, },
+    { .ctl_name = "AIF2ADCR Source", .intval = 1, },
+    { .ctl_name = "DAC1L Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = "DAC1R Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = "AIF1ADC1R Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = "AIF1ADC1L Mixer AIF2 Switch", .intval = 1, },
+    { .ctl_name = "AIF1ADC1 Volume", .intval = 96, },
+    { .ctl_name = "AIF2DAC Volume", .intval = 96, },
+    { .ctl_name = NULL, },
+};
+
+struct route_setting bt_disable[] = {
+    { .ctl_name = "AIF2ADC Mux", .intval = 0, },
+    { .ctl_name = "MIXINL IN2L Volume", .intval = 0, },
+    { .ctl_name = "LINEOUT1N Switch", .intval = 1, },
+    { .ctl_name = "LINEOUT1P Switch", .intval = 1, },
+    { .ctl_name = "AIF2ADC HPF Mode", .intval = 0, },
+    { .ctl_name = "AIF2ADC HPF Switch", .intval = 0, },
+    { .ctl_name = "AIF2DAC2R Mixer AIF2 Switch", .intval = 0, },
+    { .ctl_name = "AIF2DAC2L Mixer AIF2 Switch", .intval = 0, },
+    { .ctl_name = "AIF1ADC1R Mixer AIF2 Switch", .intval = 0, },
+    { .ctl_name = "AIF1ADC1L Mixer AIF2 Switch", .intval = 0, },
+    { .ctl_name = NULL, },
+};
diff --git a/audio/ril_interface.c b/audio/ril_interface.c
new file mode 100755
index 0000000..89a0aef
--- /dev/null
+++ b/audio/ril_interface.c
@@ -0,0 +1,183 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define ALOG_TAG "audio_hw_primary"
+/*#define ALOG_NDEBUG 0*/
+
+#include <dlfcn.h>
+#include <stdlib.h>
+
+#include <utils/Log.h>
+#include <cutils/properties.h>
+
+#include "ril_interface.h"
+
+#define VOLUME_STEPS_DEFAULT  "5"
+#define VOLUME_STEPS_PROPERTY "ro.config.vc_call_vol_steps"
+
+/* Function pointers */
+void *(*_ril_open_client)(void);
+int (*_ril_close_client)(void *);
+int (*_ril_connect)(void *);
+int (*_ril_is_connected)(void *);
+int (*_ril_disconnect)(void *);
+int (*_ril_set_call_volume)(void *, enum ril_sound_type, int);
+int (*_ril_set_call_audio_path)(void *, enum ril_audio_path);
+int (*_ril_set_call_clock_sync)(void *, enum ril_clock_state);
+int (*_ril_register_unsolicited_handler)(void *, int, void *);
+int (*_ril_get_wb_amr)(void *, void *);
+
+/* Audio WB AMR callback */
+void (*_audio_set_wb_amr_callback)(void *, int);
+void *callback_data = NULL;
+
+void ril_register_set_wb_amr_callback(void *function, void *data)
+{
+    _audio_set_wb_amr_callback = function;
+    callback_data = data;
+}
+
+/* This is the callback function that the RIL uses to
+set the wideband AMR state */
+static int ril_set_wb_amr_callback(void *ril_client,
+                                   const void *data,
+                                   size_t datalen)
+{
+    int enable = ((int *)data)[0];
+
+    if (!callback_data || !_audio_set_wb_amr_callback)
+        return -1;
+
+    _audio_set_wb_amr_callback(callback_data, enable);
+
+    return 0;
+}
+
+static int ril_connect_if_required(struct ril_handle *ril)
+{
+    if (_ril_is_connected(ril->client))
+        return 0;
+
+    if (_ril_connect(ril->client) != RIL_CLIENT_ERR_SUCCESS) {
+        ALOGE("ril_connect() failed");
+        return -1;
+    }
+
+    /* get wb amr status to set pcm samplerate depending on
+       wb amr status when ril is connected. */
+    if(_ril_get_wb_amr)
+        _ril_get_wb_amr(ril->client, ril_set_wb_amr_callback);
+
+    return 0;
+}
+
+int ril_open(struct ril_handle *ril)
+{
+    char property[PROPERTY_VALUE_MAX];
+
+    if (!ril)
+        return -1;
+
+    ril->handle = dlopen(RIL_CLIENT_LIBPATH, RTLD_NOW);
+
+    if (!ril->handle) {
+        ALOGE("Cannot open '%s'", RIL_CLIENT_LIBPATH);
+        return -1;
+    }
+
+    _ril_open_client = dlsym(ril->handle, "OpenClient_RILD");
+    _ril_close_client = dlsym(ril->handle, "CloseClient_RILD");
+    _ril_connect = dlsym(ril->handle, "Connect_RILD");
+    _ril_is_connected = dlsym(ril->handle, "isConnected_RILD");
+    _ril_disconnect = dlsym(ril->handle, "Disconnect_RILD");
+    _ril_set_call_volume = dlsym(ril->handle, "SetCallVolume");
+    _ril_set_call_audio_path = dlsym(ril->handle, "SetCallAudioPath");
+    _ril_set_call_clock_sync = dlsym(ril->handle, "SetCallClockSync");
+    _ril_register_unsolicited_handler = dlsym(ril->handle,
+                                              "RegisterUnsolicitedHandler");
+    /* since this function is not supported in all RILs, don't require it */
+    _ril_get_wb_amr = dlsym(ril->handle, "GetWB_AMR");
+
+    if (!_ril_open_client || !_ril_close_client || !_ril_connect ||
+        !_ril_is_connected || !_ril_disconnect || !_ril_set_call_volume ||
+        !_ril_set_call_audio_path || !_ril_set_call_clock_sync ||
+        !_ril_register_unsolicited_handler) {
+        ALOGE("Cannot get symbols from '%s'", RIL_CLIENT_LIBPATH);
+        dlclose(ril->handle);
+        return -1;
+    }
+
+    ril->client = _ril_open_client();
+    if (!ril->client) {
+        ALOGE("ril_open_client() failed");
+        dlclose(ril->handle);
+        return -1;
+    }
+
+    /* register the wideband AMR callback */
+    _ril_register_unsolicited_handler(ril->client, RIL_UNSOL_WB_AMR_STATE,
+                                      ril_set_wb_amr_callback);
+
+    property_get(VOLUME_STEPS_PROPERTY, property, VOLUME_STEPS_DEFAULT);
+    ril->volume_steps_max = atoi(property);
+    /* this catches the case where VOLUME_STEPS_PROPERTY does not contain
+    an integer */
+    if (ril->volume_steps_max == 0)
+        ril->volume_steps_max = atoi(VOLUME_STEPS_DEFAULT);
+
+    return 0;
+}
+
+int ril_close(struct ril_handle *ril)
+{
+    if (!ril || !ril->handle || !ril->client)
+        return -1;
+
+    if ((_ril_disconnect(ril->client) != RIL_CLIENT_ERR_SUCCESS) ||
+        (_ril_close_client(ril->client) != RIL_CLIENT_ERR_SUCCESS)) {
+        ALOGE("ril_disconnect() or ril_close_client() failed");
+        return -1;
+    }
+
+    dlclose(ril->handle);
+    return 0;
+}
+
+int ril_set_call_volume(struct ril_handle *ril, enum ril_sound_type sound_type,
+                        float volume)
+{
+    if (ril_connect_if_required(ril))
+        return 0;
+
+    return _ril_set_call_volume(ril->client, sound_type,
+                                (int)(volume * ril->volume_steps_max));
+}
+
+int ril_set_call_audio_path(struct ril_handle *ril, enum ril_audio_path path)
+{
+    if (ril_connect_if_required(ril))
+        return 0;
+
+    return _ril_set_call_audio_path(ril->client, path);
+}
+
+int ril_set_call_clock_sync(struct ril_handle *ril, enum ril_clock_state state)
+{
+    if (ril_connect_if_required(ril))
+        return 0;
+
+    return _ril_set_call_clock_sync(ril->client, state);
+}
diff --git a/audio/ril_interface.h b/audio/ril_interface.h
new file mode 100755
index 0000000..676772c
--- /dev/null
+++ b/audio/ril_interface.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef RIL_INTERFACE_H
+#define RIL_INTERFACE_H
+
+#define RIL_CLIENT_LIBPATH "libsecril-client.so"
+
+#define RIL_CLIENT_ERR_SUCCESS      0
+#define RIL_CLIENT_ERR_AGAIN        1
+#define RIL_CLIENT_ERR_INIT         2 // Client is not initialized
+#define RIL_CLIENT_ERR_INVAL        3 // Invalid value
+#define RIL_CLIENT_ERR_CONNECT      4 // Connection error
+#define RIL_CLIENT_ERR_IO           5 // IO error
+#define RIL_CLIENT_ERR_RESOURCE     6 // Resource not available
+#define RIL_CLIENT_ERR_UNKNOWN      7
+
+#define RIL_OEM_UNSOL_RESPONSE_BASE 11000 // RIL response base index
+#define RIL_UNSOL_WB_AMR_STATE \
+    (RIL_OEM_UNSOL_RESPONSE_BASE + 17)    // RIL AMR state index
+
+struct ril_handle
+{
+    void *handle;
+    void *client;
+    int volume_steps_max;
+};
+
+enum ril_sound_type {
+    SOUND_TYPE_VOICE,
+    SOUND_TYPE_SPEAKER,
+    SOUND_TYPE_HEADSET,
+    SOUND_TYPE_BTVOICE
+};
+
+enum ril_audio_path {
+    SOUND_AUDIO_PATH_HANDSET,
+    SOUND_AUDIO_PATH_HEADSET,
+    SOUND_AUDIO_PATH_SPEAKER,
+    SOUND_AUDIO_PATH_BLUETOOTH,
+    SOUND_AUDIO_PATH_BLUETOOTH_NO_NR,
+    SOUND_AUDIO_PATH_HEADPHONE
+};
+
+enum ril_clock_state {
+    SOUND_CLOCK_STOP,
+    SOUND_CLOCK_START
+};
+
+/* Function prototypes */
+int ril_open(struct ril_handle *ril);
+int ril_close(struct ril_handle *ril);
+int ril_set_call_volume(struct ril_handle *ril, enum ril_sound_type sound_type,
+                        float volume);
+int ril_set_call_audio_path(struct ril_handle *ril, enum ril_audio_path path);
+int ril_set_call_clock_sync(struct ril_handle *ril, enum ril_clock_state state);
+void ril_register_set_wb_amr_callback(void *function, void *data);
+#endif
+