diff --git a/audio/Android.mk b/audio/Android.mk
index 228ccf2..90c18fa 100644
--- a/audio/Android.mk
+++ b/audio/Android.mk
@@ -27,12 +27,16 @@
 LOCAL_MODULE_RELATIVE_PATH := hw
 LOCAL_VENDOR_MODULE := true
 
-LOCAL_SRC_FILES := audio_hw.c
-LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa
+LOCAL_SRC_FILES := audio_hw.c \
+    audio_aec.c \
+    fifo_wrapper.cpp \
+    fir_filter.c
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioroute libaudioutils
 LOCAL_CFLAGS := -Wno-unused-parameter
 LOCAL_C_INCLUDES += \
         external/tinyalsa/include \
         external/expat/lib \
+        $(call include-path-for, audio-route) \
         system/media/audio_utils/include \
         system/media/audio_effects/include
 
diff --git a/audio/audio_aec.c b/audio/audio_aec.c
new file mode 100644
index 0000000..ab99c93
--- /dev/null
+++ b/audio/audio_aec.c
@@ -0,0 +1,700 @@
+/*
+ * Copyright (C) 2019 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.
+ */
+
+// clang-format off
+/*
+ * Typical AEC signal flow:
+ *
+ *                          Microphone Audio
+ *                          Timestamps
+ *                        +--------------------------------------+
+ *                        |                                      |       +---------------+
+ *                        |    Microphone +---------------+      |       |               |
+ *             O|======   |    Audio      | Sample Rate   |      +------->               |
+ *    (from         .  +--+    Samples    | +             |              |               |
+ *     mic          .  +==================> Format        |==============>               |
+ *     codec)       .                     | Conversion    |              |               |   Cleaned
+ *             O|======                   | (if required) |              |   Acoustic    |   Audio
+ *                                        +---------------+              |   Echo        |   Samples
+ *                                                                       |   Canceller   |===================>
+ *                                                                       |   (AEC)       |
+ *                            Reference   +---------------+              |               |
+ *                            Audio       | Sample Rate   |              |               |
+ *                            Samples     | +             |              |               |
+ *                          +=============> Format        |==============>               |
+ *                          |             | Conversion    |              |               |
+ *                          |             | (if required) |      +------->               |
+ *                          |             +---------------+      |       |               |
+ *                          |                                    |       +---------------+
+ *                          |    +-------------------------------+
+ *                          |    |  Reference Audio
+ *                          |    |  Timestamps
+ *                          |    |
+ *                       +--+----+---------+                                                       AUDIO CAPTURE
+ *                       | Speaker         |
+ *          +------------+ Audio/Timestamp +---------------------------------------------------------------------------+
+ *                       | Buffer          |
+ *                       +--^----^---------+                                                       AUDIO PLAYBACK
+ *                          |    |
+ *                          |    |
+ *                          |    |
+ *                          |    |
+ *                |\        |    |
+ *                | +-+     |    |
+ *      (to       | | +-----C----+
+ *       speaker  | | |     |                                                                  Playback
+ *       codec)   | | <=====+================================================================+ Audio
+ *                | +-+                                                                        Samples
+ *                |/
+ *
+ */
+// clang-format on
+
+#define LOG_TAG "audio_hw_aec"
+// #define LOG_NDEBUG 0
+
+#include <audio_utils/primitives.h>
+#include <stdio.h>
+#include <inttypes.h>
+#include <errno.h>
+#include <malloc.h>
+#include <sys/time.h>
+#include <tinyalsa/asoundlib.h>
+#include <unistd.h>
+#include <log/log.h>
+#include "audio_aec.h"
+
+#ifdef AEC_HAL
+#include "audio_aec_process.h"
+#else
+#define aec_spk_mic_init(...) ((int)0)
+#define aec_spk_mic_reset(...) ((void)0)
+#define aec_spk_mic_process(...) ((int32_t)0)
+#define aec_spk_mic_release(...) ((void)0)
+#endif
+
+#define MAX_TIMESTAMP_DIFF_USEC 200000
+
+#define MAX_READ_WAIT_TIME_MSEC 80
+
+uint64_t timespec_to_usec(struct timespec ts) {
+    return (ts.tv_sec * 1e6L + ts.tv_nsec/1000);
+}
+
+void get_reference_audio_in_place(struct aec_t *aec, size_t frames) {
+    if (aec->num_reference_channels == aec->spk_num_channels) {
+        /* Reference count equals speaker channels, nothing to do here. */
+        return;
+    } else if (aec->num_reference_channels != 1) {
+        /* We don't have  a rule for non-mono references, show error on log */
+        ALOGE("Invalid reference count - must be 1 or match number of playback channels!");
+        return;
+    }
+    int16_t *src_Nch = &aec->spk_buf_playback_format[0];
+    int16_t *dst_1ch = &aec->spk_buf_playback_format[0];
+    int32_t num_channels = (int32_t)aec->spk_num_channels;
+    size_t frame, ch;
+    for (frame = 0; frame < frames; frame++) {
+        int32_t acc = 0;
+        for (ch = 0; ch < aec->spk_num_channels; ch++) {
+            acc += src_Nch[ch];
+        }
+        *dst_1ch++ = clamp16(acc/num_channels);
+        src_Nch += aec->spk_num_channels;
+    }
+}
+
+void print_queue_status_to_log(struct aec_t *aec, bool write_side) {
+    ssize_t q1 = fifo_available_to_read(aec->spk_fifo);
+    ssize_t q2 = fifo_available_to_read(aec->ts_fifo);
+
+    ALOGV("Queue available %s: Spk %zd (count %zd) TS %zd (count %zd)",
+        (write_side) ? "(POST-WRITE)" : "(PRE-READ)",
+        q1, q1/aec->spk_frame_size_bytes/PLAYBACK_PERIOD_SIZE,
+        q2, q2/sizeof(struct aec_info));
+}
+
+void flush_aec_fifos(struct aec_t *aec) {
+    if (aec == NULL) {
+        return;
+    }
+    if (aec->spk_fifo != NULL) {
+        ALOGV("Flushing AEC Spk FIFO...");
+        fifo_flush(aec->spk_fifo);
+    }
+    if (aec->ts_fifo != NULL) {
+        ALOGV("Flushing AEC Timestamp FIFO...");
+        fifo_flush(aec->ts_fifo);
+    }
+    /* Reset FIFO read-write offset tracker */
+    aec->read_write_diff_bytes = 0;
+}
+
+void aec_set_spk_running_no_lock(struct aec_t* aec, bool state) {
+    aec->spk_running = state;
+}
+
+bool aec_get_spk_running_no_lock(struct aec_t* aec) {
+    return aec->spk_running;
+}
+
+void destroy_aec_reference_config_no_lock(struct aec_t* aec) {
+    if (!aec->spk_initialized) {
+        return;
+    }
+    aec_set_spk_running_no_lock(aec, false);
+    fifo_release(aec->spk_fifo);
+    fifo_release(aec->ts_fifo);
+    memset(&aec->last_spk_info, 0, sizeof(struct aec_info));
+    aec->spk_initialized = false;
+}
+
+void destroy_aec_mic_config_no_lock(struct aec_t* aec) {
+    if (!aec->mic_initialized) {
+        return;
+    }
+    release_resampler(aec->spk_resampler);
+    free(aec->mic_buf);
+    free(aec->spk_buf);
+    free(aec->spk_buf_playback_format);
+    free(aec->spk_buf_resampler_out);
+    memset(&aec->last_mic_info, 0, sizeof(struct aec_info));
+    aec->mic_initialized = false;
+}
+
+struct aec_t *init_aec_interface() {
+    ALOGV("%s enter", __func__);
+    struct aec_t *aec = (struct aec_t *)calloc(1, sizeof(struct aec_t));
+    if (aec == NULL) {
+        ALOGE("Failed to allocate memory for AEC interface!");
+    } else {
+        pthread_mutex_init(&aec->lock, NULL);
+    }
+
+    ALOGV("%s exit", __func__);
+    return aec;
+}
+
+void release_aec_interface(struct aec_t *aec) {
+    ALOGV("%s enter", __func__);
+    pthread_mutex_lock(&aec->lock);
+    destroy_aec_mic_config_no_lock(aec);
+    destroy_aec_reference_config_no_lock(aec);
+    pthread_mutex_unlock(&aec->lock);
+    free(aec);
+    ALOGV("%s exit", __func__);
+}
+
+int init_aec(int sampling_rate, int num_reference_channels,
+                int num_microphone_channels, struct aec_t **aec_ptr) {
+    ALOGV("%s enter", __func__);
+    int ret = 0;
+    int aec_ret = aec_spk_mic_init(
+                    sampling_rate,
+                    num_reference_channels,
+                    num_microphone_channels);
+    if (aec_ret) {
+        ALOGE("AEC object failed to initialize!");
+        ret = -EINVAL;
+    }
+    struct aec_t *aec = init_aec_interface();
+    if (!ret) {
+        aec->num_reference_channels = num_reference_channels;
+        /* Set defaults, will be overridden by settings in init_aec_(mic|referece_config) */
+        /* Capture uses 2-ch, 32-bit frames */
+        aec->mic_sampling_rate = CAPTURE_CODEC_SAMPLING_RATE;
+        aec->mic_frame_size_bytes = CHANNEL_STEREO * sizeof(int32_t);
+        aec->mic_num_channels = CHANNEL_STEREO;
+
+        /* Playback uses 2-ch, 16-bit frames */
+        aec->spk_sampling_rate = PLAYBACK_CODEC_SAMPLING_RATE;
+        aec->spk_frame_size_bytes = CHANNEL_STEREO * sizeof(int16_t);
+        aec->spk_num_channels = CHANNEL_STEREO;
+    }
+
+    (*aec_ptr) = aec;
+    ALOGV("%s exit", __func__);
+    return ret;
+}
+
+void release_aec(struct aec_t *aec) {
+    ALOGV("%s enter", __func__);
+    if (aec == NULL) {
+        return;
+    }
+    release_aec_interface(aec);
+    aec_spk_mic_release();
+    ALOGV("%s exit", __func__);
+}
+
+int init_aec_reference_config(struct aec_t *aec, struct alsa_stream_out *out) {
+    ALOGV("%s enter", __func__);
+    if (!aec) {
+        ALOGE("AEC: No valid interface found!");
+        return -EINVAL;
+    }
+
+    int ret = 0;
+    pthread_mutex_lock(&aec->lock);
+    if (aec->spk_initialized) {
+        destroy_aec_reference_config_no_lock(aec);
+    }
+
+    aec->spk_fifo = fifo_init(
+            out->config.period_count * out->config.period_size *
+                audio_stream_out_frame_size(&out->stream),
+            false /* reader_throttles_writer */);
+    if (aec->spk_fifo == NULL) {
+        ALOGE("AEC: Speaker loopback FIFO Init failed!");
+        ret = -EINVAL;
+        goto exit;
+    }
+    aec->ts_fifo = fifo_init(
+            out->config.period_count * sizeof(struct aec_info),
+            false /* reader_throttles_writer */);
+    if (aec->ts_fifo == NULL) {
+        ALOGE("AEC: Speaker timestamp FIFO Init failed!");
+        ret = -EINVAL;
+        fifo_release(aec->spk_fifo);
+        goto exit;
+    }
+
+    aec->spk_sampling_rate = out->config.rate;
+    aec->spk_frame_size_bytes = audio_stream_out_frame_size(&out->stream);
+    aec->spk_num_channels = out->config.channels;
+    aec->spk_initialized = true;
+exit:
+    pthread_mutex_unlock(&aec->lock);
+    ALOGV("%s exit", __func__);
+    return ret;
+}
+
+void destroy_aec_reference_config(struct aec_t* aec) {
+    ALOGV("%s enter", __func__);
+    if (aec == NULL) {
+        ALOGV("%s exit", __func__);
+        return;
+    }
+    pthread_mutex_lock(&aec->lock);
+    destroy_aec_reference_config_no_lock(aec);
+    pthread_mutex_unlock(&aec->lock);
+    ALOGV("%s exit", __func__);
+}
+
+int write_to_reference_fifo(struct aec_t* aec, void* buffer, struct aec_info* info) {
+    ALOGV("%s enter", __func__);
+    int ret = 0;
+    size_t bytes = info->bytes;
+
+    /* Write audio samples to FIFO */
+    ssize_t written_bytes = fifo_write(aec->spk_fifo, buffer, bytes);
+    if (written_bytes != bytes) {
+        ALOGE("Could only write %zu of %zu bytes", written_bytes, bytes);
+        ret = -ENOMEM;
+    }
+
+    /* Write timestamp to FIFO */
+    info->bytes = written_bytes;
+    ALOGV("Speaker timestamp: %ld s, %ld nsec", info->timestamp.tv_sec, info->timestamp.tv_nsec);
+    ssize_t ts_bytes = fifo_write(aec->ts_fifo, info, sizeof(struct aec_info));
+    ALOGV("Wrote TS bytes: %zu", ts_bytes);
+    print_queue_status_to_log(aec, true);
+    ALOGV("%s exit", __func__);
+    return ret;
+}
+
+void get_spk_timestamp(struct aec_t* aec, ssize_t read_bytes, uint64_t* spk_time) {
+    *spk_time = 0;
+    uint64_t spk_time_offset = 0;
+    float usec_per_byte = 1E6 / ((float)(aec->spk_frame_size_bytes * aec->spk_sampling_rate));
+    if (aec->read_write_diff_bytes < 0) {
+        /* We're still reading a previous write packet. (We only need the first sample's timestamp,
+         * so even if we straddle packets we only care about the first one)
+         * So we just use the previous timestamp, with an appropriate offset
+         * based on the number of bytes remaining to be read from that write packet. */
+        spk_time_offset = (aec->last_spk_info.bytes + aec->read_write_diff_bytes) * usec_per_byte;
+        ALOGV("Reusing previous timestamp, calculated offset (usec) %" PRIu64, spk_time_offset);
+    } else {
+        /* If read_write_diff_bytes > 0, there are no new writes, so there won't be timestamps in
+         * the FIFO, and the check below will fail. */
+        if (!fifo_available_to_read(aec->ts_fifo)) {
+            ALOGE("Timestamp error: no new timestamps!");
+            return;
+        }
+        /* We just read valid data, so if we're here, we should have a valid timestamp to use. */
+        ssize_t ts_bytes = fifo_read(aec->ts_fifo, &aec->last_spk_info, sizeof(struct aec_info));
+        ALOGV("Read TS bytes: %zd, expected %zu", ts_bytes, sizeof(struct aec_info));
+        aec->read_write_diff_bytes -= aec->last_spk_info.bytes;
+    }
+
+    *spk_time = timespec_to_usec(aec->last_spk_info.timestamp) + spk_time_offset;
+
+    aec->read_write_diff_bytes += read_bytes;
+    struct aec_info spk_info = aec->last_spk_info;
+    while (aec->read_write_diff_bytes > 0) {
+        /* If read_write_diff_bytes > 0, it means that there are more write packet timestamps
+         * in FIFO (since there we read more valid data the size of the current timestamp's
+         * packet). Keep reading timestamps from FIFO to get to the most recent one. */
+        if (!fifo_available_to_read(aec->ts_fifo)) {
+            /* There are no more timestamps, we have the most recent one. */
+            ALOGV("At the end of timestamp FIFO, breaking...");
+            break;
+        }
+        fifo_read(aec->ts_fifo, &spk_info, sizeof(struct aec_info));
+        ALOGV("Fast-forwarded timestamp by %zd bytes, remaining bytes: %zd,"
+              " new timestamp (usec) %" PRIu64,
+              spk_info.bytes, aec->read_write_diff_bytes, timespec_to_usec(spk_info.timestamp));
+        aec->read_write_diff_bytes -= spk_info.bytes;
+    }
+    aec->last_spk_info = spk_info;
+}
+
+int get_reference_samples(struct aec_t* aec, void* buffer, struct aec_info* info) {
+    ALOGV("%s enter", __func__);
+
+    if (!aec->spk_initialized) {
+        ALOGE("%s called with no reference initialized", __func__);
+        return -EINVAL;
+    }
+
+    size_t bytes = info->bytes;
+    const size_t frames = bytes / aec->mic_frame_size_bytes;
+    const size_t sample_rate_ratio = aec->spk_sampling_rate / aec->mic_sampling_rate;
+
+    /* Read audio samples from FIFO */
+    const size_t req_bytes = frames * sample_rate_ratio * aec->spk_frame_size_bytes;
+    ssize_t available_bytes = 0;
+    unsigned int wait_count = MAX_READ_WAIT_TIME_MSEC;
+    while (true) {
+        available_bytes = fifo_available_to_read(aec->spk_fifo);
+        if (available_bytes >= req_bytes) {
+            break;
+        } else if (available_bytes < 0) {
+            ALOGE("fifo_read returned code %zu ", available_bytes);
+            return -ENOMEM;
+        }
+
+        ALOGV("Sleeping, required bytes: %zu, available bytes: %zd", req_bytes, available_bytes);
+        usleep(1000);
+        if ((wait_count--) == 0) {
+            ALOGE("Timed out waiting for read from reference FIFO");
+            return -ETIMEDOUT;
+        }
+    }
+
+    const size_t read_bytes = fifo_read(aec->spk_fifo, aec->spk_buf_playback_format, req_bytes);
+
+    /* Get timestamp*/
+    get_spk_timestamp(aec, read_bytes, &info->timestamp_usec);
+
+    /* Get reference - could be mono, downmixed from multichannel.
+     * Reference stored at spk_buf_playback_format */
+    const size_t resampler_in_frames = frames * sample_rate_ratio;
+    get_reference_audio_in_place(aec, resampler_in_frames);
+
+    int16_t* resampler_out_buf;
+    /* Resample to mic sampling rate (16-bit resampler) */
+    if (aec->spk_resampler != NULL) {
+        size_t in_frame_count = resampler_in_frames;
+        size_t out_frame_count = frames;
+        aec->spk_resampler->resample_from_input(aec->spk_resampler, aec->spk_buf_playback_format,
+                                                &in_frame_count, aec->spk_buf_resampler_out,
+                                                &out_frame_count);
+        resampler_out_buf = aec->spk_buf_resampler_out;
+    } else {
+        if (sample_rate_ratio != 1) {
+            ALOGE("Speaker sample rate %d, mic sample rate %d but no resampler defined!",
+                  aec->spk_sampling_rate, aec->mic_sampling_rate);
+        }
+        resampler_out_buf = aec->spk_buf_playback_format;
+    }
+
+    /* Convert to 32 bit */
+    int16_t* src16 = resampler_out_buf;
+    int32_t* dst32 = buffer;
+    size_t frame, ch;
+    for (frame = 0; frame < frames; frame++) {
+        for (ch = 0; ch < aec->num_reference_channels; ch++) {
+            *dst32++ = ((int32_t)*src16++) << 16;
+        }
+    }
+
+    info->bytes = bytes;
+
+    ALOGV("%s exit", __func__);
+    return 0;
+}
+
+int init_aec_mic_config(struct aec_t *aec, struct alsa_stream_in *in) {
+    ALOGV("%s enter", __func__);
+#if DEBUG_AEC
+    remove("/data/local/traces/aec_in.pcm");
+    remove("/data/local/traces/aec_out.pcm");
+    remove("/data/local/traces/aec_ref.pcm");
+    remove("/data/local/traces/aec_timestamps.txt");
+#endif /* #if DEBUG_AEC */
+
+    if (!aec) {
+        ALOGE("AEC: No valid interface found!");
+        return -EINVAL;
+    }
+
+    int ret = 0;
+    pthread_mutex_lock(&aec->lock);
+    if (aec->mic_initialized) {
+        destroy_aec_mic_config_no_lock(aec);
+    }
+    aec->mic_sampling_rate = in->config.rate;
+    aec->mic_frame_size_bytes = audio_stream_in_frame_size(&in->stream);
+    aec->mic_num_channels = in->config.channels;
+
+    aec->mic_buf_size_bytes = in->config.period_size * audio_stream_in_frame_size(&in->stream);
+    aec->mic_buf = (int32_t *)malloc(aec->mic_buf_size_bytes);
+    if (aec->mic_buf == NULL) {
+        ret = -ENOMEM;
+        goto exit;
+    }
+    memset(aec->mic_buf, 0, aec->mic_buf_size_bytes);
+    /* Reference buffer is the same number of frames as mic,
+     * only with a different number of channels in the frame. */
+    aec->spk_buf_size_bytes = in->config.period_size * aec->spk_frame_size_bytes;
+    aec->spk_buf = (int32_t *)malloc(aec->spk_buf_size_bytes);
+    if (aec->spk_buf == NULL) {
+        ret = -ENOMEM;
+        goto exit_1;
+    }
+    memset(aec->spk_buf, 0, aec->spk_buf_size_bytes);
+
+    /* Pre-resampler buffer */
+    size_t spk_frame_out_format_bytes = aec->spk_sampling_rate / aec->mic_sampling_rate *
+                                            aec->spk_buf_size_bytes;
+    aec->spk_buf_playback_format = (int16_t *)malloc(spk_frame_out_format_bytes);
+    if (aec->spk_buf_playback_format == NULL) {
+        ret = -ENOMEM;
+        goto exit_2;
+    }
+    /* Resampler is 16-bit */
+    aec->spk_buf_resampler_out = (int16_t *)malloc(aec->spk_buf_size_bytes);
+    if (aec->spk_buf_resampler_out == NULL) {
+        ret = -ENOMEM;
+        goto exit_3;
+    }
+
+    /* Don't use resampler if it's not required */
+    if (in->config.rate == aec->spk_sampling_rate) {
+        aec->spk_resampler = NULL;
+    } else {
+        int resampler_ret = create_resampler(
+                aec->spk_sampling_rate, in->config.rate, aec->num_reference_channels,
+                RESAMPLER_QUALITY_MAX - 1, /* MAX - 1 is the real max */
+                NULL,                      /* resampler_buffer_provider */
+                &aec->spk_resampler);
+        if (resampler_ret) {
+            ALOGE("AEC: Resampler initialization failed! Error code %d", resampler_ret);
+            ret = resampler_ret;
+            goto exit_4;
+        }
+    }
+
+    flush_aec_fifos(aec);
+    aec_spk_mic_reset();
+    aec->mic_initialized = true;
+
+exit:
+    pthread_mutex_unlock(&aec->lock);
+    ALOGV("%s exit", __func__);
+    return ret;
+
+exit_4:
+    free(aec->spk_buf_resampler_out);
+exit_3:
+    free(aec->spk_buf_playback_format);
+exit_2:
+    free(aec->spk_buf);
+exit_1:
+    free(aec->mic_buf);
+    pthread_mutex_unlock(&aec->lock);
+    ALOGV("%s exit", __func__);
+    return ret;
+}
+
+void aec_set_spk_running(struct aec_t *aec, bool state) {
+    ALOGV("%s enter", __func__);
+    pthread_mutex_lock(&aec->lock);
+    aec_set_spk_running_no_lock(aec, state);
+    pthread_mutex_unlock(&aec->lock);
+    ALOGV("%s exit", __func__);
+}
+
+bool aec_get_spk_running(struct aec_t *aec) {
+    ALOGV("%s enter", __func__);
+    pthread_mutex_lock(&aec->lock);
+    bool state = aec_get_spk_running_no_lock(aec);
+    pthread_mutex_unlock(&aec->lock);
+    ALOGV("%s exit", __func__);
+    return state;
+}
+
+void destroy_aec_mic_config(struct aec_t* aec) {
+    ALOGV("%s enter", __func__);
+    if (aec == NULL) {
+        ALOGV("%s exit", __func__);
+        return;
+    }
+
+    pthread_mutex_lock(&aec->lock);
+    destroy_aec_mic_config_no_lock(aec);
+    pthread_mutex_unlock(&aec->lock);
+    ALOGV("%s exit", __func__);
+}
+
+#ifdef AEC_HAL
+int process_aec(struct aec_t *aec, void* buffer, struct aec_info *info) {
+    ALOGV("%s enter", __func__);
+    int ret = 0;
+
+    if (aec == NULL) {
+        ALOGE("AEC: Interface uninitialized! Cannot process.");
+        return -EINVAL;
+    }
+
+    if ((!aec->mic_initialized) || (!aec->spk_initialized)) {
+        ALOGE("%s called with initialization: mic: %d, spk: %d", __func__, aec->mic_initialized,
+              aec->spk_initialized);
+        return -EINVAL;
+    }
+
+    size_t bytes = info->bytes;
+
+    size_t frame_size = aec->mic_frame_size_bytes;
+    size_t in_frames = bytes / frame_size;
+
+    /* Copy raw mic samples to AEC input buffer */
+    memcpy(aec->mic_buf, buffer, bytes);
+
+    uint64_t mic_time = timespec_to_usec(info->timestamp);
+    uint64_t spk_time = 0;
+
+    /*
+     * Only run AEC if there is speaker playback.
+     * The first time speaker state changes to running, flush FIFOs, so we're not stuck
+     * processing stale reference input.
+     */
+    bool spk_running = aec_get_spk_running(aec);
+
+    if (!spk_running) {
+        /* No new playback samples, so don't run AEC.
+         * 'buffer' already contains input samples. */
+        ALOGV("Speaker not running, skipping AEC..");
+        goto exit;
+    }
+
+    if (!aec->prev_spk_running) {
+        flush_aec_fifos(aec);
+    }
+
+    /* If there's no data in FIFO, exit */
+    if (fifo_available_to_read(aec->spk_fifo) <= 0) {
+        ALOGV("Echo reference buffer empty, zeroing reference....");
+        goto exit;
+    }
+
+    print_queue_status_to_log(aec, false);
+
+    /* Get reference, with format and sample rate required by AEC */
+    struct aec_info spk_info;
+    spk_info.bytes = bytes;
+    int ref_ret = get_reference_samples(aec, aec->spk_buf, &spk_info);
+    spk_time = spk_info.timestamp_usec;
+
+    if (ref_ret) {
+        ALOGE("get_reference_samples returned code %d", ref_ret);
+        ret = -ENOMEM;
+        goto exit;
+    }
+
+    int64_t time_diff = (mic_time > spk_time) ? (mic_time - spk_time) : (spk_time - mic_time);
+    if ((spk_time == 0) || (mic_time == 0) || (time_diff > MAX_TIMESTAMP_DIFF_USEC)) {
+        ALOGV("Speaker-mic timestamps diverged, skipping AEC");
+        flush_aec_fifos(aec);
+        aec_spk_mic_reset();
+        goto exit;
+    }
+
+    ALOGV("Mic time: %"PRIu64", spk time: %"PRIu64, mic_time, spk_time);
+
+    /*
+     * AEC processing call - output stored at 'buffer'
+     */
+    int32_t aec_status = aec_spk_mic_process(
+        aec->spk_buf, spk_time,
+        aec->mic_buf, mic_time,
+        in_frames,
+        buffer);
+
+    if (!aec_status) {
+        ALOGE("AEC processing failed!");
+        ret = -EINVAL;
+    }
+
+exit:
+    aec->prev_spk_running = spk_running;
+    ALOGV("Mic time: %"PRIu64", spk time: %"PRIu64, mic_time, spk_time);
+    if (ret) {
+        /* Best we can do is copy over the raw mic signal */
+        memcpy(buffer, aec->mic_buf, bytes);
+        flush_aec_fifos(aec);
+        aec_spk_mic_reset();
+    }
+
+#if DEBUG_AEC
+    /* ref data is 32-bit at this point */
+    size_t ref_bytes = in_frames*aec->num_reference_channels*sizeof(int32_t);
+
+    FILE *fp_in = fopen("/data/local/traces/aec_in.pcm", "a+");
+    if (fp_in) {
+        fwrite((char *)aec->mic_buf, 1, bytes, fp_in);
+        fclose(fp_in);
+    } else {
+        ALOGE("AEC debug: Could not open file aec_in.pcm!");
+    }
+    FILE *fp_out = fopen("/data/local/traces/aec_out.pcm", "a+");
+    if (fp_out) {
+        fwrite((char *)buffer, 1, bytes, fp_out);
+        fclose(fp_out);
+    } else {
+        ALOGE("AEC debug: Could not open file aec_out.pcm!");
+    }
+    FILE *fp_ref = fopen("/data/local/traces/aec_ref.pcm", "a+");
+    if (fp_ref) {
+        fwrite((char *)aec->spk_buf, 1, ref_bytes, fp_ref);
+        fclose(fp_ref);
+    } else {
+        ALOGE("AEC debug: Could not open file aec_ref.pcm!");
+    }
+    FILE *fp_ts = fopen("/data/local/traces/aec_timestamps.txt", "a+");
+    if (fp_ts) {
+        fprintf(fp_ts, "%"PRIu64",%"PRIu64"\n", mic_time, spk_time);
+        fclose(fp_ts);
+    } else {
+        ALOGE("AEC debug: Could not open file aec_timestamps.txt!");
+    }
+#endif /* #if DEBUG_AEC */
+    ALOGV("%s exit", __func__);
+    return ret;
+}
+
+#endif /*#ifdef AEC_HAL*/
diff --git a/audio/audio_aec.h b/audio/audio_aec.h
new file mode 100644
index 0000000..ac7a1dd
--- /dev/null
+++ b/audio/audio_aec.h
@@ -0,0 +1,132 @@
+/*
+ * Copyright (C) 2019 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.
+ */
+
+/*
+ * Definitions and interface related to HAL implementations of Acoustic Echo Canceller (AEC).
+ *
+ * AEC cleans the microphone signal by removing from it audio data corresponding to loudspeaker
+ * playback. Note that this process can be nonlinear.
+ *
+ */
+
+#ifndef _AUDIO_AEC_H_
+#define _AUDIO_AEC_H_
+
+#include <stdint.h>
+#include <pthread.h>
+#include <sys/time.h>
+#include <hardware/audio.h>
+#include <audio_utils/resampler.h>
+#include "audio_hw.h"
+#include "fifo_wrapper.h"
+
+struct aec_t {
+    pthread_mutex_t lock;
+    size_t num_reference_channels;
+    bool mic_initialized;
+    int32_t *mic_buf;
+    size_t mic_num_channels;
+    size_t mic_buf_size_bytes;
+    size_t mic_frame_size_bytes;
+    uint32_t mic_sampling_rate;
+    struct aec_info last_mic_info;
+    bool spk_initialized;
+    int32_t *spk_buf;
+    size_t spk_num_channels;
+    size_t spk_buf_size_bytes;
+    size_t spk_frame_size_bytes;
+    uint32_t spk_sampling_rate;
+    struct aec_info last_spk_info;
+    int16_t *spk_buf_playback_format;
+    int16_t *spk_buf_resampler_out;
+    void *spk_fifo;
+    void *ts_fifo;
+    ssize_t read_write_diff_bytes;
+    struct resampler_itfe *spk_resampler;
+    bool spk_running;
+    bool prev_spk_running;
+};
+
+/* Initialize AEC object.
+ * This must be called when the audio device is opened.
+ * ALSA device mutex must be held before calling this API.
+ * Returns -EINVAL if AEC object fails to initialize, else returns 0. */
+int init_aec (int sampling_rate, int num_reference_channels,
+                int num_microphone_channels, struct aec_t **);
+
+/* Release AEC object.
+ * This must be called when the audio device is closed. */
+void release_aec(struct aec_t* aec);
+
+/* Initialize reference configuration for AEC.
+ * Must be called when a new output stream is opened.
+ * Returns -EINVAL if any processing block fails to initialize,
+ * else returns 0. */
+int init_aec_reference_config (struct aec_t *aec, struct alsa_stream_out *out);
+
+/* Clear reference configuration for AEC.
+ * Must be called when the output stream is closed. */
+void destroy_aec_reference_config (struct aec_t *aec);
+
+/* Initialize microphone configuration for AEC.
+ * Must be called when a new input stream is opened.
+ * Returns -EINVAL if any processing block fails to initialize,
+ * else returns 0. */
+int init_aec_mic_config(struct aec_t* aec, struct alsa_stream_in* in);
+
+/* Clear microphone configuration for AEC.
+ * Must be called when the input stream is closed. */
+void destroy_aec_mic_config (struct aec_t *aec);
+
+/* Used to communicate playback state (running or not) to AEC interface.
+ * This is used by process_aec() to determine if AEC processing is to be run. */
+void aec_set_spk_running (struct aec_t *aec, bool state);
+
+/* Used to communicate playback state (running or not) to the caller. */
+bool aec_get_spk_running(struct aec_t* aec);
+
+/* Write audio samples to AEC reference FIFO for use in AEC.
+ * Both audio samples and timestamps are added in FIFO fashion.
+ * Must be called after every write to PCM.
+ * Returns -ENOMEM if the write fails, else returns 0. */
+int write_to_reference_fifo(struct aec_t* aec, void* buffer, struct aec_info* info);
+
+/* Get reference audio samples + timestamp, in the format expected by AEC,
+ * i.e. same sample rate and bit rate as microphone audio.
+ * Timestamp is updated in field 'timestamp_usec', and not in 'timestamp'.
+ * Returns:
+ *  -EINVAL    if the AEC object is invalid.
+ *  -ENOMEM    if the reference FIFO overflows or is corrupted.
+ *  -ETIMEDOUT if we timed out waiting for the requested number of bytes
+ *  0          otherwise */
+int get_reference_samples(struct aec_t* aec, void* buffer, struct aec_info* info);
+
+#ifdef AEC_HAL
+
+/* Processing function call for AEC.
+ * AEC output is updated at location pointed to by 'buffer'.
+ * This function does not run AEC when there is no playback -
+ * as communicated to this AEC interface using aec_set_spk_running().
+ * Returns -EINVAL if processing fails, else returns 0. */
+int process_aec(struct aec_t* aec, void* buffer, struct aec_info* info);
+
+#else /* #ifdef AEC_HAL */
+
+#define process_aec(...) ((int)0)
+
+#endif /* #ifdef AEC_HAL */
+
+#endif /* _AUDIO_AEC_H_ */
diff --git a/audio/audio_hw.c b/audio/audio_hw.c
index 805e2cd..4a16ac1 100644
--- a/audio/audio_hw.c
+++ b/audio/audio_hw.c
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012 The Android Open Source Project
+ * Copyright (C) 2016 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.
@@ -13,811 +13,494 @@
  * See the License for the specific language governing permissions and
  * limitations under the License.
  *
- * Copied as it is from device/google/cuttlefish/guest/hals/audio/audio_hw.c
- * and fixed couple of typos pointed out by Lint during review.
+ * Copied as it is from device/amlogic/generic/hal/audio/
  */
 
-#define LOG_TAG "audio_hw_generic"
+#define LOG_TAG "audio_hw_yukawa"
+//#define LOG_NDEBUG 0
 
-#include <assert.h>
 #include <errno.h>
 #include <inttypes.h>
+#include <malloc.h>
 #include <pthread.h>
 #include <stdint.h>
 #include <stdlib.h>
 #include <sys/time.h>
-#include <dlfcn.h>
-#include <fcntl.h>
 #include <unistd.h>
 
 #include <log/log.h>
-#include <cutils/list.h>
 #include <cutils/str_parms.h>
+#include <cutils/properties.h>
 
 #include <hardware/hardware.h>
 #include <system/audio.h>
 #include <hardware/audio.h>
+
+#include <audio_effects/effect_aec.h>
+#include <audio_route/audio_route.h>
+#include <audio_utils/clock.h>
+#include <audio_utils/echo_reference.h>
+#include <audio_utils/resampler.h>
+#include <hardware/audio_alsaops.h>
+#include <hardware/audio_effect.h>
+#include <sound/asound.h>
 #include <tinyalsa/asoundlib.h>
 
-#define PCM_CARD 0
-#define PCM_DEVICE 0
+#include <sys/ioctl.h>
 
+#include "audio_aec.h"
+#include "audio_hw.h"
 
-#define OUT_PERIOD_MS 15
-#define OUT_PERIOD_COUNT 4
+static int adev_get_mic_mute(const struct audio_hw_device* dev, bool* state);
+static int adev_get_microphones(const struct audio_hw_device* dev,
+                                struct audio_microphone_characteristic_t* mic_array,
+                                size_t* mic_count);
+static size_t out_get_buffer_size(const struct audio_stream* stream);
 
-#define IN_PERIOD_MS 15
-#define IN_PERIOD_COUNT 4
-
-struct generic_audio_device {
-    struct audio_hw_device device;          // Constant after init
-    pthread_mutex_t lock;
-    bool mic_mute;                          // Protected by this->lock
-    struct mixer* mixer;                    // Protected by this->lock
-    struct listnode out_streams;            // Record for output streams, protected by this->lock
-    struct listnode in_streams;             // Record for input streams, protected by this->lock
-    audio_patch_handle_t next_patch_handle; // Protected by this->lock
-};
-
-/* If not NULL, this is a pointer to the fallback module.
- * This really is the original goldfish audio device /dev/eac which we will use
- * if no alsa devices are detected.
- */
-static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state);
-static int adev_get_microphones(const audio_hw_device_t *dev,
-                                struct audio_microphone_characteristic_t *mic_array,
-                                size_t *mic_count);
-
-
-typedef struct audio_vbuffer {
-    pthread_mutex_t lock;
-    uint8_t *  data;
-    size_t     frame_size;
-    size_t     frame_count;
-    size_t     head;
-    size_t     tail;
-    size_t     live;
-} audio_vbuffer_t;
-
-static int audio_vbuffer_init (audio_vbuffer_t * audio_vbuffer, size_t frame_count,
-                              size_t frame_size) {
-    if (!audio_vbuffer) {
-        return -EINVAL;
-    }
-    audio_vbuffer->frame_size = frame_size;
-    audio_vbuffer->frame_count = frame_count;
-    size_t bytes = frame_count * frame_size;
-    audio_vbuffer->data = calloc(bytes, 1);
-    if (!audio_vbuffer->data) {
-        return -ENOMEM;
-    }
-    audio_vbuffer->head = 0;
-    audio_vbuffer->tail = 0;
-    audio_vbuffer->live = 0;
-    pthread_mutex_init (&audio_vbuffer->lock, (const pthread_mutexattr_t *) NULL);
-    return 0;
+static int get_audio_output_port(audio_devices_t devices) {
+    /* Only HDMI out for now #FIXME */
+    return PORT_HDMI;
 }
 
-static int audio_vbuffer_destroy (audio_vbuffer_t * audio_vbuffer) {
-    if (!audio_vbuffer) {
-        return -EINVAL;
+static void timestamp_adjust(struct timespec* ts, ssize_t frames, uint32_t sampling_rate) {
+    /* This function assumes the adjustment (in nsec) is less than the max value of long,
+     * which for 32-bit long this is 2^31 * 1e-9 seconds, slightly over 2 seconds.
+     * For 64-bit long it is  9e+9 seconds. */
+    long adj_nsec = (frames / (float) sampling_rate) * 1E9L;
+    ts->tv_nsec += adj_nsec;
+    while (ts->tv_nsec > 1E9L) {
+        ts->tv_sec++;
+        ts->tv_nsec -= 1E9L;
     }
-    free(audio_vbuffer->data);
-    pthread_mutex_destroy(&audio_vbuffer->lock);
-    return 0;
+    if (ts->tv_nsec < 0) {
+        ts->tv_sec--;
+        ts->tv_nsec += 1E9L;
+    }
 }
 
-static int audio_vbuffer_live (audio_vbuffer_t * audio_vbuffer) {
-    if (!audio_vbuffer) {
+/* Helper function to get PCM hardware timestamp.
+ * Only the field 'timestamp' of argument 'ts' is updated. */
+static int get_pcm_timestamp(struct pcm* pcm, uint32_t sample_rate, struct aec_info* info,
+                             bool isOutput) {
+    int ret = 0;
+    if (pcm_get_htimestamp(pcm, &info->available, &info->timestamp) < 0) {
+        ALOGE("Error getting PCM timestamp!");
+        info->timestamp.tv_sec = 0;
+        info->timestamp.tv_nsec = 0;
         return -EINVAL;
     }
-    pthread_mutex_lock (&audio_vbuffer->lock);
-    int live = audio_vbuffer->live;
-    pthread_mutex_unlock (&audio_vbuffer->lock);
-    return live;
+    ssize_t frames;
+    if (isOutput) {
+        frames = pcm_get_buffer_size(pcm) - info->available;
+    } else {
+        frames = -info->available; /* rewind timestamp */
+    }
+    timestamp_adjust(&info->timestamp, frames, sample_rate);
+    return ret;
 }
 
-#define MIN(a,b) (((a)<(b))?(a):(b))
-static size_t audio_vbuffer_write (audio_vbuffer_t * audio_vbuffer, const void * buffer, size_t frame_count) {
-    size_t frames_written = 0;
-    pthread_mutex_lock (&audio_vbuffer->lock);
-
-    while (frame_count != 0) {
-        int frames = 0;
-        if (audio_vbuffer->live == 0 || audio_vbuffer->head > audio_vbuffer->tail) {
-            frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->head);
-        } else if (audio_vbuffer->head < audio_vbuffer->tail) {
-            frames = MIN(frame_count, audio_vbuffer->tail - (audio_vbuffer->head));
-        } else {
-            // Full
+static int read_filter_from_file(const char* filename, int16_t* filter, int max_length) {
+    FILE* fp = fopen(filename, "r");
+    if (fp == NULL) {
+        ALOGI("%s: File %s not found.", __func__, filename);
+        return 0;
+    }
+    int num_taps = 0;
+    char* line = NULL;
+    size_t len = 0;
+    while (!feof(fp)) {
+        size_t size = getline(&line, &len, fp);
+        if ((line[0] == '#') || (size < 2)) {
+            continue;
+        }
+        int n = sscanf(line, "%" SCNd16 "\n", &filter[num_taps++]);
+        if (n < 1) {
+            ALOGE("Could not find coefficient %d! Exiting...", num_taps - 1);
+            return 0;
+        }
+        ALOGV("Coeff %d : %" PRId16, num_taps, filter[num_taps - 1]);
+        if (num_taps == max_length) {
+            ALOGI("%s: max tap length %d reached.", __func__, max_length);
             break;
         }
-        memcpy(&audio_vbuffer->data[audio_vbuffer->head*audio_vbuffer->frame_size],
-               &((uint8_t*)buffer)[frames_written*audio_vbuffer->frame_size],
-               frames*audio_vbuffer->frame_size);
-        audio_vbuffer->live += frames;
-        frames_written += frames;
-        frame_count -= frames;
-        audio_vbuffer->head = (audio_vbuffer->head + frames) % audio_vbuffer->frame_count;
     }
-
-    pthread_mutex_unlock (&audio_vbuffer->lock);
-    return frames_written;
+    free(line);
+    fclose(fp);
+    return num_taps;
 }
 
-static size_t audio_vbuffer_read (audio_vbuffer_t * audio_vbuffer, void * buffer, size_t frame_count) {
-    size_t frames_read = 0;
-    pthread_mutex_lock (&audio_vbuffer->lock);
+static void out_set_eq(struct alsa_stream_out* out) {
+    out->speaker_eq = NULL;
+    int16_t* speaker_eq_coeffs = (int16_t*)calloc(SPEAKER_MAX_EQ_LENGTH, sizeof(int16_t));
+    if (speaker_eq_coeffs == NULL) {
+        ALOGE("%s: Failed to allocate speaker EQ", __func__);
+        return;
+    }
+    int num_taps = read_filter_from_file(SPEAKER_EQ_FILE, speaker_eq_coeffs, SPEAKER_MAX_EQ_LENGTH);
+    if (num_taps == 0) {
+        ALOGI("%s: Empty filter file or 0 taps set.", __func__);
+        free(speaker_eq_coeffs);
+        return;
+    }
+    out->speaker_eq = fir_init(
+            out->config.channels, FIR_SINGLE_FILTER, num_taps,
+            out_get_buffer_size(&out->stream.common) / out->config.channels / sizeof(int16_t),
+            speaker_eq_coeffs);
+    free(speaker_eq_coeffs);
+}
 
-    while (frame_count != 0) {
-        int frames = 0;
-        if (audio_vbuffer->live == audio_vbuffer->frame_count ||
-            audio_vbuffer->tail > audio_vbuffer->head) {
-            frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->tail);
-        } else if (audio_vbuffer->tail < audio_vbuffer->head) {
-            frames = MIN(frame_count, audio_vbuffer->head - audio_vbuffer->tail);
-        } else {
+/* must be called with hw device and output stream mutexes locked */
+static int start_output_stream(struct alsa_stream_out *out)
+{
+    struct alsa_audio_device *adev = out->dev;
+
+    /* default to low power: will be corrected in out_write if necessary before first write to
+     * tinyalsa.
+     */
+    out->write_threshold = PLAYBACK_PERIOD_COUNT * PLAYBACK_PERIOD_SIZE;
+    out->config.start_threshold = PLAYBACK_PERIOD_START_THRESHOLD * PLAYBACK_PERIOD_SIZE;
+    out->config.avail_min = PLAYBACK_PERIOD_SIZE;
+    out->unavailable = true;
+    unsigned int pcm_retry_count = PCM_OPEN_RETRIES;
+    int out_port = get_audio_output_port(out->devices);
+
+    while (1) {
+        out->pcm = pcm_open(CARD_OUT, out_port, PCM_OUT | PCM_MONOTONIC, &out->config);
+        if ((out->pcm != NULL) && pcm_is_ready(out->pcm)) {
             break;
+        } else {
+            ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm));
+            if (out->pcm != NULL) {
+                pcm_close(out->pcm);
+                out->pcm = NULL;
+            }
+            if (--pcm_retry_count == 0) {
+                ALOGE("Failed to open pcm_out after %d tries", PCM_OPEN_RETRIES);
+                return -ENODEV;
+            }
+            usleep(PCM_OPEN_WAIT_TIME_MS * 1000);
         }
-        memcpy(&((uint8_t*)buffer)[frames_read*audio_vbuffer->frame_size],
-               &audio_vbuffer->data[audio_vbuffer->tail*audio_vbuffer->frame_size],
-               frames*audio_vbuffer->frame_size);
-        audio_vbuffer->live -= frames;
-        frames_read += frames;
-        frame_count -= frames;
-        audio_vbuffer->tail = (audio_vbuffer->tail + frames) % audio_vbuffer->frame_count;
     }
-
-    pthread_mutex_unlock (&audio_vbuffer->lock);
-    return frames_read;
+    out->unavailable = false;
+    adev->active_output = out;
+    return 0;
 }
 
-struct generic_stream_out {
-    struct audio_stream_out stream;                 // Constant after init
-    pthread_mutex_t lock;
-    struct generic_audio_device *dev;               // Constant after init
-    uint32_t num_devices;                           // Protected by this->lock
-    audio_devices_t devices[AUDIO_PATCH_PORTS_MAX]; // Protected by this->lock
-    struct audio_config req_config;                 // Constant after init
-    struct pcm_config pcm_config;                   // Constant after init
-    audio_vbuffer_t buffer;                         // Constant after init
-
-    // Time & Position Keeping
-    bool standby;                      // Protected by this->lock
-    uint64_t underrun_position;        // Protected by this->lock
-    struct timespec underrun_time;     // Protected by this->lock
-    uint64_t last_write_time_us;       // Protected by this->lock
-    uint64_t frames_total_buffered;    // Protected by this->lock
-    uint64_t frames_written;           // Protected by this->lock
-    uint64_t frames_rendered;          // Protected by this->lock
-
-    // Worker
-    pthread_t worker_thread;          // Constant after init
-    pthread_cond_t worker_wake;       // Protected by this->lock
-    bool worker_standby;              // Protected by this->lock
-    bool worker_exit;                 // Protected by this->lock
-
-    audio_io_handle_t handle;          // Constant after init
-    audio_patch_handle_t patch_handle; // Protected by this->dev->lock
-
-    struct listnode stream_node;       // Protected by this->dev->lock
-};
-
-struct generic_stream_in {
-    struct audio_stream_in stream;    // Constant after init
-    pthread_mutex_t lock;
-    struct generic_audio_device *dev; // Constant after init
-    audio_devices_t device;           // Protected by this->lock
-    struct audio_config req_config;   // Constant after init
-    struct pcm *pcm;                  // Protected by this->lock
-    struct pcm_config pcm_config;     // Constant after init
-    int16_t *stereo_to_mono_buf;      // Protected by this->lock
-    size_t stereo_to_mono_buf_size;   // Protected by this->lock
-    audio_vbuffer_t buffer;           // Protected by this->lock
-
-    // Time & Position Keeping
-    bool standby;                     // Protected by this->lock
-    int64_t standby_position;         // Protected by this->lock
-    struct timespec standby_exit_time;// Protected by this->lock
-    int64_t standby_frames_read;      // Protected by this->lock
-
-    // Worker
-    pthread_t worker_thread;          // Constant after init
-    pthread_cond_t worker_wake;       // Protected by this->lock
-    bool worker_standby;              // Protected by this->lock
-    bool worker_exit;                 // Protected by this->lock
-
-    audio_io_handle_t handle;          // Constant after init
-    audio_patch_handle_t patch_handle; // Protected by this->dev->lock
-
-    struct listnode stream_node;       // Protected by this->dev->lock
-};
-
-static struct pcm_config pcm_config_out = {
-    .channels = 2,
-    .rate = 0,
-    .period_size = 0,
-    .period_count = OUT_PERIOD_COUNT,
-    .format = PCM_FORMAT_S16_LE,
-    .start_threshold = 0,
-};
-
-static struct pcm_config pcm_config_in = {
-    .channels = 2,
-    .rate = 0,
-    .period_size = 0,
-    .period_count = IN_PERIOD_COUNT,
-    .format = PCM_FORMAT_S16_LE,
-    .start_threshold = 0,
-    .stop_threshold = INT_MAX,
-};
-
-static pthread_mutex_t adev_init_lock = PTHREAD_MUTEX_INITIALIZER;
-static unsigned int audio_device_ref_count = 0;
-
 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    return out->req_config.sample_rate;
+    struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
+    return out->config.rate;
 }
 
 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
 {
+    ALOGV("out_set_sample_rate: %d", 0);
     return -ENOSYS;
 }
 
 static size_t out_get_buffer_size(const struct audio_stream *stream)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    int size = out->pcm_config.period_size *
-                audio_stream_out_frame_size(&out->stream);
+    ALOGV("out_get_buffer_size: %d", 4096);
 
-    return size;
+    /* return the closest majoring multiple of 16 frames, as
+     * audioflinger expects audio buffers to be a multiple of 16 frames */
+    size_t size = PLAYBACK_PERIOD_SIZE;
+    size = ((size + 15) / 16) * 16;
+    return size * audio_stream_out_frame_size((struct audio_stream_out *)stream);
 }
 
 static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    return out->req_config.channel_mask;
+    ALOGV("out_get_channels");
+    struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
+    return audio_channel_out_mask_from_count(out->config.channels);
 }
 
 static audio_format_t out_get_format(const struct audio_stream *stream)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-
-    return out->req_config.format;
+    ALOGV("out_get_format");
+    struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
+    return audio_format_from_pcm_format(out->config.format);
 }
 
 static int out_set_format(struct audio_stream *stream, audio_format_t format)
 {
+    ALOGV("out_set_format: %d",format);
     return -ENOSYS;
 }
 
+static int do_output_standby(struct alsa_stream_out *out)
+{
+    struct alsa_audio_device *adev = out->dev;
+
+    fir_reset(out->speaker_eq);
+
+    if (!out->standby) {
+        pcm_close(out->pcm);
+        out->pcm = NULL;
+        adev->active_output = NULL;
+        out->standby = 1;
+    }
+    aec_set_spk_running(adev->aec, false);
+    return 0;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    ALOGV("out_standby");
+    struct alsa_stream_out *out = (struct alsa_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)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    pthread_mutex_lock(&out->lock);
-    dprintf(fd, "\tout_dump:\n"
-                "\t\tsample rate: %u\n"
-                "\t\tbuffer size: %zu\n"
-                "\t\tchannel mask: %08x\n"
-                "\t\tformat: %d\n"
-                "\t\tdevice(s): ",
-                out_get_sample_rate(stream),
-                out_get_buffer_size(stream),
-                out_get_channels(stream),
-                out_get_format(stream));
-    if (out->num_devices == 0) {
-        dprintf(fd, "%08x\n", AUDIO_DEVICE_NONE);
-    } else {
-        for (uint32_t i = 0; i < out->num_devices; i++) {
-            if (i != 0) {
-                dprintf(fd, ", ");
-            }
-            dprintf(fd, "%08x", out->devices[i]);
-        }
-        dprintf(fd, "\n");
-    }
-    dprintf(fd, "\t\taudio dev: %p\n\n", out->dev);
-    pthread_mutex_unlock(&out->lock);
+    ALOGV("out_dump");
     return 0;
 }
 
 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
 {
+    ALOGV("out_set_parameters");
+    struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
+    struct alsa_audio_device *adev = out->dev;
     struct str_parms *parms;
     char value[32];
-    int success;
-    int ret = -EINVAL;
+    int ret, val = 0;
 
-    if (kvpairs == NULL || kvpairs[0] == 0) {
-        return 0;
-    }
     parms = str_parms_create_str(kvpairs);
-    success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
-            value, sizeof(value));
-    // As the hal version is 3.0, it must not use set parameters API to set audio devices.
-    // Instead, it should use create_audio_patch API.
-    assert(("Must not use set parameters API to set audio devices", success < 0));
 
-    if (str_parms_has_key(parms, AUDIO_PARAMETER_STREAM_FORMAT)) {
-        // match the return value of out_set_format
-        ret = -ENOSYS;
+    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 (((out->devices & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) {
+            out->devices &= ~AUDIO_DEVICE_OUT_ALL;
+            out->devices |= val;
+        }
+        pthread_mutex_unlock(&out->lock);
+        pthread_mutex_unlock(&adev->lock);
     }
 
     str_parms_destroy(parms);
-
-    if (ret == -EINVAL) {
-        ALOGW("%s(), unsupported parameter %s", __func__, kvpairs);
-        // There is not any key supported for set_parameters API.
-        // Return error when there is non-null value passed in.
-    }
-    return ret;
+    return 0;
 }
 
 static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    struct str_parms *query = str_parms_create_str(keys);
-    char *str = NULL;
-    char value[256];
-    struct str_parms *reply = str_parms_create();
-    int ret;
-    bool get = false;
-
-    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
-    if (ret >= 0) {
-        pthread_mutex_lock(&out->lock);
-        audio_devices_t device = AUDIO_DEVICE_NONE;
-        for (uint32_t i = 0; i < out->num_devices; i++) {
-            device |= out->devices[i];
-        }
-        str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, device);
-        pthread_mutex_unlock(&out->lock);
-        get = true;
-    }
-
-    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
-        value[0] = 0;
-        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
-        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
-        get = true;
-    }
-
-    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
-        value[0] = 0;
-        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
-        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
-        get = true;
-    }
-
-    if (get) {
-        str = str_parms_to_str(reply);
-    }
-    else {
-        ALOGD("%s Unsupported parameter: %s", __FUNCTION__, keys);
-    }
-
-    str_parms_destroy(query);
-    str_parms_destroy(reply);
-    return str;
+    ALOGV("out_get_parameters");
+    return strdup("");
 }
 
 static uint32_t out_get_latency(const struct audio_stream_out *stream)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    return (out->pcm_config.period_size * 1000) / out->pcm_config.rate;
+    ALOGV("out_get_latency");
+    struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
+    return (PLAYBACK_PERIOD_SIZE * PLAYBACK_PERIOD_COUNT * 1000) / out->config.rate;
 }
 
 static int out_set_volume(struct audio_stream_out *stream, float left,
-                          float right)
+        float right)
 {
+    ALOGV("out_set_volume: Left:%f Right:%f", left, right);
     return -ENOSYS;
 }
 
-static void *out_write_worker(void * args)
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
+        size_t bytes)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)args;
-    struct pcm *pcm = NULL;
-    uint8_t *buffer = NULL;
-    int buffer_frames;
-    int buffer_size;
-    bool restart = false;
-    bool shutdown = false;
-    while (true) {
-        pthread_mutex_lock(&out->lock);
-        while (out->worker_standby || restart) {
-            restart = false;
-            if (pcm) {
-                pcm_close(pcm); // Frees pcm
-                pcm = NULL;
-                free(buffer);
-                buffer=NULL;
-            }
-            if (out->worker_exit) {
-                break;
-            }
-            pthread_cond_wait(&out->worker_wake, &out->lock);
-        }
+    int ret;
+    struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
+    struct alsa_audio_device *adev = out->dev;
+    size_t frame_size = audio_stream_out_frame_size(stream);
+    size_t out_frames = bytes / frame_size;
 
-        if (out->worker_exit) {
-            if (!out->worker_standby) {
-                ALOGE("Out worker not in standby before exiting");
-            }
-            shutdown = true;
-        }
+    ALOGV("%s: devices: %d, bytes %zu", __func__, out->devices, bytes);
 
-        while (!shutdown && audio_vbuffer_live(&out->buffer) == 0) {
-            pthread_cond_wait(&out->worker_wake, &out->lock);
-        }
-
-        if (shutdown) {
-            pthread_mutex_unlock(&out->lock);
-            break;
-        }
-
-        if (!pcm) {
-            pcm = pcm_open(PCM_CARD, PCM_DEVICE,
-                          PCM_OUT | PCM_MONOTONIC, &out->pcm_config);
-            if (!pcm_is_ready(pcm)) {
-                ALOGE("pcm_open(out) failed: %s: channels %d format %d rate %d",
-                  pcm_get_error(pcm),
-                  out->pcm_config.channels,
-                  out->pcm_config.format,
-                  out->pcm_config.rate
-                   );
-                pthread_mutex_unlock(&out->lock);
-                break;
-            }
-            buffer_frames = out->pcm_config.period_size;
-            buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
-            buffer = malloc(buffer_size);
-            if (!buffer) {
-                ALOGE("could not allocate write buffer");
-                pthread_mutex_unlock(&out->lock);
-                break;
-            }
-        }
-        int frames = audio_vbuffer_read(&out->buffer, buffer, buffer_frames);
-        pthread_mutex_unlock(&out->lock);
-        int ret = pcm_write(pcm, buffer, pcm_frames_to_bytes(pcm, frames));
-        if (ret != 0) {
-            ALOGE("pcm_write failed %s", pcm_get_error(pcm));
-            restart = true;
-        }
-    }
-    if (buffer) {
-        free(buffer);
-    }
-
-    return NULL;
-}
-
-// Call with in->lock held
-static void get_current_output_position(struct generic_stream_out *out,
-                                       uint64_t * position,
-                                       struct timespec * timestamp) {
-    struct timespec curtime = { .tv_sec = 0, .tv_nsec = 0 };
-    clock_gettime(CLOCK_MONOTONIC, &curtime);
-    const int64_t now_us = (curtime.tv_sec * 1000000000LL + curtime.tv_nsec) / 1000;
-    if (timestamp) {
-        *timestamp = curtime;
-    }
-    int64_t position_since_underrun;
-    if (out->standby) {
-        position_since_underrun = 0;
-    } else {
-        const int64_t first_us = (out->underrun_time.tv_sec * 1000000000LL +
-                                  out->underrun_time.tv_nsec) / 1000;
-        position_since_underrun = (now_us - first_us) *
-                out_get_sample_rate(&out->stream.common) /
-                1000000;
-        if (position_since_underrun < 0) {
-            position_since_underrun = 0;
-        }
-    }
-    *position = out->underrun_position + position_since_underrun;
-
-    // The device will reuse the same output stream leading to periods of
-    // underrun.
-    if (*position > out->frames_written) {
-        ALOGW("Not supplying enough data to HAL, expected position %" PRIu64 " , only wrote "
-              "%" PRIu64,
-              *position, out->frames_written);
-
-        *position = out->frames_written;
-        out->underrun_position = *position;
-        out->underrun_time = curtime;
-        out->frames_total_buffered = 0;
-    }
-}
-
-
-static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
-                         size_t bytes)
-{
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    const size_t frames =  bytes / audio_stream_out_frame_size(stream);
-
+    /* 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->worker_standby) {
-        out->worker_standby = false;
-    }
-
-    uint64_t current_position;
-    struct timespec current_time;
-
-    get_current_output_position(out, &current_position, &current_time);
-    const uint64_t now_us = (current_time.tv_sec * 1000000000LL +
-                             current_time.tv_nsec) / 1000;
     if (out->standby) {
-        out->standby = false;
-        out->underrun_time = current_time;
-        out->frames_rendered = 0;
-        out->frames_total_buffered = 0;
+        ret = start_output_stream(out);
+        if (ret != 0) {
+            pthread_mutex_unlock(&adev->lock);
+            goto exit;
+        }
+        out->standby = 0;
+        aec_set_spk_running(adev->aec, true);
     }
 
-    size_t frames_written = audio_vbuffer_write(&out->buffer, buffer, frames);
-    pthread_cond_signal(&out->worker_wake);
+    pthread_mutex_unlock(&adev->lock);
 
-    /* Implementation just consumes bytes if we start getting backed up */
-    out->frames_written += frames;
-    out->frames_rendered += frames;
-    out->frames_total_buffered += frames;
-
-    // We simulate the audio device blocking when it's write buffers become
-    // full.
-
-    // At the beginning or after an underrun, try to fill up the vbuffer.
-    // This will be throttled by the PlaybackThread
-    int frames_sleep = out->frames_total_buffered < out->buffer.frame_count ? 0 : frames;
-
-    uint64_t sleep_time_us = frames_sleep * 1000000LL /
-                            out_get_sample_rate(&stream->common);
-
-    // If the write calls are delayed, subtract time off of the sleep to
-    // compensate
-    uint64_t time_since_last_write_us = now_us - out->last_write_time_us;
-    if (time_since_last_write_us < sleep_time_us) {
-        sleep_time_us -= time_since_last_write_us;
-    } else {
-        sleep_time_us = 0;
+    if (out->speaker_eq != NULL) {
+        fir_process_interleaved(out->speaker_eq, (int16_t*)buffer, (int16_t*)buffer, out_frames);
     }
-    out->last_write_time_us = now_us + sleep_time_us;
 
+    ret = pcm_write(out->pcm, buffer, out_frames * frame_size);
+    if (ret == 0) {
+        out->frames_written += out_frames;
+
+        struct aec_info info;
+        get_pcm_timestamp(out->pcm, out->config.rate, &info, true /*isOutput*/);
+        out->timestamp = info.timestamp;
+        info.bytes = out_frames * frame_size;
+        int aec_ret = write_to_reference_fifo(adev->aec, (void *)buffer, &info);
+        if (aec_ret) {
+            ALOGE("AEC: Write to speaker loopback FIFO failed!");
+        }
+    }
+
+exit:
     pthread_mutex_unlock(&out->lock);
 
-    if (sleep_time_us > 0) {
-        usleep(sleep_time_us);
+    if (ret != 0) {
+        usleep((int64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
+                out_get_sample_rate(&stream->common));
     }
 
-    if (frames_written < frames) {
-        ALOGW("Hardware backing HAL too slow, could only write %zu of %zu frames", frames_written, frames);
-    }
-
-    /* Always consume all bytes */
     return bytes;
 }
 
+static int out_get_render_position(const struct audio_stream_out *stream,
+        uint32_t *dsp_frames)
+{
+    ALOGV("out_get_render_position: dsp_frames: %p", dsp_frames);
+    return -ENOSYS;
+}
+
 static int out_get_presentation_position(const struct audio_stream_out *stream,
                                    uint64_t *frames, struct timespec *timestamp)
-
 {
     if (stream == NULL || frames == NULL || timestamp == NULL) {
         return -EINVAL;
     }
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
+    struct alsa_stream_out* out = (struct alsa_stream_out*)stream;
 
-    pthread_mutex_lock(&out->lock);
-    get_current_output_position(out, frames, timestamp);
-    pthread_mutex_unlock(&out->lock);
+    *frames = out->frames_written;
+    *timestamp = out->timestamp;
+    ALOGV("%s: frames: %" PRIu64 ", timestamp (nsec): %" PRIu64, __func__, *frames,
+          audio_utils_ns_from_timespec(timestamp));
 
     return 0;
 }
 
-static int out_get_render_position(const struct audio_stream_out *stream,
-                                   uint32_t *dsp_frames)
-{
-    if (stream == NULL || dsp_frames == NULL) {
-        return -EINVAL;
-    }
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    pthread_mutex_lock(&out->lock);
-    *dsp_frames = out->frames_rendered;
-    pthread_mutex_unlock(&out->lock);
-    return 0;
-}
-
-// Must be called with out->lock held
-static void do_out_standby(struct generic_stream_out *out)
-{
-    int frames_sleep = 0;
-    uint64_t sleep_time_us = 0;
-    if (out->standby) {
-        return;
-    }
-    while (true) {
-        get_current_output_position(out, &out->underrun_position, NULL);
-        frames_sleep = out->frames_written - out->underrun_position;
-
-        if (frames_sleep == 0) {
-            break;
-        }
-
-        sleep_time_us = frames_sleep * 1000000LL /
-                        out_get_sample_rate(&out->stream.common);
-
-        pthread_mutex_unlock(&out->lock);
-        usleep(sleep_time_us);
-        pthread_mutex_lock(&out->lock);
-    }
-    out->worker_standby = true;
-    out->standby = true;
-}
-
-static int out_standby(struct audio_stream *stream)
-{
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    pthread_mutex_lock(&out->lock);
-    do_out_standby(out);
-    pthread_mutex_unlock(&out->lock);
-    return 0;
-}
 
 static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
 {
-    // out_add_audio_effect is a no op
+    ALOGV("out_add_audio_effect: %p", effect);
     return 0;
 }
 
 static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
 {
-    // out_remove_audio_effect is a no op
+    ALOGV("out_remove_audio_effect: %p", effect);
     return 0;
 }
 
 static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
-                                        int64_t *timestamp)
+        int64_t *timestamp)
 {
+    *timestamp = 0;
+    ALOGV("out_get_next_write_timestamp: %ld", (long int)(*timestamp));
     return -ENOSYS;
 }
 
+/** audio_stream_in implementation **/
+
+/* must be called with hw device and input stream mutexes locked */
+static int start_input_stream(struct alsa_stream_in *in)
+{
+    struct alsa_audio_device *adev = in->dev;
+    in->unavailable = true;
+    unsigned int pcm_retry_count = PCM_OPEN_RETRIES;
+
+    while (1) {
+        in->pcm = pcm_open(CARD_IN, PORT_BUILTIN_MIC, PCM_IN | PCM_MONOTONIC, &in->config);
+        if ((in->pcm != NULL) && pcm_is_ready(in->pcm)) {
+            break;
+        } else {
+            ALOGE("cannot open pcm_in driver: %s", pcm_get_error(in->pcm));
+            if (in->pcm != NULL) {
+                pcm_close(in->pcm);
+                in->pcm = NULL;
+            }
+            if (--pcm_retry_count == 0) {
+                ALOGE("Failed to open pcm_in after %d tries", PCM_OPEN_RETRIES);
+                return -ENODEV;
+            }
+            usleep(PCM_OPEN_WAIT_TIME_MS * 1000);
+        }
+    }
+    in->unavailable = false;
+    adev->active_input = in;
+    return 0;
+}
+
+static void get_mic_characteristics(struct audio_microphone_characteristic_t* mic_data,
+                                    size_t* mic_count) {
+    *mic_count = 1;
+    memset(mic_data, 0, sizeof(struct audio_microphone_characteristic_t));
+    strlcpy(mic_data->device_id, "builtin_mic", AUDIO_MICROPHONE_ID_MAX_LEN - 1);
+    strlcpy(mic_data->address, "top", AUDIO_DEVICE_MAX_ADDRESS_LEN - 1);
+    memset(mic_data->channel_mapping, AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED,
+           sizeof(mic_data->channel_mapping));
+    mic_data->device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+    mic_data->sensitivity = -37.0;
+    mic_data->max_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
+    mic_data->min_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
+    mic_data->orientation.x = 0.0f;
+    mic_data->orientation.y = 0.0f;
+    mic_data->orientation.z = 0.0f;
+    mic_data->geometric_location.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+    mic_data->geometric_location.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+    mic_data->geometric_location.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
+}
+
 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
 {
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    return in->req_config.sample_rate;
+    struct alsa_stream_in *in = (struct alsa_stream_in *)stream;
+    return in->config.rate;
 }
 
 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
 {
+    ALOGV("in_set_sample_rate: %d", rate);
     return -ENOSYS;
 }
 
-static int refine_output_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
-{
-    static const uint32_t sample_rates [] = {8000,11025,16000,22050,24000,32000,
-                                            44100,48000};
-    static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
-    bool inval = false;
-    if (*format != AUDIO_FORMAT_PCM_16_BIT) {
-        *format = AUDIO_FORMAT_PCM_16_BIT;
-        inval = true;
-    }
-
-    int channel_count = popcount(*channel_mask);
-    if (channel_count != 1 && channel_count != 2) {
-        *channel_mask = AUDIO_CHANNEL_IN_STEREO;
-        inval = true;
-    }
-
-    int i;
-    for (i = 0; i < sample_rates_count; i++) {
-        if (*sample_rate < sample_rates[i]) {
-            *sample_rate = sample_rates[i];
-            inval=true;
-            break;
-        }
-        else if (*sample_rate == sample_rates[i]) {
-            break;
-        }
-        else if (i == sample_rates_count-1) {
-            // Cap it to the highest rate we support
-            *sample_rate = sample_rates[i];
-            inval=true;
-        }
-    }
-
-    if (inval) {
-        return -EINVAL;
-    }
-    return 0;
-}
-
-static int refine_input_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
-{
-    static const uint32_t sample_rates [] = {8000, 11025, 16000, 22050, 44100, 48000};
-    static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
-    bool inval = false;
-    // Only PCM_16_bit is supported. If this is changed, stereo to mono drop
-    // must be fixed in in_read
-    if (*format != AUDIO_FORMAT_PCM_16_BIT) {
-        *format = AUDIO_FORMAT_PCM_16_BIT;
-        inval = true;
-    }
-
-    int channel_count = popcount(*channel_mask);
-    if (channel_count != 1 && channel_count != 2) {
-        *channel_mask = AUDIO_CHANNEL_IN_STEREO;
-        inval = true;
-    }
-
-    int i;
-    for (i = 0; i < sample_rates_count; i++) {
-        if (*sample_rate < sample_rates[i]) {
-            *sample_rate = sample_rates[i];
-            inval=true;
-            break;
-        }
-        else if (*sample_rate == sample_rates[i]) {
-            break;
-        }
-        else if (i == sample_rates_count-1) {
-            // Cap it to the highest rate we support
-            *sample_rate = sample_rates[i];
-            inval=true;
-        }
-    }
-
-    if (inval) {
-        return -EINVAL;
-    }
-    return 0;
-}
-
-static int check_input_parameters(uint32_t sample_rate, audio_format_t format,
-                                  audio_channel_mask_t channel_mask)
-{
-    return refine_input_parameters(&sample_rate, &format, &channel_mask);
-}
-
-static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format,
-                                    audio_channel_mask_t channel_mask)
-{
-    size_t size;
-    int channel_count = popcount(channel_mask);
-    if (check_input_parameters(sample_rate, format, channel_mask) != 0)
-        return 0;
-
-    size = sample_rate*IN_PERIOD_MS/1000;
-    // Audioflinger expects audio buffers to be multiple of 16 frames
-    size = ((size + 15) / 16) * 16;
-    size *= sizeof(short) * channel_count;
-
-    return size;
-}
-
-
-static size_t in_get_buffer_size(const struct audio_stream *stream)
-{
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    int size = get_input_buffer_size(in->req_config.sample_rate,
-                                 in->req_config.format,
-                                 in->req_config.channel_mask);
-
-    return size;
+static size_t get_input_buffer_size(size_t frames, audio_format_t format,
+                                    audio_channel_mask_t channel_mask) {
+    /* return the closest majoring multiple of 16 frames, as
+     * audioflinger expects audio buffers to be a multiple of 16 frames */
+    frames = ((frames + 15) / 16) * 16;
+    size_t bytes_per_frame = audio_channel_count_from_in_mask(channel_mask) *
+                            audio_bytes_per_sample(format);
+    size_t buffer_size = frames * bytes_per_frame;
+    return buffer_size;
 }
 
 static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
 {
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    return in->req_config.channel_mask;
+    struct alsa_stream_in *in = (struct alsa_stream_in *)stream;
+    ALOGV("in_get_channels: %d", in->config.channels);
+    return audio_channel_in_mask_from_count(in->config.channels);
 }
 
 static audio_format_t in_get_format(const struct audio_stream *stream)
 {
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    return in->req_config.format;
+    struct alsa_stream_in *in = (struct alsa_stream_in *)stream;
+    ALOGV("in_get_format: %d", in->config.format);
+    return audio_format_from_pcm_format(in->config.format);
 }
 
 static int in_set_format(struct audio_stream *stream, audio_format_t format)
@@ -825,386 +508,293 @@
     return -ENOSYS;
 }
 
-static int in_dump(const struct audio_stream *stream, int fd)
+static size_t in_get_buffer_size(const struct audio_stream *stream)
 {
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
+    struct alsa_stream_in* in = (struct alsa_stream_in*)stream;
+    size_t frames = CAPTURE_PERIOD_SIZE;
+    if (in->source == AUDIO_SOURCE_ECHO_REFERENCE) {
+        frames = CAPTURE_PERIOD_SIZE * PLAYBACK_CODEC_SAMPLING_RATE / CAPTURE_CODEC_SAMPLING_RATE;
+    }
+
+    size_t buffer_size =
+            get_input_buffer_size(frames, stream->get_format(stream), stream->get_channels(stream));
+    ALOGV("in_get_buffer_size: %zu", buffer_size);
+    return buffer_size;
+}
+
+static int in_get_active_microphones(const struct audio_stream_in* stream,
+                                     struct audio_microphone_characteristic_t* mic_array,
+                                     size_t* mic_count) {
+    ALOGV("in_get_active_microphones");
+    if ((mic_array == NULL) || (mic_count == NULL)) {
+        return -EINVAL;
+    }
+    struct alsa_stream_in* in = (struct alsa_stream_in*)stream;
+    struct audio_hw_device* dev = (struct audio_hw_device*)in->dev;
+    bool mic_muted = false;
+    adev_get_mic_mute(dev, &mic_muted);
+    if ((in->source == AUDIO_SOURCE_ECHO_REFERENCE) || mic_muted) {
+        *mic_count = 0;
+        return 0;
+    }
+    adev_get_microphones(dev, mic_array, mic_count);
+    return 0;
+}
+
+static int do_input_standby(struct alsa_stream_in *in)
+{
+    struct alsa_audio_device *adev = in->dev;
+
+    if (!in->standby) {
+        pcm_close(in->pcm);
+        in->pcm = NULL;
+        adev->active_input = NULL;
+        in->standby = true;
+    }
+    return 0;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    struct alsa_stream_in *in = (struct alsa_stream_in *)stream;
+    int status;
 
     pthread_mutex_lock(&in->lock);
-    dprintf(fd, "\tin_dump:\n"
-                "\t\tsample rate: %u\n"
-                "\t\tbuffer size: %zu\n"
-                "\t\tchannel mask: %08x\n"
-                "\t\tformat: %d\n"
-                "\t\tdevice: %08x\n"
-                "\t\taudio dev: %p\n\n",
-                in_get_sample_rate(stream),
-                in_get_buffer_size(stream),
-                in_get_channels(stream),
-                in_get_format(stream),
-                in->device,
-                in->dev);
+    pthread_mutex_lock(&in->dev->lock);
+    status = do_input_standby(in);
+    pthread_mutex_unlock(&in->dev->lock);
     pthread_mutex_unlock(&in->lock);
+    return status;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    struct alsa_stream_in* in = (struct alsa_stream_in*)stream;
+    if (in->source == AUDIO_SOURCE_ECHO_REFERENCE) {
+        return 0;
+    }
+
+    struct audio_microphone_characteristic_t mic_array[AUDIO_MICROPHONE_MAX_COUNT];
+    size_t mic_count;
+
+    get_mic_characteristics(mic_array, &mic_count);
+
+    dprintf(fd, "  Microphone count: %zd\n", mic_count);
+    size_t idx;
+    for (idx = 0; idx < mic_count; idx++) {
+        dprintf(fd, "  Microphone: %zd\n", idx);
+        dprintf(fd, "    Address: %s\n", mic_array[idx].address);
+        dprintf(fd, "    Device: %d\n", mic_array[idx].device);
+        dprintf(fd, "    Sensitivity (dB): %.2f\n", mic_array[idx].sensitivity);
+    }
+
     return 0;
 }
 
 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
 {
-    struct str_parms *parms;
-    char value[32];
-    int success;
-    int ret = -EINVAL;
-
-    if (kvpairs == NULL || kvpairs[0] == 0) {
-        return 0;
-    }
-    parms = str_parms_create_str(kvpairs);
-    success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
-            value, sizeof(value));
-    // As the hal version is 3.0, it must not use set parameters API to set audio device.
-    // Instead, it should use create_audio_patch API.
-    assert(("Must not use set parameters API to set audio devices", success < 0));
-
-    if (str_parms_has_key(parms, AUDIO_PARAMETER_STREAM_FORMAT)) {
-        // match the return value of in_set_format
-        ret = -ENOSYS;
-    }
-
-    str_parms_destroy(parms);
-
-    if (ret == -EINVAL) {
-        ALOGW("%s(), unsupported parameter %s", __func__, kvpairs);
-        // There is not any key supported for set_parameters API.
-        // Return error when there is non-null value passed in.
-    }
-    return ret;
+    return 0;
 }
 
 static char * in_get_parameters(const struct audio_stream *stream,
-                                const char *keys)
+        const char *keys)
 {
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    struct str_parms *query = str_parms_create_str(keys);
-    char *str = NULL;
-    char value[256];
-    struct str_parms *reply = str_parms_create();
-    int ret;
-    bool get = false;
-
-    ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
-    if (ret >= 0) {
-        str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, in->device);
-        get = true;
-    }
-
-    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
-        value[0] = 0;
-        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
-        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
-        get = true;
-    }
-
-    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
-        value[0] = 0;
-        strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
-        str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
-        get = true;
-    }
-
-    if (get) {
-        str = str_parms_to_str(reply);
-    }
-    else {
-        ALOGD("%s Unsupported parameter: %s", __FUNCTION__, keys);
-    }
-
-    str_parms_destroy(query);
-    str_parms_destroy(reply);
-    return str;
+    return strdup("");
 }
 
 static int in_set_gain(struct audio_stream_in *stream, float gain)
 {
-    // in_set_gain is a no op
     return 0;
 }
 
-// Call with in->lock held
-static void get_current_input_position(struct generic_stream_in *in,
-                                       int64_t * position,
-                                       struct timespec * timestamp) {
-    struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
-    clock_gettime(CLOCK_MONOTONIC, &t);
-    const int64_t now_us = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
-    if (timestamp) {
-        *timestamp = t;
-    }
-    int64_t position_since_standby;
-    if (in->standby) {
-        position_since_standby = 0;
-    } else {
-        const int64_t first_us = (in->standby_exit_time.tv_sec * 1000000000LL +
-                                  in->standby_exit_time.tv_nsec) / 1000;
-        position_since_standby = (now_us - first_us) *
-                in_get_sample_rate(&in->stream.common) /
-                1000000;
-        if (position_since_standby < 0) {
-            position_since_standby = 0;
-        }
-    }
-    *position = in->standby_position + position_since_standby;
-}
-
-// Must be called with in->lock held
-static void do_in_standby(struct generic_stream_in *in)
-{
-    if (in->standby) {
-        return;
-    }
-    in->worker_standby = true;
-    get_current_input_position(in, &in->standby_position, NULL);
-    in->standby = true;
-}
-
-static int in_standby(struct audio_stream *stream)
-{
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    pthread_mutex_lock(&in->lock);
-    do_in_standby(in);
-    pthread_mutex_unlock(&in->lock);
-    return 0;
-}
-
-static void *in_read_worker(void * args)
-{
-    struct generic_stream_in *in = (struct generic_stream_in *)args;
-    struct pcm *pcm = NULL;
-    uint8_t *buffer = NULL;
-    size_t buffer_frames;
-    int buffer_size;
-
-    bool restart = false;
-    bool shutdown = false;
-    while (true) {
-        pthread_mutex_lock(&in->lock);
-        while (in->worker_standby || restart) {
-            restart = false;
-            if (pcm) {
-                pcm_close(pcm); // Frees pcm
-                pcm = NULL;
-                free(buffer);
-                buffer=NULL;
-            }
-            if (in->worker_exit) {
-                break;
-            }
-            pthread_cond_wait(&in->worker_wake, &in->lock);
-        }
-
-        if (in->worker_exit) {
-            if (!in->worker_standby) {
-                ALOGE("In worker not in standby before exiting");
-            }
-            shutdown = true;
-        }
-        if (shutdown) {
-            pthread_mutex_unlock(&in->lock);
-            break;
-        }
-        if (!pcm) {
-            pcm = pcm_open(PCM_CARD, PCM_DEVICE,
-                          PCM_IN | PCM_MONOTONIC, &in->pcm_config);
-            if (!pcm_is_ready(pcm)) {
-                ALOGE("pcm_open(in) failed: %s: channels %d format %d rate %d",
-                  pcm_get_error(pcm),
-                  in->pcm_config.channels,
-                  in->pcm_config.format,
-                  in->pcm_config.rate
-                   );
-                pthread_mutex_unlock(&in->lock);
-                break;
-            }
-            buffer_frames = in->pcm_config.period_size;
-            buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
-            buffer = malloc(buffer_size);
-            if (!buffer) {
-                ALOGE("could not allocate worker read buffer");
-                pthread_mutex_unlock(&in->lock);
-                break;
-            }
-        }
-        pthread_mutex_unlock(&in->lock);
-        int ret = pcm_read(pcm, buffer, pcm_frames_to_bytes(pcm, buffer_frames));
-        if (ret != 0) {
-            ALOGW("pcm_read failed %s", pcm_get_error(pcm));
-            restart = true;
-            continue;
-        }
-
-        pthread_mutex_lock(&in->lock);
-        size_t frames_written = audio_vbuffer_write(&in->buffer, buffer, buffer_frames);
-        pthread_mutex_unlock(&in->lock);
-
-        if (frames_written != buffer_frames) {
-            ALOGW("in_read_worker only could write %zu / %zu frames", frames_written, buffer_frames);
-        }
-    }
-    if (buffer) {
-        free(buffer);
-    }
-    return NULL;
-}
-
 static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
-                       size_t bytes)
+        size_t bytes)
 {
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    struct generic_audio_device *adev = in->dev;
-    const size_t frames =  bytes / audio_stream_in_frame_size(stream);
-    bool mic_mute = false;
-    size_t read_bytes = 0;
+    int ret;
+    struct alsa_stream_in *in = (struct alsa_stream_in *)stream;
+    struct alsa_audio_device *adev = in->dev;
+    size_t frame_size = audio_stream_in_frame_size(stream);
+    size_t in_frames = bytes / frame_size;
 
-    adev_get_mic_mute(&adev->device, &mic_mute);
-    pthread_mutex_lock(&in->lock);
+    ALOGV("in_read: stream: %d, bytes %zu", in->source, bytes);
 
-    if (in->worker_standby) {
-        in->worker_standby = false;
-    }
-    pthread_cond_signal(&in->worker_wake);
+    /* Special handling for Echo Reference: simply get the reference from FIFO.
+     * The format and sample rate should be specified by arguments to adev_open_input_stream. */
+    if (in->source == AUDIO_SOURCE_ECHO_REFERENCE) {
+        struct aec_info info;
+        info.bytes = bytes;
 
-    int64_t current_position;
-    struct timespec current_time;
-
-    get_current_input_position(in, &current_position, &current_time);
-    if (in->standby) {
-        in->standby = false;
-        in->standby_exit_time = current_time;
-        in->standby_frames_read = 0;
-    }
-
-    const int64_t frames_available = current_position - in->standby_position - in->standby_frames_read;
-    assert(frames_available >= 0);
-
-    const size_t frames_wait = ((uint64_t)frames_available > frames) ? 0 : frames - frames_available;
-
-    int64_t sleep_time_us  = frames_wait * 1000000LL /
-                             in_get_sample_rate(&stream->common);
-
-    pthread_mutex_unlock(&in->lock);
-
-    if (sleep_time_us > 0) {
-        usleep(sleep_time_us);
-    }
-
-    pthread_mutex_lock(&in->lock);
-    int read_frames = 0;
-    if (in->standby) {
-        ALOGW("Input put to sleep while read in progress");
-        goto exit;
-    }
-    in->standby_frames_read += frames;
-
-    if (popcount(in->req_config.channel_mask) == 1 &&
-        in->pcm_config.channels == 2) {
-        // Need to resample to mono
-        if (in->stereo_to_mono_buf_size < bytes*2) {
-            in->stereo_to_mono_buf = realloc(in->stereo_to_mono_buf,
-                                             bytes*2);
-            if (!in->stereo_to_mono_buf) {
-                ALOGE("Failed to allocate stereo_to_mono_buff");
-                goto exit;
+        const uint64_t time_increment_nsec = (uint64_t)bytes * NANOS_PER_SECOND /
+                                             audio_stream_in_frame_size(stream) /
+                                             in_get_sample_rate(&stream->common);
+        if (!aec_get_spk_running(adev->aec)) {
+            if (in->timestamp_nsec == 0) {
+                struct timespec now;
+                clock_gettime(CLOCK_MONOTONIC, &now);
+                const uint64_t timestamp_nsec = audio_utils_ns_from_timespec(&now);
+                in->timestamp_nsec = timestamp_nsec;
+            } else {
+                in->timestamp_nsec += time_increment_nsec;
+            }
+            memset(buffer, 0, bytes);
+            const uint64_t time_increment_usec = time_increment_nsec / 1000;
+            usleep(time_increment_usec);
+        } else {
+            int ref_ret = get_reference_samples(adev->aec, buffer, &info);
+            if ((ref_ret) || (info.timestamp_usec == 0)) {
+                memset(buffer, 0, bytes);
+                in->timestamp_nsec += time_increment_nsec;
+            } else {
+                in->timestamp_nsec = 1000 * info.timestamp_usec;
             }
         }
+        in->frames_read += in_frames;
 
-        read_frames = audio_vbuffer_read(&in->buffer, in->stereo_to_mono_buf, frames);
-
-        // Currently only pcm 16 is supported.
-        uint16_t *src = (uint16_t *)in->stereo_to_mono_buf;
-        uint16_t *dst = (uint16_t *)buffer;
-        size_t i;
-        // Resample stereo 16 to mono 16 by dropping one channel.
-        // The stereo stream is interleaved L-R-L-R
-        for (i = 0; i < frames; i++) {
-            *dst = *src;
-            src += 2;
-            dst += 1;
+#if DEBUG_AEC
+        FILE* fp_ref = fopen("/data/local/traces/aec_ref.pcm", "a+");
+        if (fp_ref) {
+            fwrite((char*)buffer, 1, bytes, fp_ref);
+            fclose(fp_ref);
+        } else {
+            ALOGE("AEC debug: Could not open file aec_ref.pcm!");
         }
-    } else {
-        read_frames = audio_vbuffer_read(&in->buffer, buffer, frames);
+        FILE* fp_ref_ts = fopen("/data/local/traces/aec_ref_timestamps.txt", "a+");
+        if (fp_ref_ts) {
+            fprintf(fp_ref_ts, "%" PRIu64 "\n", in->timestamp_nsec);
+            fclose(fp_ref_ts);
+        } else {
+            ALOGE("AEC debug: Could not open file aec_ref_timestamps.txt!");
+        }
+#endif
+        return info.bytes;
+    }
+
+    /* Microphone input stream read */
+
+    /* 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(&in->lock);
+    pthread_mutex_lock(&adev->lock);
+    if (in->standby) {
+        ret = start_input_stream(in);
+        if (ret != 0) {
+            pthread_mutex_unlock(&adev->lock);
+            ALOGE("start_input_stream failed with code %d", ret);
+            goto exit;
+        }
+        in->standby = false;
+    }
+
+    pthread_mutex_unlock(&adev->lock);
+
+    ret = pcm_read(in->pcm, buffer, in_frames * frame_size);
+    struct aec_info info;
+    get_pcm_timestamp(in->pcm, in->config.rate, &info, false /*isOutput*/);
+    if (ret == 0) {
+        in->frames_read += in_frames;
+        in->timestamp_nsec = audio_utils_ns_from_timespec(&info.timestamp);
+    }
+    else {
+        ALOGE("pcm_read failed with code %d", ret);
     }
 
 exit:
-    read_bytes = read_frames*audio_stream_in_frame_size(stream);
-
-    if (mic_mute) {
-        read_bytes = 0;
-    }
-
-    if (read_bytes < bytes) {
-        memset (&((uint8_t *)buffer)[read_bytes], 0, bytes-read_bytes);
-    }
-
     pthread_mutex_unlock(&in->lock);
 
+    bool mic_muted = false;
+    adev_get_mic_mute((struct audio_hw_device*)adev, &mic_muted);
+    if (mic_muted) {
+        memset(buffer, 0, bytes);
+    }
+
+    if (ret != 0) {
+        usleep((int64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
+                in_get_sample_rate(&stream->common));
+    } else {
+        /* Process AEC if available */
+        /* TODO move to a separate thread */
+        if (!mic_muted) {
+            info.bytes = bytes;
+            int aec_ret = process_aec(adev->aec, buffer, &info);
+            if (aec_ret) {
+                ALOGE("process_aec returned error code %d", aec_ret);
+            }
+        }
+    }
+
+#if DEBUG_AEC && !defined(AEC_HAL)
+    FILE* fp_in = fopen("/data/local/traces/aec_in.pcm", "a+");
+    if (fp_in) {
+        fwrite((char*)buffer, 1, bytes, fp_in);
+        fclose(fp_in);
+    } else {
+        ALOGE("AEC debug: Could not open file aec_in.pcm!");
+    }
+    FILE* fp_mic_ts = fopen("/data/local/traces/aec_in_timestamps.txt", "a+");
+    if (fp_mic_ts) {
+        fprintf(fp_mic_ts, "%" PRIu64 "\n", in->timestamp_nsec);
+        fclose(fp_mic_ts);
+    } else {
+        ALOGE("AEC debug: Could not open file aec_in_timestamps.txt!");
+    }
+#endif
+
     return bytes;
 }
 
+static int in_get_capture_position(const struct audio_stream_in* stream, int64_t* frames,
+                                   int64_t* time) {
+    if (stream == NULL || frames == NULL || time == NULL) {
+        return -EINVAL;
+    }
+    struct alsa_stream_in* in = (struct alsa_stream_in*)stream;
+
+    *frames = in->frames_read;
+    *time = in->timestamp_nsec;
+    ALOGV("%s: source: %d, timestamp (nsec): %" PRIu64, __func__, in->source, *time);
+
+    return 0;
+}
+
 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
 {
     return 0;
 }
 
-static int in_get_capture_position(const struct audio_stream_in *stream,
-                                int64_t *frames, int64_t *time)
-{
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    pthread_mutex_lock(&in->lock);
-    struct timespec current_time;
-    get_current_input_position(in, frames, &current_time);
-    *time = (current_time.tv_sec * 1000000000LL + current_time.tv_nsec);
-    pthread_mutex_unlock(&in->lock);
-    return 0;
-}
-
-static int in_get_active_microphones(const struct audio_stream_in *stream,
-                                     struct audio_microphone_characteristic_t *mic_array,
-                                     size_t *mic_count)
-{
-    return adev_get_microphones(NULL, mic_array, mic_count);
-}
-
 static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
 {
-    // in_add_audio_effect is a no op
     return 0;
 }
 
 static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
 {
-    // in_add_audio_effect is a no op
     return 0;
 }
 
 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,
-                                   const char *address __unused)
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        audio_output_flags_t flags,
+        struct audio_config *config,
+        struct audio_stream_out **stream_out,
+        const char *address __unused)
 {
-    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
-    struct generic_stream_out *out;
+    ALOGV("adev_open_output_stream...");
+
+    struct alsa_audio_device *ladev = (struct alsa_audio_device *)dev;
+    struct alsa_stream_out *out;
+    struct pcm_params *params;
     int ret = 0;
 
-    if (refine_output_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
-        ALOGE("Error opening output stream format %d, channel_mask %04x, sample_rate %u",
-              config->format, config->channel_mask, config->sample_rate);
-        ret = -EINVAL;
-        goto error;
-    }
+    int out_port = get_audio_output_port(devices);
 
-    out = (struct generic_stream_out *)calloc(1, sizeof(struct generic_stream_out));
+    params = pcm_params_get(CARD_OUT, out_port, PCM_OUT);
+    if (!params)
+        return -ENOSYS;
 
+    out = (struct alsa_stream_out *)calloc(1, sizeof(struct alsa_stream_out));
     if (!out)
         return -ENOMEM;
 
@@ -1224,141 +814,141 @@
     out->stream.set_volume = out_set_volume;
     out->stream.write = out_write;
     out->stream.get_render_position = out_get_render_position;
-    out->stream.get_presentation_position = out_get_presentation_position;
     out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
+    out->stream.get_presentation_position = out_get_presentation_position;
 
-    out->handle = handle;
+    out->config.channels = CHANNEL_STEREO;
+    out->config.rate = PLAYBACK_CODEC_SAMPLING_RATE;
+    out->config.format = PCM_FORMAT_S16_LE;
+    out->config.period_size = PLAYBACK_PERIOD_SIZE;
+    out->config.period_count = PLAYBACK_PERIOD_COUNT;
 
-    pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
-    out->dev = adev;
-    // Only 1 device is expected despite the argument being named 'devices'
-    out->num_devices = 1;
-    out->devices[0] = devices;
-    memcpy(&out->req_config, config, sizeof(struct audio_config));
-    memcpy(&out->pcm_config, &pcm_config_out, sizeof(struct pcm_config));
-    out->pcm_config.rate = config->sample_rate;
-    out->pcm_config.period_size = out->pcm_config.rate*OUT_PERIOD_MS/1000;
-
-    out->standby = true;
-    out->underrun_position = 0;
-    out->underrun_time.tv_sec = 0;
-    out->underrun_time.tv_nsec = 0;
-    out->last_write_time_us = 0;
-    out->frames_total_buffered = 0;
-    out->frames_written = 0;
-    out->frames_rendered = 0;
-
-    ret = audio_vbuffer_init(&out->buffer,
-                      out->pcm_config.period_size*out->pcm_config.period_count,
-                      out->pcm_config.channels *
-                      pcm_format_to_bits(out->pcm_config.format) >> 3);
-    if (ret == 0) {
-        pthread_cond_init(&out->worker_wake, NULL);
-        out->worker_standby = true;
-        out->worker_exit = false;
-        pthread_create(&out->worker_thread, NULL, out_write_worker, out);
-
+    if (out->config.rate != config->sample_rate ||
+           audio_channel_count_from_out_mask(config->channel_mask) != CHANNEL_STEREO ||
+               out->config.format !=  pcm_format_from_audio_format(config->format) ) {
+        config->sample_rate = out->config.rate;
+        config->format = audio_format_from_pcm_format(out->config.format);
+        config->channel_mask = audio_channel_out_mask_from_count(CHANNEL_STEREO);
+        ret = -EINVAL;
     }
 
-    pthread_mutex_lock(&adev->lock);
-    list_add_tail(&adev->out_streams, &out->stream_node);
-    pthread_mutex_unlock(&adev->lock);
+    ALOGI("adev_open_output_stream selects channels=%d rate=%d format=%d, devices=%d",
+          out->config.channels, out->config.rate, out->config.format, devices);
+
+    out->dev = ladev;
+    out->standby = 1;
+    out->unavailable = false;
+    out->devices = devices;
+
+    config->format = out_get_format(&out->stream.common);
+    config->channel_mask = out_get_channels(&out->stream.common);
+    config->sample_rate = out_get_sample_rate(&out->stream.common);
 
     *stream_out = &out->stream;
 
-error:
+    out->speaker_eq = NULL;
+    if (out_port == PORT_INTERNAL_SPEAKER) {
+        out_set_eq(out);
+        if (out->speaker_eq == NULL) {
+            ALOGE("%s: Failed to initialize speaker EQ", __func__);
+        }
+    }
+
+    /* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
+    ret = 0;
+
+    if (ret == 0) {
+        int aec_ret = init_aec_reference_config(ladev->aec, out);
+        if (aec_ret) {
+            ALOGE("AEC: Speaker config init failed!");
+            return -EINVAL;
+        }
+    }
 
     return ret;
 }
 
-// This must be called with adev->lock held.
-struct generic_stream_out *get_stream_out_by_io_handle_l(
-        struct generic_audio_device *adev, audio_io_handle_t handle) {
-    struct listnode *node;
-
-    list_for_each(node, &adev->out_streams) {
-        struct generic_stream_out *out = node_to_item(
-                node, struct generic_stream_out, stream_node);
-        if (out->handle == handle) {
-            return out;
-        }
-    }
-    return NULL;
-}
-
 static void adev_close_output_stream(struct audio_hw_device *dev,
-                                     struct audio_stream_out *stream)
+        struct audio_stream_out *stream)
 {
-    struct generic_stream_out *out = (struct generic_stream_out *)stream;
-    pthread_mutex_lock(&out->lock);
-    do_out_standby(out);
-
-    out->worker_exit = true;
-    pthread_cond_signal(&out->worker_wake);
-    pthread_mutex_unlock(&out->lock);
-
-    pthread_join(out->worker_thread, NULL);
-    pthread_mutex_destroy(&out->lock);
-    audio_vbuffer_destroy(&out->buffer);
-
-    struct generic_audio_device *adev = (struct generic_audio_device *) dev;
-    pthread_mutex_lock(&adev->lock);
-    list_remove(&out->stream_node);
-    pthread_mutex_unlock(&adev->lock);
+    ALOGV("adev_close_output_stream...");
+    struct alsa_audio_device *adev = (struct alsa_audio_device *)dev;
+    destroy_aec_reference_config(adev->aec);
+    struct alsa_stream_out* out = (struct alsa_stream_out*)stream;
+    fir_release(out->speaker_eq);
     free(stream);
 }
 
 static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
 {
-    return 0;
+    ALOGV("adev_set_parameters");
+    return -ENOSYS;
 }
 
 static char * adev_get_parameters(const struct audio_hw_device *dev,
-                                  const char *keys)
+        const char *keys)
 {
+    ALOGV("adev_get_parameters");
     return strdup("");
 }
 
+static int adev_get_microphones(const struct audio_hw_device* dev,
+                                struct audio_microphone_characteristic_t* mic_array,
+                                size_t* mic_count) {
+    ALOGV("adev_get_microphones");
+    if ((mic_array == NULL) || (mic_count == NULL)) {
+        return -EINVAL;
+    }
+    get_mic_characteristics(mic_array, mic_count);
+    return 0;
+}
+
 static int adev_init_check(const struct audio_hw_device *dev)
 {
+    ALOGV("adev_init_check");
     return 0;
 }
 
 static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
 {
-    // adev_set_voice_volume is a no op (simulates phones)
-    return 0;
+    ALOGV("adev_set_voice_volume: %f", volume);
+    return -ENOSYS;
 }
 
 static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
 {
+    ALOGV("adev_set_master_volume: %f", volume);
     return -ENOSYS;
 }
 
 static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
 {
+    ALOGV("adev_get_master_volume: %f", *volume);
     return -ENOSYS;
 }
 
 static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
 {
+    ALOGV("adev_set_master_mute: %d", muted);
     return -ENOSYS;
 }
 
 static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
 {
+    ALOGV("adev_get_master_mute: %d", *muted);
     return -ENOSYS;
 }
 
 static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
 {
-    // adev_set_mode is a no op (simulates phones)
+    ALOGV("adev_set_mode: %d", mode);
     return 0;
 }
 
 static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
 {
-    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    ALOGV("adev_set_mic_mute: %d",state);
+    struct alsa_audio_device *adev = (struct alsa_audio_device *)dev;
     pthread_mutex_lock(&adev->lock);
     adev->mic_mute = state;
     pthread_mutex_unlock(&adev->lock);
@@ -1367,455 +957,206 @@
 
 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
 {
-    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
+    ALOGV("adev_get_mic_mute");
+    struct alsa_audio_device *adev = (struct alsa_audio_device *)dev;
     pthread_mutex_lock(&adev->lock);
     *state = adev->mic_mute;
     pthread_mutex_unlock(&adev->lock);
     return 0;
 }
 
-
 static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
-                                         const struct audio_config *config)
+        const struct audio_config *config)
 {
-    return get_input_buffer_size(config->sample_rate, config->format, config->channel_mask);
+    size_t buffer_size =
+            get_input_buffer_size(CAPTURE_PERIOD_SIZE, config->format, config->channel_mask);
+    ALOGV("adev_get_input_buffer_size: %zu", buffer_size);
+    return buffer_size;
 }
 
-// This must be called with adev->lock held.
-struct generic_stream_in *get_stream_in_by_io_handle_l(
-        struct generic_audio_device *adev, audio_io_handle_t handle) {
-    struct listnode *node;
+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,
+                                  audio_input_flags_t flags __unused, const char* address __unused,
+                                  audio_source_t source) {
+    ALOGV("adev_open_input_stream...");
 
-    list_for_each(node, &adev->in_streams) {
-        struct generic_stream_in *in = node_to_item(
-                node, struct generic_stream_in, stream_node);
-        if (in->handle == handle) {
-            return in;
-        }
-    }
-    return NULL;
-}
-
-static void adev_close_input_stream(struct audio_hw_device *dev,
-                                   struct audio_stream_in *stream)
-{
-    struct generic_stream_in *in = (struct generic_stream_in *)stream;
-    pthread_mutex_lock(&in->lock);
-    do_in_standby(in);
-
-    in->worker_exit = true;
-    pthread_cond_signal(&in->worker_wake);
-    pthread_mutex_unlock(&in->lock);
-    pthread_join(in->worker_thread, NULL);
-
-    if (in->stereo_to_mono_buf != NULL) {
-        free(in->stereo_to_mono_buf);
-        in->stereo_to_mono_buf_size = 0;
-    }
-
-    pthread_mutex_destroy(&in->lock);
-    audio_vbuffer_destroy(&in->buffer);
-
-    struct generic_audio_device *adev = (struct generic_audio_device *) dev;
-    pthread_mutex_lock(&adev->lock);
-    list_remove(&in->stream_node);
-    pthread_mutex_unlock(&adev->lock);
-    free(stream);
-}
-
-
-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,
-                                  audio_input_flags_t flags __unused,
-                                  const char *address __unused,
-                                  audio_source_t source __unused)
-{
-    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
-    struct generic_stream_in *in;
+    struct alsa_audio_device *ladev = (struct alsa_audio_device *)dev;
+    struct alsa_stream_in *in;
+    struct pcm_params *params;
     int ret = 0;
-    if (refine_input_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
-        ALOGE("Error opening input stream format %d, channel_mask %04x, sample_rate %u",
-              config->format, config->channel_mask, config->sample_rate);
-        ret = -EINVAL;
-        goto error;
-    }
 
-    in = (struct generic_stream_in *)calloc(1, sizeof(struct generic_stream_in));
-    if (!in) {
-        ret = -ENOMEM;
-        goto error;
-    }
+    params = pcm_params_get(CARD_IN, PORT_BUILTIN_MIC, PCM_IN);
+    if (!params)
+        return -ENOSYS;
+
+    in = (struct alsa_stream_in *)calloc(1, sizeof(struct alsa_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;         // no op
+    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;                   // no op
+    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;       // no op
-    in->stream.common.remove_audio_effect = in_remove_audio_effect; // no op
-    in->stream.set_gain = in_set_gain;                              // no op
+    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;    // no op
+    in->stream.get_input_frames_lost = in_get_input_frames_lost;
     in->stream.get_capture_position = in_get_capture_position;
     in->stream.get_active_microphones = in_get_active_microphones;
 
-    pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
-    in->dev = adev;
-    in->device = devices;
-    memcpy(&in->req_config, config, sizeof(struct audio_config));
-    memcpy(&in->pcm_config, &pcm_config_in, sizeof(struct pcm_config));
-    in->pcm_config.rate = config->sample_rate;
-    in->pcm_config.period_size = in->pcm_config.rate*IN_PERIOD_MS/1000;
+    in->config.channels = CHANNEL_STEREO;
+    if (source == AUDIO_SOURCE_ECHO_REFERENCE) {
+        in->config.rate = PLAYBACK_CODEC_SAMPLING_RATE;
+    } else {
+        in->config.rate = CAPTURE_CODEC_SAMPLING_RATE;
+    }
+    in->config.format = PCM_FORMAT_S32_LE;
+    in->config.period_size = CAPTURE_PERIOD_SIZE;
+    in->config.period_count = CAPTURE_PERIOD_COUNT;
 
-    in->stereo_to_mono_buf = NULL;
-    in->stereo_to_mono_buf_size = 0;
+    if (in->config.rate != config->sample_rate ||
+           audio_channel_count_from_in_mask(config->channel_mask) != CHANNEL_STEREO ||
+               in->config.format !=  pcm_format_from_audio_format(config->format) ) {
+        ret = -EINVAL;
+    }
 
+    ALOGI("adev_open_input_stream selects channels=%d rate=%d format=%d source=%d",
+          in->config.channels, in->config.rate, in->config.format, source);
+
+    in->dev = ladev;
     in->standby = true;
-    in->standby_position = 0;
-    in->standby_exit_time.tv_sec = 0;
-    in->standby_exit_time.tv_nsec = 0;
-    in->standby_frames_read = 0;
+    in->unavailable = false;
+    in->source = source;
+    in->devices = devices;
 
-    ret = audio_vbuffer_init(&in->buffer,
-                      in->pcm_config.period_size*in->pcm_config.period_count,
-                      in->pcm_config.channels *
-                      pcm_format_to_bits(in->pcm_config.format) >> 3);
-    if (ret == 0) {
-        pthread_cond_init(&in->worker_wake, NULL);
-        in->worker_standby = true;
-        in->worker_exit = false;
-        pthread_create(&in->worker_thread, NULL, in_read_worker, in);
-    }
-    in->handle = handle;
+    config->format = in_get_format(&in->stream.common);
+    config->channel_mask = in_get_channels(&in->stream.common);
+    config->sample_rate = in_get_sample_rate(&in->stream.common);
 
-    pthread_mutex_lock(&adev->lock);
-    list_add_tail(&adev->in_streams, &in->stream_node);
-    pthread_mutex_unlock(&adev->lock);
+    /* If AEC is in the app, only configure based on ECHO_REFERENCE spec.
+     * If AEC is in the HAL, configure using the given mic stream. */
+    bool aecInput = true;
+#if !defined(AEC_HAL)
+    aecInput = (in->source == AUDIO_SOURCE_ECHO_REFERENCE);
+#endif
 
-    *stream_in = &in->stream;
-
-error:
-    return ret;
-}
-
-
-static int adev_dump(const audio_hw_device_t *dev, int fd)
-{
-    return 0;
-}
-
-static int adev_get_microphones(const audio_hw_device_t *dev,
-                                struct audio_microphone_characteristic_t *mic_array,
-                                size_t *mic_count)
-{
-    if (mic_count == NULL) {
-        return -ENOSYS;
-    }
-
-    if (*mic_count == 0) {
-        *mic_count = 1;
-        return 0;
-    }
-
-    if (mic_array == NULL) {
-        return -ENOSYS;
-    }
-
-    strncpy(mic_array->device_id, "mic_goldfish", AUDIO_MICROPHONE_ID_MAX_LEN - 1);
-    mic_array->device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-    strncpy(mic_array->address, AUDIO_BOTTOM_MICROPHONE_ADDRESS,
-            AUDIO_DEVICE_MAX_ADDRESS_LEN - 1);
-    memset(mic_array->channel_mapping, AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED,
-           sizeof(mic_array->channel_mapping));
-    mic_array->location = AUDIO_MICROPHONE_LOCATION_UNKNOWN;
-    mic_array->group = 0;
-    mic_array->index_in_the_group = 0;
-    mic_array->sensitivity = AUDIO_MICROPHONE_SENSITIVITY_UNKNOWN;
-    mic_array->max_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
-    mic_array->min_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
-    mic_array->directionality = AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN;
-    mic_array->num_frequency_responses = 0;
-    mic_array->geometric_location.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
-    mic_array->geometric_location.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
-    mic_array->geometric_location.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
-    mic_array->orientation.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
-    mic_array->orientation.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
-    mic_array->orientation.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
-
-    *mic_count = 1;
-    return 0;
-}
-
-static int adev_create_audio_patch(struct audio_hw_device *dev,
-                                   unsigned int num_sources,
-                                   const struct audio_port_config *sources,
-                                   unsigned int num_sinks,
-                                   const struct audio_port_config *sinks,
-                                   audio_patch_handle_t *handle) {
-    if (num_sources != 1 || num_sinks == 0 || num_sinks > AUDIO_PATCH_PORTS_MAX) {
-        return -EINVAL;
-    }
-
-    if (sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
-        // If source is a device, the number of sinks should be 1.
-        if (num_sinks != 1 || sinks[0].type != AUDIO_PORT_TYPE_MIX) {
+    if ((ret == 0) && aecInput) {
+        int aec_ret = init_aec_mic_config(ladev->aec, in);
+        if (aec_ret) {
+            ALOGE("AEC: Mic config init failed!");
             return -EINVAL;
         }
-    } else if (sources[0].type == AUDIO_PORT_TYPE_MIX) {
-        // If source is a mix, all sinks should be device.
-        for (unsigned int i = 0; i < num_sinks; i++) {
-            if (sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
-                ALOGE("%s() invalid sink type %#x for mix source", __func__, sinks[i].type);
-                return -EINVAL;
-            }
-        }
+    }
+
+    if (ret) {
+        free(in);
     } else {
-        // All other cases are invalid.
-        return -EINVAL;
+        *stream_in = &in->stream;
     }
 
-    struct generic_audio_device* adev = (struct generic_audio_device*) dev;
-    int ret = 0;
-    bool generatedPatchHandle = false;
-    pthread_mutex_lock(&adev->lock);
-    if (*handle == AUDIO_PATCH_HANDLE_NONE) {
-        *handle = ++adev->next_patch_handle;
-        generatedPatchHandle = true;
-    }
+#if DEBUG_AEC
+    remove("/data/local/traces/aec_ref.pcm");
+    remove("/data/local/traces/aec_in.pcm");
+    remove("/data/local/traces/aec_ref_timestamps.txt");
+    remove("/data/local/traces/aec_in_timestamps.txt");
+#endif
+    return ret;
+}
 
-    // Only handle patches for mix->devices and device->mix case.
-    if (sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
-        struct generic_stream_in *in =
-                get_stream_in_by_io_handle_l(adev, sinks[0].ext.mix.handle);
-        if (in == NULL) {
-            ALOGE("%s()can not find stream with handle(%d)", __func__, sources[0].ext.mix.handle);
-            ret = -EINVAL;
-            goto error;
-        }
+static void adev_close_input_stream(struct audio_hw_device *dev,
+        struct audio_stream_in *stream)
+{
+    ALOGV("adev_close_input_stream...");
+    struct alsa_audio_device *adev = (struct alsa_audio_device *)dev;
+    destroy_aec_mic_config(adev->aec);
+    free(stream);
+    return;
+}
 
-        // Check if the patch handle match the recorded one if a valid patch handle is passed.
-        if (!generatedPatchHandle && in->patch_handle != *handle) {
-            ALOGE("%s() the patch handle(%d) does not match recorded one(%d) for stream "
-                  "with handle(%d) when creating audio patch for device->mix",
-                  __func__, *handle, in->patch_handle, in->handle);
-            ret = -EINVAL;
-            goto error;
-        }
-        pthread_mutex_lock(&in->lock);
-        in->device = sources[0].ext.device.type;
-        pthread_mutex_unlock(&in->lock);
-        in->patch_handle = *handle;
-    } else {
-        struct generic_stream_out *out =
-                get_stream_out_by_io_handle_l(adev, sources[0].ext.mix.handle);
-        if (out == NULL) {
-            ALOGE("%s()can not find stream with handle(%d)", __func__, sources[0].ext.mix.handle);
-            ret = -EINVAL;
-            goto error;
-        }
-
-        // Check if the patch handle match the recorded one if a valid patch handle is passed.
-        if (!generatedPatchHandle && out->patch_handle != *handle) {
-            ALOGE("%s() the patch handle(%d) does not match recorded one(%d) for stream "
-                  "with handle(%d) when creating audio patch for mix->device",
-                  __func__, *handle, out->patch_handle, out->handle);
-            ret = -EINVAL;
-            pthread_mutex_unlock(&out->lock);
-            goto error;
-        }
-        pthread_mutex_lock(&out->lock);
-        for (out->num_devices = 0; out->num_devices < num_sinks; out->num_devices++) {
-            out->devices[out->num_devices] = sinks[out->num_devices].ext.device.type;
-        }
-        pthread_mutex_unlock(&out->lock);
-        out->patch_handle = *handle;
-    }
-
-error:
-    if (ret != 0 && generatedPatchHandle) {
-        *handle = AUDIO_PATCH_HANDLE_NONE;
-    }
-    pthread_mutex_unlock(&adev->lock);
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+    ALOGV("adev_dump");
     return 0;
 }
 
-// This must be called with adev->lock held.
-struct generic_stream_out *get_stream_out_by_patch_handle_l(
-        struct generic_audio_device *adev, audio_patch_handle_t patch_handle) {
-    struct listnode *node;
-
-    list_for_each(node, &adev->out_streams) {
-        struct generic_stream_out *out = node_to_item(
-                node, struct generic_stream_out, stream_node);
-        if (out->patch_handle == patch_handle) {
-            return out;
-        }
-    }
-    return NULL;
-}
-
-// This must be called with adev->lock held.
-struct generic_stream_in *get_stream_in_by_patch_handle_l(
-        struct generic_audio_device *adev, audio_patch_handle_t patch_handle) {
-    struct listnode *node;
-
-    list_for_each(node, &adev->in_streams) {
-        struct generic_stream_in *in = node_to_item(
-                node, struct generic_stream_in, stream_node);
-        if (in->patch_handle == patch_handle) {
-            return in;
-        }
-    }
-    return NULL;
-}
-
-static int adev_release_audio_patch(struct audio_hw_device *dev,
-                                    audio_patch_handle_t patch_handle) {
-    struct generic_audio_device *adev = (struct generic_audio_device *) dev;
-
-    pthread_mutex_lock(&adev->lock);
-    struct generic_stream_out *out = get_stream_out_by_patch_handle_l(adev, patch_handle);
-    if (out != NULL) {
-        pthread_mutex_lock(&out->lock);
-        out->num_devices = 0;
-        memset(out->devices, 0, sizeof(out->devices));
-        pthread_mutex_unlock(&out->lock);
-        out->patch_handle = AUDIO_PATCH_HANDLE_NONE;
-        pthread_mutex_unlock(&adev->lock);
-        return 0;
-    }
-    struct generic_stream_in *in = get_stream_in_by_patch_handle_l(adev, patch_handle);
-    if (in != NULL) {
-        pthread_mutex_lock(&in->lock);
-        in->device = AUDIO_DEVICE_NONE;
-        pthread_mutex_unlock(&in->lock);
-        in->patch_handle = AUDIO_PATCH_HANDLE_NONE;
-        pthread_mutex_unlock(&adev->lock);
-        return 0;
-    }
-
-    pthread_mutex_unlock(&adev->lock);
-    ALOGW("%s() cannot find stream for patch handle: %d", __func__, patch_handle);
-    return -EINVAL;
-}
-
-static int adev_close(hw_device_t *dev)
+static int adev_close(hw_device_t *device)
 {
-    struct generic_audio_device *adev = (struct generic_audio_device *)dev;
-    int ret = 0;
-    if (!adev)
-        return 0;
+    ALOGV("adev_close");
 
-    pthread_mutex_lock(&adev_init_lock);
-
-    if (audio_device_ref_count == 0) {
-        ALOGE("adev_close called when ref_count 0");
-        ret = -EINVAL;
-        goto error;
-    }
-
-    if ((--audio_device_ref_count) == 0) {
-        if (adev->mixer) {
-            mixer_close(adev->mixer);
-        }
-        free(adev);
-    }
-
-error:
-    pthread_mutex_unlock(&adev_init_lock);
-    return ret;
+    struct alsa_audio_device *adev = (struct alsa_audio_device *)device;
+    release_aec(adev->aec);
+    free(device);
+    return 0;
 }
 
 static int adev_open(const hw_module_t* module, const char* name,
-                     hw_device_t** device)
+        hw_device_t** device)
 {
-    static struct generic_audio_device *adev;
+    struct alsa_audio_device *adev;
+
+    ALOGV("adev_open: %s", name);
 
     if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
         return -EINVAL;
 
-    pthread_mutex_lock(&adev_init_lock);
-    if (audio_device_ref_count != 0) {
-        *device = &adev->device.common;
-        audio_device_ref_count++;
-        ALOGV("%s: returning existing instance of adev", __func__);
-        ALOGV("%s: exit", __func__);
-        goto unlock;
-    }
-    adev = calloc(1, sizeof(struct generic_audio_device));
+    adev = calloc(1, sizeof(struct alsa_audio_device));
+    if (!adev)
+        return -ENOMEM;
 
-    pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
+    adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
+    adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
+    adev->hw_device.common.module = (struct hw_module_t *) module;
+    adev->hw_device.common.close = adev_close;
+    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.get_master_volume = adev_get_master_volume;
+    adev->hw_device.set_master_mute = adev_set_master_mute;
+    adev->hw_device.get_master_mute = adev_get_master_mute;
+    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->hw_device.get_microphones = adev_get_microphones;
 
-    adev->device.common.tag = HARDWARE_DEVICE_TAG;
-    adev->device.common.version = AUDIO_DEVICE_API_VERSION_3_0;
-    adev->device.common.module = (struct hw_module_t *) module;
-    adev->device.common.close = adev_close;
+    *device = &adev->hw_device.common;
 
-    adev->device.init_check = adev_init_check;               // no op
-    adev->device.set_voice_volume = adev_set_voice_volume;   // no op
-    adev->device.set_master_volume = adev_set_master_volume; // no op
-    adev->device.get_master_volume = adev_get_master_volume; // no op
-    adev->device.set_master_mute = adev_set_master_mute;     // no op
-    adev->device.get_master_mute = adev_get_master_mute;     // no op
-    adev->device.set_mode = adev_set_mode;                   // no op
-    adev->device.set_mic_mute = adev_set_mic_mute;
-    adev->device.get_mic_mute = adev_get_mic_mute;
-    adev->device.set_parameters = adev_set_parameters;       // no op
-    adev->device.get_parameters = adev_get_parameters;       // no op
-    adev->device.get_input_buffer_size = adev_get_input_buffer_size;
-    adev->device.open_output_stream = adev_open_output_stream;
-    adev->device.close_output_stream = adev_close_output_stream;
-    adev->device.open_input_stream = adev_open_input_stream;
-    adev->device.close_input_stream = adev_close_input_stream;
-    adev->device.dump = adev_dump;
-    adev->device.get_microphones = adev_get_microphones;
-    adev->device.create_audio_patch = adev_create_audio_patch;
-    adev->device.release_audio_patch = adev_release_audio_patch;
+    adev->mixer = mixer_open(CARD_OUT);
 
-    *device = &adev->device.common;
-
-    adev->next_patch_handle = AUDIO_PATCH_HANDLE_NONE;
-    list_init(&adev->out_streams);
-    list_init(&adev->in_streams);
-
-    adev->mixer = mixer_open(PCM_CARD);
-    struct mixer_ctl *ctl;
-
-    // Set default mixer ctls
-    // Enable channels and set volume
-    for (int i = 0; i < (int)mixer_get_num_ctls(adev->mixer); i++) {
-        ctl = mixer_get_ctl(adev->mixer, i);
-        ALOGD("mixer %d name %s", i, mixer_ctl_get_name(ctl));
-        if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Volume") ||
-            !strcmp(mixer_ctl_get_name(ctl), "Capture Volume")) {
-            for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
-                ALOGD("set ctl %d to %d", z, 100);
-                mixer_ctl_set_percent(ctl, z, 100);
-            }
-            continue;
-        }
-        if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Switch") ||
-            !strcmp(mixer_ctl_get_name(ctl), "Capture Switch")) {
-            for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
-                ALOGD("set ctl %d to %d", z, 1);
-                mixer_ctl_set_value(ctl, z, 1);
-            }
-            continue;
-        }
+    if (!adev->mixer) {
+        ALOGE("Unable to open the mixer, aborting.");
+        return -EINVAL;
     }
 
-    audio_device_ref_count++;
+    adev->audio_route = audio_route_init(CARD_OUT, MIXER_XML_PATH);
+    if (!adev->audio_route) {
+        ALOGE("%s: Failed to init audio route controls, aborting.", __func__);
+        return -EINVAL;
+    }
 
-unlock:
-    pthread_mutex_unlock(&adev_init_lock);
+    pthread_mutex_lock(&adev->lock);
+    if (init_aec(CAPTURE_CODEC_SAMPLING_RATE, NUM_AEC_REFERENCE_CHANNELS,
+                    CHANNEL_STEREO, &adev->aec)) {
+        pthread_mutex_unlock(&adev->lock);
+        return -EINVAL;
+    }
+    pthread_mutex_unlock(&adev->lock);
+
     return 0;
 }
 
@@ -1829,7 +1170,7 @@
         .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
         .hal_api_version = HARDWARE_HAL_API_VERSION,
         .id = AUDIO_HARDWARE_MODULE_ID,
-        .name = "Generic audio HW HAL",
+        .name = "Yukawa audio HW HAL",
         .author = "The Android Open Source Project",
         .methods = &hal_module_methods,
     },
diff --git a/audio/audio_hw.h b/audio/audio_hw.h
new file mode 100644
index 0000000..3e8e27c
--- /dev/null
+++ b/audio/audio_hw.h
@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2019 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 _YUKAWA_AUDIO_HW_H_
+#define _YUKAWA_AUDIO_HW_H_
+
+#include <hardware/audio.h>
+#include <tinyalsa/asoundlib.h>
+
+#include "fir_filter.h"
+
+#define CARD_OUT 0
+#define PORT_HDMI 0
+#define PORT_INTERNAL_SPEAKER 1
+#define CARD_IN 0
+#define PORT_BUILTIN_MIC 3
+
+#define MIXER_XML_PATH "/vendor/etc/mixer_paths.xml"
+/* Minimum granularity - Arbitrary but small value */
+#define CODEC_BASE_FRAME_COUNT 32
+
+#define CHANNEL_STEREO 2
+
+#ifdef AEC_HAL
+#define NUM_AEC_REFERENCE_CHANNELS 1
+#else
+/* App AEC uses 2-channel reference */
+#define NUM_AEC_REFERENCE_CHANNELS 2
+#endif /* #ifdef AEC_HAL */
+
+#define DEBUG_AEC 0
+
+#define PCM_OPEN_RETRIES 100
+#define PCM_OPEN_WAIT_TIME_MS 20
+
+/* Capture codec parameters */
+/* Set up a capture period of 32 ms:
+ * CAPTURE_PERIOD = PERIOD_SIZE / SAMPLE_RATE, so (32e-3) = PERIOD_SIZE / (16e3)
+ * => PERIOD_SIZE = 512 frames, where each "frame" consists of 1 sample of every channel (here, 2ch) */
+#define CAPTURE_PERIOD_MULTIPLIER 16
+#define CAPTURE_PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * CAPTURE_PERIOD_MULTIPLIER)
+#define CAPTURE_PERIOD_COUNT 4
+#define CAPTURE_PERIOD_START_THRESHOLD 0
+#define CAPTURE_CODEC_SAMPLING_RATE 16000
+
+/* Playback codec parameters */
+/* number of base blocks in a short period (low latency) */
+#define PLAYBACK_PERIOD_MULTIPLIER 32  /* 21 ms */
+/* number of frames per short period (low latency) */
+#define PLAYBACK_PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * PLAYBACK_PERIOD_MULTIPLIER)
+/* number of pseudo periods for low latency playback */
+#define PLAYBACK_PERIOD_COUNT 4
+#define PLAYBACK_PERIOD_START_THRESHOLD 2
+#define PLAYBACK_CODEC_SAMPLING_RATE 48000
+#define MIN_WRITE_SLEEP_US      5000
+
+#define SPEAKER_EQ_FILE "/vendor/etc/speaker_eq.fir"
+#define SPEAKER_MAX_EQ_LENGTH 512
+
+struct alsa_audio_device {
+    struct audio_hw_device hw_device;
+
+    pthread_mutex_t lock;   /* see notes in in_read/out_write on mutex acquisition order */
+    struct alsa_stream_in *active_input;
+    struct alsa_stream_out *active_output;
+    struct audio_route *audio_route;
+    struct mixer *mixer;
+    bool mic_mute;
+    struct aec_t *aec;
+};
+
+struct alsa_stream_in {
+    struct audio_stream_in stream;
+
+    pthread_mutex_t lock;   /* see note in in_read() on mutex acquisition order */
+    audio_devices_t devices;
+    struct pcm_config config;
+    struct pcm *pcm;
+    bool unavailable;
+    bool standby;
+    struct alsa_audio_device *dev;
+    int read_threshold;
+    unsigned int frames_read;
+    uint64_t timestamp_nsec;
+    audio_source_t source;
+};
+
+struct alsa_stream_out {
+    struct audio_stream_out stream;
+
+    pthread_mutex_t lock;   /* see note in out_write() on mutex acquisition order */
+    audio_devices_t devices;
+    struct pcm_config config;
+    struct pcm *pcm;
+    bool unavailable;
+    int standby;
+    struct alsa_audio_device *dev;
+    int write_threshold;
+    unsigned int frames_written;
+    struct timespec timestamp;
+    fir_filter_t* speaker_eq;
+};
+
+/* 'bytes' are the number of bytes written to audio FIFO, for which 'timestamp' is valid.
+ * 'available' is the number of frames available to read (for input) or yet to be played
+ * (for output) frames in the PCM buffer.
+ * timestamp and available are updated by pcm_get_htimestamp(), so they use the same
+ * datatypes as the corresponding arguments to that function. */
+struct aec_info {
+    struct timespec timestamp;
+    uint64_t timestamp_usec;
+    unsigned int available;
+    size_t bytes;
+};
+
+#endif /* #ifndef _YUKAWA_AUDIO_HW_H_ */
diff --git a/audio/fifo_wrapper.cpp b/audio/fifo_wrapper.cpp
new file mode 100644
index 0000000..7bc9079
--- /dev/null
+++ b/audio/fifo_wrapper.cpp
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2019 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 LOG_TAG "audio_utils_fifo_wrapper"
+// #define LOG_NDEBUG 0
+
+#include <stdint.h>
+#include <errno.h>
+#include <log/log.h>
+#include <audio_utils/fifo.h>
+#include "fifo_wrapper.h"
+
+struct audio_fifo_itfe {
+    audio_utils_fifo *p_fifo;
+    audio_utils_fifo_reader *p_fifo_reader;
+    audio_utils_fifo_writer *p_fifo_writer;
+    int8_t *p_buffer;
+};
+
+void *fifo_init(uint32_t bytes, bool reader_throttles_writer) {
+    struct audio_fifo_itfe *interface = new struct audio_fifo_itfe;
+    interface->p_buffer = new int8_t[bytes];
+    if (interface->p_buffer == NULL) {
+        ALOGE("Failed to allocate fifo buffer!");
+        return NULL;
+    }
+    interface->p_fifo = new audio_utils_fifo(bytes, 1, interface->p_buffer, reader_throttles_writer);
+    interface->p_fifo_writer = new audio_utils_fifo_writer(*interface->p_fifo);
+    interface->p_fifo_reader = new audio_utils_fifo_reader(*interface->p_fifo);
+
+    return (void *)interface;
+}
+
+void fifo_release(void *fifo_itfe) {
+    struct audio_fifo_itfe *interface = static_cast<struct audio_fifo_itfe *>(fifo_itfe);
+    delete interface->p_fifo_writer;
+    delete interface->p_fifo_reader;
+    delete interface->p_fifo;
+    delete[] interface->p_buffer;
+    delete interface;
+}
+
+ssize_t fifo_read(void *fifo_itfe, void *buffer, size_t bytes) {
+    struct audio_fifo_itfe *interface = static_cast<struct audio_fifo_itfe *>(fifo_itfe);
+    return interface->p_fifo_reader->read(buffer, bytes);
+}
+
+ssize_t fifo_write(void *fifo_itfe, void *buffer, size_t bytes) {
+    struct audio_fifo_itfe *interface = static_cast<struct audio_fifo_itfe *>(fifo_itfe);
+    return interface->p_fifo_writer->write(buffer, bytes);
+}
+
+ssize_t fifo_available_to_read(void *fifo_itfe) {
+    struct audio_fifo_itfe *interface = static_cast<struct audio_fifo_itfe *>(fifo_itfe);
+    return interface->p_fifo_reader->available();
+}
+
+ssize_t fifo_available_to_write(void *fifo_itfe) {
+    struct audio_fifo_itfe *interface = static_cast<struct audio_fifo_itfe *>(fifo_itfe);
+    return interface->p_fifo_writer->available();
+}
+
+ssize_t fifo_flush(void *fifo_itfe) {
+    struct audio_fifo_itfe *interface = static_cast<struct audio_fifo_itfe *>(fifo_itfe);
+    return interface->p_fifo_reader->flush();
+}
diff --git a/audio/fifo_wrapper.h b/audio/fifo_wrapper.h
new file mode 100644
index 0000000..e9469ef
--- /dev/null
+++ b/audio/fifo_wrapper.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) 2019 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 _AUDIO_FIFO_WRAPPER_H_
+#define _AUDIO_FIFO_WRAPPER_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+void *fifo_init(uint32_t bytes, bool reader_throttles_writer);
+void fifo_release(void *fifo_itfe);
+ssize_t fifo_read(void *fifo_itfe, void *buffer, size_t bytes);
+ssize_t fifo_write(void *fifo_itfe, void *buffer, size_t bytes);
+ssize_t fifo_available_to_read(void *fifo_itfe);
+ssize_t fifo_available_to_write(void *fifo_itfe);
+ssize_t fifo_flush(void *fifo_itfe);
+
+#ifdef __cplusplus
+}
+#endif
+#endif /* #ifndef _AUDIO_FIFO_WRAPPER_H_ */
diff --git a/audio/fir_filter.c b/audio/fir_filter.c
new file mode 100644
index 0000000..c648fc0
--- /dev/null
+++ b/audio/fir_filter.c
@@ -0,0 +1,154 @@
+/*
+ * Copyright (C) 2020 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 LOG_TAG "audio_hw_fir_filter"
+//#define LOG_NDEBUG 0
+
+#include <assert.h>
+#include <audio_utils/primitives.h>
+#include <errno.h>
+#include <inttypes.h>
+#include <log/log.h>
+#include <malloc.h>
+#include <string.h>
+
+#include "fir_filter.h"
+
+#ifdef __ARM_NEON
+#include "arm_neon.h"
+#endif /* #ifdef __ARM_NEON */
+
+fir_filter_t* fir_init(uint32_t channels, fir_filter_mode_t mode, uint32_t filter_length,
+                       uint32_t input_length, int16_t* coeffs) {
+    if ((channels == 0) || (filter_length == 0) || (coeffs == NULL)) {
+        ALOGE("%s: Invalid channel count, filter length or coefficient array.", __func__);
+        return NULL;
+    }
+
+    fir_filter_t* fir = (fir_filter_t*)calloc(1, sizeof(fir_filter_t));
+    if (fir == NULL) {
+        ALOGE("%s: Unable to allocate memory for fir_filter.", __func__);
+        return NULL;
+    }
+
+    fir->channels = channels;
+    fir->filter_length = filter_length;
+    /* Default: same filter coeffs for all channels */
+    fir->mode = FIR_SINGLE_FILTER;
+    uint32_t coeff_bytes = fir->filter_length * sizeof(int16_t);
+    if (mode == FIR_PER_CHANNEL_FILTER) {
+        fir->mode = FIR_PER_CHANNEL_FILTER;
+        coeff_bytes = fir->filter_length * fir->channels * sizeof(int16_t);
+    }
+
+    fir->coeffs = (int16_t*)malloc(coeff_bytes);
+    if (fir->coeffs == NULL) {
+        ALOGE("%s: Unable to allocate memory for FIR coeffs", __func__);
+        goto exit_1;
+    }
+    memcpy(fir->coeffs, coeffs, coeff_bytes);
+
+    fir->buffer_size = (input_length + fir->filter_length) * fir->channels;
+    fir->state = (int16_t*)malloc(fir->buffer_size * sizeof(int16_t));
+    if (fir->state == NULL) {
+        ALOGE("%s: Unable to allocate memory for FIR state", __func__);
+        goto exit_2;
+    }
+
+#ifdef __ARM_NEON
+    ALOGI("%s: Using ARM Neon", __func__);
+#endif /* #ifdef __ARM_NEON */
+
+    fir_reset(fir);
+    return fir;
+
+exit_2:
+    free(fir->coeffs);
+exit_1:
+    free(fir);
+    return NULL;
+}
+
+void fir_release(fir_filter_t* fir) {
+    if (fir == NULL) {
+        return;
+    }
+    free(fir->state);
+    free(fir->coeffs);
+    free(fir);
+}
+
+void fir_reset(fir_filter_t* fir) {
+    if (fir == NULL) {
+        return;
+    }
+    memset(fir->state, 0, fir->buffer_size * sizeof(int16_t));
+}
+
+void fir_process_interleaved(fir_filter_t* fir, int16_t* input, int16_t* output, uint32_t samples) {
+    assert(fir != NULL);
+
+    int start_offset = (fir->filter_length - 1) * fir->channels;
+    memcpy(&fir->state[start_offset], input, samples * fir->channels * sizeof(int16_t));
+    // int ch;
+    bool use_2nd_set_coeffs = (fir->channels > 1) && (fir->mode == FIR_PER_CHANNEL_FILTER);
+    int16_t* p_coeff_A = &fir->coeffs[0];
+    int16_t* p_coeff_B = use_2nd_set_coeffs ? &fir->coeffs[fir->filter_length] : &fir->coeffs[0];
+    int16_t* p_output;
+    for (int ch = 0; ch < fir->channels; ch += 2) {
+        p_output = &output[ch];
+        int offset = start_offset + ch;
+        for (int s = 0; s < samples; s++) {
+            int32_t acc_A = 0;
+            int32_t acc_B = 0;
+
+#ifdef __ARM_NEON
+            int32x4_t acc_vec = vdupq_n_s32(0);
+            for (int k = 0; k < fir->filter_length; k++, offset -= fir->channels) {
+                int16x4_t coeff_vec = vdup_n_s16(p_coeff_A[k]);
+                coeff_vec = vset_lane_s16(p_coeff_B[k], coeff_vec, 1);
+                int16x4_t input_vec = vld1_s16(&fir->state[offset]);
+                acc_vec = vmlal_s16(acc_vec, coeff_vec, input_vec);
+            }
+            acc_A = vgetq_lane_s32(acc_vec, 0);
+            acc_B = vgetq_lane_s32(acc_vec, 1);
+#else
+            for (int k = 0; k < fir->filter_length; k++, offset -= fir->channels) {
+                int32_t input_A = (int32_t)(fir->state[offset]);
+                int32_t coeff_A = (int32_t)(p_coeff_A[k]);
+                int32_t input_B = (int32_t)(fir->state[offset + 1]);
+                int32_t coeff_B = (int32_t)(p_coeff_B[k]);
+                acc_A += (input_A * coeff_A);
+                acc_B += (input_B * coeff_B);
+            }
+#endif /* #ifdef __ARM_NEON */
+
+            *p_output = clamp16(acc_A >> 15);
+            if (ch < fir->channels - 1) {
+                *(p_output + 1) = clamp16(acc_B >> 15);
+            }
+            /* Move to next sample */
+            p_output += fir->channels;
+            offset += (fir->filter_length + 1) * fir->channels;
+        }
+        if (use_2nd_set_coeffs) {
+            p_coeff_A += (fir->filter_length << 1);
+            p_coeff_B += (fir->filter_length << 1);
+        }
+    }
+    memmove(fir->state, &fir->state[samples * fir->channels],
+            (fir->filter_length - 1) * fir->channels * sizeof(int16_t));
+}
diff --git a/audio/fir_filter.h b/audio/fir_filter.h
new file mode 100644
index 0000000..d8c6e91
--- /dev/null
+++ b/audio/fir_filter.h
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2020 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 FIR_FILTER_H
+#define FIR_FILTER_H
+
+#include <stdint.h>
+
+typedef enum fir_filter_mode { FIR_SINGLE_FILTER = 0, FIR_PER_CHANNEL_FILTER } fir_filter_mode_t;
+
+typedef struct fir_filter {
+    fir_filter_mode_t mode;
+    uint32_t channels;
+    uint32_t filter_length;
+    uint32_t buffer_size;
+    int16_t* coeffs;
+    int16_t* state;
+} fir_filter_t;
+
+fir_filter_t* fir_init(uint32_t channels, fir_filter_mode_t mode, uint32_t filter_length,
+                       uint32_t input_length, int16_t* coeffs);
+void fir_release(fir_filter_t* fir);
+void fir_reset(fir_filter_t* fir);
+void fir_process_interleaved(fir_filter_t* fir, int16_t* input, int16_t* output, uint32_t samples);
+
+#endif /* #ifndef FIR_FILTER_H */
diff --git a/device-common.mk b/device-common.mk
index de39035..88d4221 100644
--- a/device-common.mk
+++ b/device-common.mk
@@ -119,6 +119,7 @@
 # audio policy configuration
 USE_XML_AUDIO_POLICY_CONF := 1
 PRODUCT_COPY_FILES += \
+    $(LOCAL_PATH)/etc/mixer_paths.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths.xml \
     $(LOCAL_PATH)/etc/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml \
     $(LOCAL_PATH)/etc/audio_policy_configuration_bluetooth_legacy_hal.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration_bluetooth_legacy_hal.xml \
     frameworks/av/services/audiopolicy/config/a2dp_audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/a2dp_audio_policy_configuration.xml \
diff --git a/etc/audio_policy_configuration.xml b/etc/audio_policy_configuration.xml
index c94718e..fec99cc 100644
--- a/etc/audio_policy_configuration.xml
+++ b/etc/audio_policy_configuration.xml
@@ -1,5 +1,5 @@
 <?xml version="1.0" encoding="UTF-8" standalone="yes"?>
-<!-- Copyright (C) 2015 The Android Open Source Project
+<!-- Copyright (C) 2019 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.
@@ -15,15 +15,40 @@
 -->
 
 <audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+    <!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” -->
 
-    <globalConfiguration speaker_drc_enabled="false"/>
+    <!-- Global configuration Decalaration -->
+    <globalConfiguration speaker_drc_enabled="true"/>
 
+
+    <!-- Modules section:
+        There is one section per audio HW module present on the platform.
+        Each module section will contains two mandatory tags for audio HAL “halVersion” and “name”.
+        The module names are the same as in current .conf file:
+                “primary”, “A2DP”, “remote_submix”, “USB”
+        Each module will contain the following sections:
+        “devicePorts”: a list of device descriptors for all input and output devices accessible via this
+        module.
+        This contains both permanently attached devices and removable devices.
+        “mixPorts”: listing all output and input streams exposed by the audio HAL
+        “routes”: list of possible connections between input and output devices or between stream and
+        devices.
+            "route": is defined by an attribute:
+                -"type": <mux|mix> means all sources are mutual exclusive (mux) or can be mixed (mix)
+                -"sink": the sink involved in this route
+                -"sources": all the sources than can be connected to the sink via vis route
+        “attachedDevices”: permanently attached devices.
+        The attachedDevices section is a list of devices names. The names correspond to device names
+        defined in <devicePorts> section.
+        “defaultOutputDevice”: device to be used by default when no policy rule applies
+    -->
     <modules>
         <!-- Primary Audio HAL -->
-        <module name="primary" halVersion="2.0">
+        <module name="primary" halVersion="3.0">
             <attachedDevices>
                 <item>Speaker</item>
                 <item>Built-In Mic</item>
+                <item>Echo Reference</item>
             </attachedDevices>
             <defaultOutputDevice>Speaker</defaultOutputDevice>
             <mixPorts>
@@ -36,10 +61,17 @@
                              samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
                              channelMasks="AUDIO_CHANNEL_IN_MONO"/>
                 </mixPort>
+                <mixPort name="echo reference" role="sink">
+                    <profile name="echo_reference" format="AUDIO_FORMAT_PCM_32_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
             </mixPorts>
             <devicePorts>
                 <!-- Output devices declaration, i.e. Sink DEVICE PORT -->
-                <devicePort tagName="Speaker" role="sink" type="AUDIO_DEVICE_OUT_SPEAKER">
+                <devicePort tagName="Speaker" role="sink" type="AUDIO_DEVICE_OUT_SPEAKER" address="">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
                 </devicePort>
                 <devicePort tagName="Wired Headset" type="AUDIO_DEVICE_OUT_WIRED_HEADSET" role="sink">
                 </devicePort>
@@ -49,10 +81,6 @@
                 </devicePort>
                 <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
                 </devicePort>
-                <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
-                </devicePort>
-                <devicePort tagName="Aux Digital" type="AUDIO_DEVICE_OUT_AUX_DIGITAL" role="sink">
-                </devicePort>
 
                 <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
                 </devicePort>
@@ -60,6 +88,11 @@
                 </devicePort>
                 <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
                 </devicePort>
+                <devicePort tagName="Echo Reference" type="AUDIO_DEVICE_IN_ECHO_REFERENCE" role="source">
+                    <profile name="echo_reference" format="AUDIO_FORMAT_PCM_32_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+                </devicePort>
             </devicePorts>
             <!-- route declaration, i.e. list all available sources for a given sink -->
             <routes>
@@ -69,16 +102,14 @@
                        sources="primary output"/>
                 <route type="mix" sink="Wired Headphones"
                        sources="primary output"/>
-                <route type="mix" sink="Aux Digital"
-                       sources="primary output"/>
                 <route type="mix" sink="BT SCO"
                        sources="primary output"/>
                 <route type="mix" sink="BT SCO Headset"
                        sources="primary output"/>
-                <route type="mix" sink="BT SCO Car Kit"
-                       sources="primary output"/>
                 <route type="mix" sink="primary input"
                        sources="Built-In Mic,Wired Headset Mic,BT SCO Headset Mic"/>
+                <route type="mix" sink="echo reference"
+                       sources="Echo Reference"/>
             </routes>
 
         </module>
diff --git a/etc/mixer_paths.xml b/etc/mixer_paths.xml
new file mode 100644
index 0000000..dc28741
--- /dev/null
+++ b/etc/mixer_paths.xml
@@ -0,0 +1,5 @@
+<?xml version="1.0" encoding="ISO-8859-1"?>
+<mixer>
+	<!-- Enable HDMI out -->
+	<ctl name="QUAT_MI2S_RX Audio Mixer MultiMedia1" value="1" />
+</mixer>
diff --git a/qcom/init.qcom.rc b/qcom/init.qcom.rc
index 94e9038..f97b502 100644
--- a/qcom/init.qcom.rc
+++ b/qcom/init.qcom.rc
@@ -32,31 +32,6 @@
 
 on post-fs
     start pd_mapper
-    exec - root -- /system/bin/sleep 1
-    exec - system audio -- /system/bin/tinymix "QUAT_MI2S_RX Audio Mixer MultiMedia1" 1
-    exec - system audio -- /system/bin/tinymix "QUAT_MI2S_RX Audio Mixer MultiMedia2" 1
-    exec - system audio -- /system/bin/tinymix "SLIM RX0 MUX" AIF1_PB
-    exec - system audio -- /system/bin/tinymix "SLIM RX0 MUX" ZERO
-    exec - system audio -- /system/bin/tinymix "SLIM RX1 MUX" ZERO
-    exec - system audio -- /system/bin/tinymix "SLIM RX2 MUX" ZERO
-    exec - system audio -- /system/bin/tinymix "SLIM RX3 MUX" ZERO
-    exec - system audio -- /system/bin/tinymix "SLIM RX4 MUX" ZERO
-    exec - system audio -- /system/bin/tinymix "SLIM RX5 MUX" ZERO
-    exec - system audio -- /system/bin/tinymix "SLIM RX6 MUX" AIF1_PB
-    exec - system audio -- /system/bin/tinymix "SLIM RX7 MUX" AIF1_PB
-    exec - system audio -- /system/bin/tinymix "RX INT7_1 MIX1 INP0" RX6
-    exec - system audio -- /system/bin/tinymix "RX INT8_1 MIX1 INP0" RX7
-    exec - system audio -- /system/bin/tinymix "COMP7 Switch" 1
-    exec - system audio -- /system/bin/tinymix "COMP8 Switch" 1
-    exec - system audio -- /system/bin/tinymix "SpkrLeft COMP Switch" 1
-    exec - system audio -- /system/bin/tinymix "SpkrLeft BOOST Switch" 1
-    exec - system audio -- /system/bin/tinymix "SpkrLeft VISENSE Switch" 0
-    exec - system audio -- /system/bin/tinymix "SpkrLeft DAC Switch" 1
-    exec - system audio -- /system/bin/tinymix "SpkrRight COMP Switch" 1
-    exec - system audio -- /system/bin/tinymix "SpkrRight BOOST Switch" 1
-    exec - system audio -- /system/bin/tinymix "SpkrRight VISENSE Switch" 0
-    exec - system audio -- /system/bin/tinymix "SpkrRight DAC Switch" 1
-    exec - system audio -- /system/bin/tinymix "SLIMBUS_0_RX Audio Mixer MultiMedia2" 1
 
 on post-fs-data
     mkdir /data/vendor