audio: Combine the JAMR3 HAL into the primary HAL

JAMR3 line-in audio support was provided through a separate audio HAL
which caused significant code duplication.  Line-in audio support is
now combined into the primary HAL.

Change-Id: Ibd1d481954f1ebb71bb543c5a9d8dbd3ba61b002
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>

6 files changed, 48 insertions, 1030 deletions

diff --git a/audio/jamr3/Android.mk b/audio/jamr3/Android.mk
deleted file mode 100644
index 277745c..0000000
--- a/audio/jamr3/Android.mk
+++ /dev/null
@@ -1,43 +0,0 @@
-# Copyright (C) 2015 Texas Instruments
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#      http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-LOCAL_PATH := $(call my-dir)
-
-ifeq ($(TARGET_BOARD_PLATFORM), $(filter $(TARGET_BOARD_PLATFORM), jacinto6))
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := audio.jamr3.$(TARGET_BOARD_PLATFORM)
-
-LOCAL_MODULE_PATH := $(TARGET_OUT_VENDOR_SHARED_LIBRARIES)/hw
-LOCAL_SRC_FILES := jamr3_audio_hw.c
-
-LOCAL_C_INCLUDES += \
-        external/tinyalsa/include \
-        system/media/audio_route/include \
-        system/media/audio_utils/include \
-        system/media/audio_effects/include
-
-LOCAL_SHARED_LIBRARIES := \
-        liblog \
-        libcutils \
-        libtinyalsa \
-        libaudioutils \
-        libaudioroute
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
-
-endif
diff --git a/audio/jamr3/jamr3_audio_hw.c b/audio/jamr3/jamr3_audio_hw.c
deleted file mode 100644
index c7b5258..0000000
--- a/audio/jamr3/jamr3_audio_hw.c
+++ /dev/null
@@ -1,970 +0,0 @@
-/*
- * Copyright (C) 2015 Texas Instruments
- *
- * 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 "jamr3_audio_hw"
-//#define LOG_NDEBUG 0
-//#define VERY_VERBOSE_LOGGING
-#ifdef VERY_VERBOSE_LOGGING
-#define ALOGVV ALOGV
-#else
-#define ALOGVV(a...) do { } while(0)
-#endif
-
-#include <errno.h>
-#include <stdint.h>
-#include <stdlib.h>
-#include <sys/time.h>
-
-#include <cutils/log.h>
-#include <cutils/str_parms.h>
-#include <cutils/properties.h>
-
-#include <audio_utils/resampler.h>
-#include <audio_route/audio_route.h>
-#include <system/audio.h>
-#include <hardware/hardware.h>
-#include <hardware/audio.h>
-#include <hardware/audio_effect.h>
-
-#include <tinyalsa/asoundlib.h>
-
-#define UNUSED(x) (void)(x)
-
-/* yet another definition of ARRAY_SIZE macro) */
-#define ARRAY_SIZE(x)           (sizeof(x)/sizeof(x[0]))
-
-/*
- * additional space in resampler buffer allowing for extra samples to be returned
- * by speex resampler when sample rates ratio is not an integer
- */
-#define RESAMPLER_HEADROOM_FRAMES   10
-
-/* buffer_remix: functor for doing in-place buffer manipulations.
- *
- * NB. When remix_func is called, the memory at `buf` must be at least
- * as large as frames * sample_size * MAX(in_chans, out_chans).
- */
-struct buffer_remix {
-    void (*remix_func)(struct buffer_remix *data, void *buf, size_t frames);
-    size_t sample_size; /* size of one audio sample, in bytes */
-    size_t in_chans;    /* number of input channels */
-    size_t out_chans;   /* number of output channels */
-};
-
-struct j6_audio_device {
-    struct audio_hw_device device;
-    struct j6_stream_in *in;
-    struct audio_route *route;
-    audio_devices_t in_device;
-    pthread_mutex_t lock;
-    unsigned int card;
-    unsigned int in_port;
-    bool mic_mute;
-};
-
-struct j6_stream_in {
-    struct audio_stream_in stream;
-    struct j6_audio_device *dev;
-    struct pcm_config config;
-    struct pcm *pcm;
-    struct buffer_remix *remix; /* adapt hw chan count to client */
-    struct resampler_itfe *resampler;
-    struct resampler_buffer_provider buf_provider;
-    int16_t *buffer;
-    size_t frames_in;
-    size_t hw_frame_size;
-    unsigned int requested_rate;
-    unsigned int requested_channels;
-    int read_status;
-    pthread_mutex_t lock;
-    bool standby;
-    int read; /* total frames read, cleared when entering standby */
-};
-
-static const char *supported_cards[] = {
-    "DRA7xx-JAMR3",
-};
-
-#define MAX_CARD_COUNT                  10
-
-#define SUPPORTED_IN_DEVICES            AUDIO_DEVICE_IN_LINE
-
-#define CAPTURE_SAMPLE_RATE             44100
-#define CAPTURE_PERIOD_SIZE             256
-#define CAPTURE_PERIOD_COUNT            4
-#define CAPTURE_BUFFER_SIZE             (CAPTURE_PERIOD_SIZE * CAPTURE_PERIOD_COUNT)
-
-#define MIXER_XML_PATH                  "/vendor/etc/jamr3_mixer_paths.xml"
-
-struct pcm_config pcm_config_capture = {
-    .channels        = 2,
-    .rate            = CAPTURE_SAMPLE_RATE,
-    .format          = PCM_FORMAT_S16_LE,
-    .period_size     = CAPTURE_PERIOD_SIZE,
-    .period_count    = CAPTURE_PERIOD_COUNT,
-    .start_threshold = 1,
-    .stop_threshold  = CAPTURE_BUFFER_SIZE,
-};
-
-static int find_card_index(const char *supported_cards[], int num_supported)
-{
-    struct mixer *mixer;
-    const char *name;
-    int card = 0;
-    int found = 0;
-    int i;
-
-    do {
-        /* returns an error after last valid card */
-        mixer = mixer_open(card);
-        if (!mixer)
-            break;
-
-        name = mixer_get_name(mixer);
-
-        for (i = 0; i < num_supported; ++i) {
-            if (supported_cards[i] && !strcmp(name, supported_cards[i])) {
-                ALOGV("Supported card '%s' found at %d", name, card);
-                found = 1;
-                break;
-            }
-        }
-
-        mixer_close(mixer);
-    } while (!found && (card++ < MAX_CARD_COUNT));
-
-    /* Use default card number if not found */
-    if (!found)
-        card = 2;
-
-    return card;
-}
-
-/* must be called with device lock held */
-static void select_input_device(struct j6_audio_device *adev)
-{
-    if (adev->in_device & ~SUPPORTED_IN_DEVICES)
-        ALOGW("select_input_device() device not supported, will use default device");
-}
-
-static size_t get_input_buffer_size(uint32_t sample_rate, int format, int channel_count)
-{
-    size_t size;
-
-    /*
-     * take resampling into account and return the closest majoring
-     * multiple of 16 frames, as audioflinger expects audio buffers to
-     * be a multiple of 16 frames
-     */
-    size = (pcm_config_capture.period_size * sample_rate) / pcm_config_capture.rate;
-    size = ((size + 15) / 16) * 16;
-
-    return size * channel_count * audio_bytes_per_sample(format);
-}
-
-/*
- * Implementation of buffer_remix::remix_func that removes
- * channels in place without doing any other processing.  The
- * extra channels are truncated.
- */
-static void remove_channels_from_buf(struct buffer_remix *data, void *buf, size_t frames)
-{
-    size_t samp_size, in_frame, out_frame;
-    size_t N, c;
-    char *s, *d;
-
-    ALOGVV("remove_channels_from_buf() remix=%p buf=%p frames=%u",
-           data, buf, frames);
-
-    if (frames == 0)
-        return;
-
-    samp_size = data->sample_size;
-    in_frame = data->in_chans * samp_size;
-    out_frame = data->out_chans * samp_size;
-
-    if (out_frame >= in_frame) {
-        ALOGE("BUG: remove_channels_from_buf() can not add channels to a buffer.\n");
-        return;
-    }
-
-    N = frames - 1;
-    d = (char*)buf + out_frame;
-    s = (char*)buf + in_frame;
-
-    /* take the first several channels and truncate the rest */
-    while (N--) {
-        for (c = 0; c < out_frame; ++c)
-            d[c] = s[c];
-        d += out_frame;
-        s += in_frame;
-    }
-}
-
-static int setup_stereo_to_mono_input_remix(struct j6_stream_in *in)
-{
-    ALOGV("setup_stereo_to_mono_input_remix() stream=%p", in);
-
-    struct buffer_remix *br = (struct buffer_remix *)calloc(1, sizeof(struct buffer_remix));
-    if (!br)
-        return -ENOMEM;
-
-    br->remix_func = remove_channels_from_buf;
-    br->sample_size = sizeof(int16_t);
-    br->in_chans = 2;
-    br->out_chans = 1;
-    in->remix = br;
-
-    return 0;
-}
-
-/*
- * Implementation of buffer_remix::remix_func that duplicates the first
- * channel into the rest of channels in the frame without doing any other
- * processing
- */
-static void duplicate_channels_from_mono(struct buffer_remix *data, void *buf, size_t frames)
-{
-    int samp_size, in_frame, out_frame;
-    int N, c;
-    char *s, *d;
-
-    ALOGVV("duplicate_channels_from_mono() remix=%p buf=%p frames=%u",
-           data, buf, frames);
-
-    if (frames == 0)
-        return;
-
-    samp_size = data->sample_size;
-    in_frame = data->in_chans * samp_size;
-    out_frame = data->out_chans * samp_size;
-
-    if (in_frame >= out_frame) {
-        ALOGE("BUG: duplicate_channels_from_mono() can not drop channels\n");
-        return;
-    }
-
-    N = frames - 1;
-    d = (char*)buf + N * out_frame;
-    s = (char*)buf + N * in_frame;
-
-    /* duplicate first channel into the rest of channels in the frame */
-    while (N-- >= 0) {
-        for (c = 0; c < out_frame; ++c)
-            d[c] = s[c % in_frame];
-        d -= out_frame;
-        s -= in_frame;
-    }
-}
-
-/** audio_stream_in implementation **/
-static uint32_t in_get_sample_rate(const struct audio_stream *stream)
-{
-    const struct j6_stream_in *in = (const struct j6_stream_in *)(stream);
-
-    ALOGVV("in_get_sample_rate() stream=%p rate=%u", stream, in->requested_rate);
-
-    return in->requested_rate;
-}
-
-static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
-{
-    ALOGV("in_set_sample_rate() stream=%p rate=%u", stream, rate);
-
-    return 0;
-}
-
-static size_t in_get_buffer_size(const struct audio_stream *stream)
-{
-    const struct j6_stream_in *in = (const struct j6_stream_in *)(stream);
-
-    size_t bytes = get_input_buffer_size(in->requested_rate,
-                                         AUDIO_FORMAT_PCM_16_BIT,
-                                         in->requested_channels);
-
-    ALOGVV("in_get_buffer_size() stream=%p bytes=%u", in, bytes);
-
-    return bytes;
-}
-
-static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
-{
-    const struct j6_stream_in *in = (const struct j6_stream_in *)(stream);
-    audio_channel_mask_t channels = audio_channel_in_mask_from_count(in->requested_channels);
-
-    ALOGVV("in_get_channels() stream=%p channels=%u", in, in->requested_channels);
-
-    return channels;
-}
-
-static audio_format_t in_get_format(const struct audio_stream *stream)
-{
-    audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
-
-    UNUSED(stream);
-    ALOGVV("in_set_format() stream=%p format=0x%08x (%u bits/sample)",
-           stream, format, audio_bytes_per_sample(format) << 3);
-
-    return format;
-}
-
-static int in_set_format(struct audio_stream *stream, audio_format_t format)
-{
-    UNUSED(stream);
-    ALOGV("in_set_format() stream=%p format=0x%08x (%u bits/sample)",
-          stream, format, audio_bytes_per_sample(format) << 3);
-
-    if (format != AUDIO_FORMAT_PCM_16_BIT) {
-        return -ENOSYS;
-    } else {
-        return 0;
-    }
-}
-
-/* must be called with locks held */
-static void do_in_standby(struct j6_stream_in *in)
-{
-    struct j6_audio_device *adev = in->dev;
-
-    if (!in->standby) {
-        ALOGI("do_in_standby() close card %u port %u", adev->card, adev->in_port);
-        pcm_close(in->pcm);
-        in->pcm = NULL;
-        in->standby = true;
-    }
-}
-
-static int in_standby(struct audio_stream *stream)
-{
-    struct j6_stream_in *in = (struct j6_stream_in *)(stream);
-    struct j6_audio_device *adev = in->dev;
-
-    ALOGV("in_standby() stream=%p", in);
-    pthread_mutex_lock(&adev->lock);
-    pthread_mutex_lock(&in->lock);
-    do_in_standby(in);
-    pthread_mutex_unlock(&in->lock);
-    pthread_mutex_unlock(&adev->lock);
-
-    return 0;
-}
-
-static int in_dump(const struct audio_stream *stream, int fd)
-{
-    UNUSED(stream);
-    UNUSED(fd);
-
-    return 0;
-}
-
-static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
-{
-    struct j6_stream_in *in = (struct j6_stream_in *)(stream);
-    struct j6_audio_device *adev = in->dev;
-    struct str_parms *parms;
-    char value[32];
-    int ret;
-    uint32_t val = 0;
-
-    ALOGV("in_set_parameters() stream=%p parameter='%s'", stream, kvpairs);
-
-    parms = str_parms_create_str(kvpairs);
-
-    /* Nothing to do for AUDIO_PARAMETER_STREAM_INPUT_SOURCE, so it's ignored */
-
-    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(&in->lock);
-        if (val != 0) {
-            if ((adev->in_device & AUDIO_DEVICE_IN_ALL) != val)
-                do_in_standby(in);
-
-            /* set the active input device */
-            adev->in_device = val;
-            select_input_device(adev);
-        }
-        pthread_mutex_unlock(&in->lock);
-        pthread_mutex_unlock(&adev->lock);
-    }
-
-    return 0;
-}
-
-static char * in_get_parameters(const struct audio_stream *stream,
-                                const char *keys)
-{
-    UNUSED(stream);
-    UNUSED(keys);
-
-    return strdup("");
-}
-
-static int in_set_gain(struct audio_stream_in *stream, float gain)
-{
-    UNUSED(stream);
-    UNUSED(gain);
-
-    return 0;
-}
-
-static int get_next_buffer(struct resampler_buffer_provider *buffer_provider,
-                           struct resampler_buffer* buffer)
-{
-    struct j6_stream_in *in;
-    struct buffer_remix *remix;
-
-    if (buffer_provider == NULL || buffer == NULL)
-        return -EINVAL;
-
-    in = (struct j6_stream_in *)((char *)buffer_provider -
-                                 offsetof(struct j6_stream_in, buf_provider));
-
-    if (in->pcm == NULL) {
-        buffer->raw = NULL;
-        buffer->frame_count = 0;
-        in->read_status = -ENODEV;
-        return -ENODEV;
-    }
-
-    if (in->frames_in == 0) {
-        in->read_status = pcm_read(in->pcm,
-                                   (void*)in->buffer,
-                                   buffer->frame_count * in->hw_frame_size);
-        if (in->read_status != 0) {
-            ALOGE("get_next_buffer() pcm_read error %d", in->read_status);
-            buffer->raw = NULL;
-            buffer->frame_count = 0;
-            return in->read_status;
-        }
-        in->frames_in = buffer->frame_count;
-
-        remix = in->remix;
-        if (remix)
-            remix->remix_func(remix, in->buffer, in->frames_in);
-    }
-
-    buffer->frame_count = (buffer->frame_count > in->frames_in) ?
-                                in->frames_in : buffer->frame_count;
-    buffer->i16 = in->buffer;
-
-    return in->read_status;
-}
-
-static void release_buffer(struct resampler_buffer_provider *buffer_provider,
-                                  struct resampler_buffer* buffer)
-{
-    struct j6_stream_in *in;
-
-    if (buffer_provider == NULL || buffer == NULL)
-        return;
-
-    in = (struct j6_stream_in *)((char *)buffer_provider -
-                                 offsetof(struct j6_stream_in, buf_provider));
-
-    in->frames_in -= buffer->frame_count;
-}
-
-/*
- * read_frames() reads frames from kernel driver, down samples to capture rate
- * if necessary and output the number of frames requested to the buffer specified
- */
-static ssize_t read_frames(struct j6_stream_in *in, void *buffer, ssize_t frames)
-{
-    const struct audio_stream_in *s = (const struct audio_stream_in *)in;
-    ssize_t frames_wr = 0;
-    size_t frame_size;
-
-    ALOGVV("read_frames() stream=%p frames=%u", in, frames);
-
-    if (in->remix)
-        frame_size = audio_stream_in_frame_size(s);
-    else
-        frame_size = in->hw_frame_size;
-
-    while (frames_wr < frames) {
-        size_t frames_rd = frames - frames_wr;
-
-        in->resampler->resample_from_provider(in->resampler,
-                    (int16_t *)((char *)buffer + frames_wr * frame_size),
-                    &frames_rd);
-        /* in->read_status is updated by getNextBuffer() also called by
-         * in->resampler->resample_from_provider() */
-        if (in->read_status != 0)
-            return in->read_status;
-
-        frames_wr += frames_rd;
-    }
-
-    return frames_wr;
-}
-
-static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
-                       size_t bytes)
-{
-    const struct audio_stream_in *s = (const struct audio_stream_in *)stream;
-    struct j6_stream_in *in = (struct j6_stream_in *)(stream);
-    struct j6_audio_device *adev = in->dev;
-    const size_t frame_size = audio_stream_in_frame_size(stream);
-    const size_t frames = bytes / frame_size;
-    uint32_t rate = in_get_sample_rate(&stream->common);
-    uint32_t read_usecs = frames * 1000000 / rate;
-    int ret;
-
-    ALOGVV("in_read() stream=%p buffer=%p size=%u/%u time=%u usecs",
-           stream, buffer, frames, rate, read_usecs);
-
-    pthread_mutex_lock(&adev->lock);
-    pthread_mutex_lock(&in->lock);
-
-    if (in->standby) {
-        select_input_device(adev);
-
-        ALOGI("in_read() open card %u port %u", adev->card, adev->in_port);
-        in->pcm = pcm_open(adev->card, adev->in_port,
-                           PCM_IN | PCM_MONOTONIC,
-                           &in->config);
-        if (!pcm_is_ready(in->pcm)) {
-            ALOGE("in_read() failed to open pcm in: %s", pcm_get_error(in->pcm));
-            pcm_close(in->pcm);
-            in->pcm = NULL;
-            usleep(read_usecs); /* limits the rate of error messages */
-            pthread_mutex_unlock(&in->lock);
-            pthread_mutex_unlock(&adev->lock);
-            return -ENODEV;
-        }
-
-        /* if no supported sample rate is available, use the resampler */
-        if (in->resampler) {
-            in->resampler->reset(in->resampler);
-            in->frames_in = 0;
-        }
-
-        in->standby = false;
-        in->read = 0;
-    }
-
-    pthread_mutex_unlock(&adev->lock);
-
-    if (in->resampler || in->remix)
-        ret = read_frames(in, buffer, frames);
-    else
-        ret = pcm_read(in->pcm, buffer, bytes);
-
-    if (ret < 0) {
-        ALOGE("in_read() failed to read audio data %d", ret);
-        usleep(read_usecs); /* limits the rate of error messages */
-        memset(buffer, 0, bytes);
-    } else {
-        in->read += frames;
-        if (adev->mic_mute)
-            memset(buffer, 0, bytes);
-    }
-
-    pthread_mutex_unlock(&in->lock);
-
-    return bytes;
-}
-
-static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
-{
-    UNUSED(stream);
-    ALOGVV("in_get_input_frames_lost() stream=%p frames=%u", stream, 0);
-
-    return 0;
-}
-
-static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
-{
-    UNUSED(stream);
-    UNUSED(effect);
-
-    return 0;
-}
-
-static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
-{
-    UNUSED(stream);
-    UNUSED(effect);
-
-    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(dev);
-    UNUSED(handle);
-    UNUSED(devices);
-    UNUSED(flags);
-    UNUSED(config);
-    UNUSED(stream_out);
-    UNUSED(address);
-
-    ALOGE("adev_open_output_stream() output stream is not supported");
-
-    return -ENOSYS;
-}
-
-static void adev_close_output_stream(struct audio_hw_device *dev,
-                                     struct audio_stream_out *stream)
-{
-    UNUSED(dev);
-    UNUSED(stream);
-
-    ALOGE("adev_close_output_stream() output stream is not supported");
-}
-
-static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
-{
-    UNUSED(dev);
-    UNUSED(kvpairs);
-
-    return -ENOSYS;
-}
-
-static char * adev_get_parameters(const struct audio_hw_device *dev,
-                                  const char *keys)
-{
-    UNUSED(dev);
-    UNUSED(keys);
-
-    return strdup("");;
-}
-
-static int adev_init_check(const struct audio_hw_device *dev)
-{
-    UNUSED(dev);
-
-    return 0;
-}
-
-static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
-{
-    UNUSED(dev);
-    UNUSED(volume);
-
-    return -ENOSYS;
-}
-
-static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
-{
-    UNUSED(dev);
-    UNUSED(volume);
-
-    return -ENOSYS;
-}
-
-static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
-{
-    UNUSED(dev);
-    UNUSED(volume);
-
-    return -ENOSYS;
-}
-
-static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
-{
-    UNUSED(dev);
-    UNUSED(muted);
-
-    return -ENOSYS;
-}
-
-static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
-{
-    UNUSED(dev);
-    UNUSED(muted);
-
-    return -ENOSYS;
-}
-
-static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
-{
-    UNUSED(dev);
-    UNUSED(mode);
-
-    ALOGV("adev_set_mode() mode=0x%08x", mode);
-
-    return 0;
-}
-
-static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
-{
-    struct j6_audio_device *adev = (struct j6_audio_device *)dev;
-
-    ALOGV("adev_set_mic_mute() state=%s", state ? "mute" : "unmute");
-    adev->mic_mute = state;
-
-    return 0;
-}
-
-static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
-{
-    const struct j6_audio_device *adev = (const struct j6_audio_device *)dev;
-
-    *state = adev->mic_mute;
-    ALOGV("adev_get_mic_mute() state=%s", *state ? "mute" : "unmute");
-
-    return 0;
-}
-
-static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
-                                         const struct audio_config *config)
-{
-    UNUSED(dev);
-
-    size_t bytes = get_input_buffer_size(config->sample_rate,
-                                        config->format,
-                                        popcount(config->channel_mask));
-
-    ALOGVV("adev_in_get_buffer_size() bytes=%u", bytes);
-
-    return bytes;
-}
-
-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,
-                                  const char *address,
-                                  audio_source_t source)
-{
-    struct j6_audio_device *adev = (struct j6_audio_device *)dev;
-    struct j6_stream_in *in;
-    int ret;
-
-    UNUSED(handle);
-    UNUSED(devices);
-    UNUSED(flags);
-    UNUSED(address);
-    UNUSED(source);
-
-    in = (struct j6_stream_in *)calloc(1, sizeof(struct j6_stream_in));
-    if (!in)
-        return -ENOMEM;
-
-    ALOGI("adev_open_input_stream() stream=%p rate=%u channels=%u format=0x%08x",
-          in, config->sample_rate, popcount(config->channel_mask), config->format);
-
-    pthread_mutex_init(&in->lock, NULL);
-
-    in->stream.common.get_sample_rate = in_get_sample_rate;
-    in->stream.common.set_sample_rate = in_set_sample_rate;
-    in->stream.common.get_buffer_size = in_get_buffer_size;
-    in->stream.common.get_channels = in_get_channels;
-    in->stream.common.get_format = in_get_format;
-    in->stream.common.set_format = in_set_format;
-    in->stream.common.standby = in_standby;
-    in->stream.common.dump = in_dump;
-    in->stream.common.set_parameters = in_set_parameters;
-    in->stream.common.get_parameters = in_get_parameters;
-    in->stream.common.add_audio_effect = in_add_audio_effect;
-    in->stream.common.remove_audio_effect = in_remove_audio_effect;
-    in->stream.set_gain = in_set_gain;
-    in->stream.read = in_read;
-    in->stream.get_input_frames_lost = in_get_input_frames_lost;
-
-    in->dev = adev;
-    in->standby = true;
-    in->config = pcm_config_capture;
-    in->requested_rate = config->sample_rate;
-    in->requested_channels = popcount(config->channel_mask);
-    in->hw_frame_size = in->config.channels * sizeof(int16_t);
-    in->remix = NULL;
-    in->resampler = NULL;
-    in->buffer = NULL;
-    adev->in = in;
-
-    /* in-place stereo-to-mono remix since capture stream is stereo */
-    if (in->requested_channels == 1) {
-        ALOGV("adev_open_input_stream() stereo-to-mono remix needed");
-
-        /*
-         * buffer size is already enough to allow stereo-to-mono remix
-         * and resample if needed
-         */
-        in->buffer = malloc(2 * in->config.period_size * in->hw_frame_size);
-        if (!in->buffer) {
-            ret = -ENOMEM;
-            goto err1;
-        }
-
-        ret = setup_stereo_to_mono_input_remix(in);
-        if (ret) {
-            ALOGE("adev_open_input_stream() failed to setup remix %d", ret);
-            goto err2;
-        }
-    }
-
-    if (in->requested_rate != in->config.rate) {
-        ALOGV("adev_open_input_stream() resample needed, req=%uHz got=%uHz",
-              in->requested_rate, in->config.rate);
-
-        in->buf_provider.get_next_buffer = get_next_buffer;
-        in->buf_provider.release_buffer = release_buffer;
-        ret = create_resampler(in->config.rate,
-                               in->requested_rate,
-                               in->requested_channels,
-                               RESAMPLER_QUALITY_DEFAULT,
-                               &in->buf_provider,
-                               &in->resampler);
-        if (ret) {
-            ALOGE("adev_open_input_stream() failed to create resampler %d", ret);
-            goto err3;
-        }
-    }
-
-    *stream_in = &in->stream;
-
-    return 0;
-
- err3:
-    free(in->remix);
- err2:
-    free(in->buffer);
- err1:
-    free(in);
-    return ret;
-}
-
-static void adev_close_input_stream(struct audio_hw_device *dev,
-                                   struct audio_stream_in *stream)
-{
-    struct j6_audio_device *adev = (struct j6_audio_device *)dev;
-    struct j6_stream_in *in = (struct j6_stream_in *)(stream);
-
-    ALOGV("adev_close_input_stream() stream=%p", stream);
-
-    in_standby(&stream->common);
-
-    if (in->resampler)
-        release_resampler(in->resampler);
-    in->resampler = NULL;
-
-    if (in->remix)
-        free(in->remix);
-    in->remix = NULL;
-
-    in->dev = NULL;
-    adev->in = NULL;
-
-    free(in->buffer);
-    free(in);
-}
-
-static int adev_dump(const audio_hw_device_t *device, int fd)
-{
-    UNUSED(device);
-    UNUSED(fd);
-
-    return 0;
-}
-
-static int adev_close(hw_device_t *device)
-{
-    struct j6_audio_device *adev = (struct j6_audio_device *)device;
-
-    ALOGI("adev_close()");
-
-    audio_route_free(adev->route);
-    free(device);
-
-    return 0;
-}
-
-static int adev_open(const hw_module_t* module, const char* name,
-                     hw_device_t** device)
-{
-    struct j6_audio_device *adev;
-
-    ALOGI("adev_open() %s", name);
-
-    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
-        return -EINVAL;
-
-    adev = (struct j6_audio_device*)calloc(1, sizeof(struct j6_audio_device));
-    if (!adev)
-        return -ENOMEM;
-
-    pthread_mutex_init(&adev->lock, NULL);
-
-    adev->device.common.tag = HARDWARE_DEVICE_TAG;
-    adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
-    adev->device.common.module = (struct hw_module_t *) module;
-    adev->device.common.close = adev_close;
-
-    adev->device.init_check = adev_init_check;
-    adev->device.set_voice_volume = adev_set_voice_volume;
-    adev->device.set_master_volume = adev_set_master_volume;
-    adev->device.get_master_volume = adev_get_master_volume;
-    adev->device.set_master_mute = adev_set_master_mute;
-    adev->device.get_master_mute = adev_get_master_mute;
-    adev->device.set_mode = adev_set_mode;
-    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;
-    adev->device.get_parameters = adev_get_parameters;
-    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->in_device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-    adev->card = find_card_index(supported_cards,
-                                 ARRAY_SIZE(supported_cards));
-    adev->in_port = 0;
-    ALOGI("JAMR3 card is hw:%d\n", adev->card);
-
-    adev->mic_mute = false;
-
-    adev->route = audio_route_init(adev->card, MIXER_XML_PATH);
-    if (!adev->route) {
-        ALOGE("Unable to initialize audio routes");
-        free(adev);
-        return -EINVAL;
-    }
-
-    *device = &adev->device.common;
-
-    return 0;
-}
-
-static struct hw_module_methods_t hal_module_methods = {
-    .open = adev_open,
-};
-
-struct audio_module HAL_MODULE_INFO_SYM = {
-    .common = {
-        .tag = HARDWARE_MODULE_TAG,
-        .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
-        .hal_api_version = HARDWARE_HAL_API_VERSION,
-        .id = AUDIO_HARDWARE_MODULE_ID,
-        .name = "Jacinto6 JAMR3 Audio HAL",
-        .author = "Texas Instruments Inc.",
-        .methods = &hal_module_methods,
-    },
-};
diff --git a/audio/primary/audio_hw.c b/audio/primary/audio_hw.c
index 0ae1f54..13dca21 100644
--- a/audio/primary/audio_hw.c
+++ b/audio/primary/audio_hw.c
@@ -94,6 +94,7 @@ struct j6_audio_device {
     struct j6_stream_out *out;
     struct j6_voice voice;
     struct audio_route *route;
+    struct audio_route *jamr_route;
     audio_devices_t in_device;
     audio_devices_t out_device;
     pthread_mutex_t lock;
@@ -102,6 +103,8 @@ struct j6_audio_device {
     unsigned int out_port;
     unsigned int bt_card;
     unsigned int bt_port;
+    unsigned int jamr_card;
+    unsigned int jamr_port;
     bool mic_mute;
     bool in_call;
     audio_mode_t mode;
@@ -120,6 +123,8 @@ struct j6_stream_in {
     size_t hw_frame_size;
     unsigned int requested_rate;
     unsigned int requested_channels;
+    unsigned int card;
+    unsigned int port;
     int read_status;
     pthread_mutex_t lock;
     bool standby;
@@ -147,10 +152,15 @@ static const char *supported_bt_cards[] = {
     "DRA7xxWiLink",
 };
 
+static const char *supported_jamr_cards[] = {
+    "DRA7xx-JAMR3",
+};
+
 #define MAX_CARD_COUNT                  10
 
 #define SUPPORTED_IN_DEVICES           (AUDIO_DEVICE_IN_BUILTIN_MIC | \
                                         AUDIO_DEVICE_IN_WIRED_HEADSET | \
+                                        AUDIO_DEVICE_IN_LINE | \
                                         AUDIO_DEVICE_IN_DEFAULT)
 #define SUPPORTED_OUT_DEVICES          (AUDIO_DEVICE_OUT_SPEAKER | \
                                         AUDIO_DEVICE_OUT_WIRED_HEADSET | \
@@ -173,6 +183,7 @@ static const char *supported_bt_cards[] = {
 #define BT_BUFFER_SIZE                  (BT_PERIOD_SIZE * BT_PERIOD_COUNT)
 
 #define MIXER_XML_PATH                  "/vendor/etc/mixer_paths.xml"
+#define JAMR_MIXER_XML_PATH             "/vendor/etc/jamr3_mixer_paths.xml"
 
 struct pcm_config pcm_config_capture = {
     .channels        = 2,
@@ -1295,8 +1306,8 @@ static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
     if (in->standby) {
         select_input_device(adev);
 
-        ALOGI("in_read() open card %u port %u", adev->card, adev->in_port);
-        in->pcm = pcm_open(adev->card, adev->in_port,
+        ALOGI("in_read() open card %u port %u", in->card, in->port);
+        in->pcm = pcm_open(in->card, in->port,
                            PCM_IN | PCM_MONOTONIC,
                            &in->config);
         if (!pcm_is_ready(in->pcm)) {
@@ -1624,6 +1635,8 @@ static int adev_open_input_stream(struct audio_hw_device *dev,
     in->remix = NULL;
     in->resampler = NULL;
     in->buffer = NULL;
+    in->card = (devices == AUDIO_DEVICE_IN_LINE) ? adev->jamr_card : adev->card;
+    in->port = 0;
     adev->in = in;
 
     /* in-place stereo-to-mono remix since capture stream is stereo */
@@ -1721,6 +1734,7 @@ static int adev_close(hw_device_t *device)
     ALOGI("adev_close()");
 
     audio_route_free(adev->route);
+    audio_route_free(adev->jamr_route);
     free(device);
 
     return 0;
@@ -1778,6 +1792,11 @@ static int adev_open(const hw_module_t* module, const char* name,
     adev->bt_port = 0;
     ALOGI("Bluetooth SCO card is hw:%d\n", adev->bt_card);
 
+    adev->jamr_card = find_card_index(supported_jamr_cards,
+                                      ARRAY_SIZE(supported_jamr_cards));
+    adev->jamr_port = 0;
+    ALOGI("JAMR card is hw:%d\n", adev->jamr_card);
+
     adev->mic_mute = false;
     adev->in_call = false;
     adev->mode = AUDIO_MODE_NORMAL;
@@ -1785,13 +1804,24 @@ static int adev_open(const hw_module_t* module, const char* name,
     adev->route = audio_route_init(adev->card, MIXER_XML_PATH);
     if (!adev->route) {
         ALOGE("Unable to initialize audio routes");
-        free(adev);
-        return -EINVAL;
+        goto err1;
+    }
+
+    adev->jamr_route = audio_route_init(adev->jamr_card, JAMR_MIXER_XML_PATH);
+    if (!adev->jamr_route) {
+        ALOGE("Unable to initialize JAMR audio routes");
+        goto err2;
     }
 
     *device = &adev->device.common;
 
     return 0;
+
+ err2:
+    audio_route_free(adev->route);
+ err1:
+    free(adev);
+    return -ENODEV;
 }
 
 static struct hw_module_methods_t hal_module_methods = {
diff --git a/audio/primary/audio_policy_configuration.xml b/audio/primary/audio_policy_configuration.xml
index f13d1bb..c5fb06e 100644
--- a/audio/primary/audio_policy_configuration.xml
+++ b/audio/primary/audio_policy_configuration.xml
@@ -4,6 +4,7 @@
     <attachedDevices>
         <item>Speaker</item>
         <item>Built-In Mic</item>
+        <item>Line In</item>
     </attachedDevices>
 
     <defaultOutputDevice>Speaker</defaultOutputDevice>
@@ -19,7 +20,11 @@
                     channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
         </mixPort>
     </mixPorts>
-
+        <mixPort name="line input" role="sink">
+            <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                    samplingRates="8000,11025,12000,16000,22050,24000,32000,44100,48000"
+                    channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+        </mixPort>
     <devicePorts>
         <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
         </devicePort>
@@ -30,18 +35,17 @@
 
         <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
         </devicePort>
+        <devicePort tagName="Line In" type="AUDIO_DEVICE_IN_LINE" role="source">
+        </devicePort>
     </devicePorts>
 
     <routes>
-        <route type="mux" sink="Speaker"
-               sources="primary output"/>
-        <route type="mux" sink="Wired Headset"
-               sources="primary output"/>
-        <route type="mux" sink="Wired Headphones"
-               sources="primary output"/>
+        <route type="mux" sink="Speaker" sources="primary output"/>
+        <route type="mux" sink="Wired Headset" sources="primary output"/>
+        <route type="mux" sink="Wired Headphones" sources="primary output"/>
 
-        <route type="mix" sink="primary input"
-               sources="Built-In Mic"/>
+        <route type="mix" sink="primary input" sources="Built-In Mic"/>
+        <route type="mix" sink="line input" sources="Line In"/>
   </routes>
 </module>
 
diff --git a/audio/jamr3/jamr3_mixer_paths.xml b/audio/primary/jamr3_mixer_paths.xml
index 1277a32..1277a32 100644
--- a/audio/jamr3/jamr3_mixer_paths.xml
+++ b/audio/primary/jamr3_mixer_paths.xml
diff --git a/device.mk b/device.mk
index 69f4dfa..5a9d2f4 100644
--- a/device.mk
+++ b/device.mk
@@ -96,6 +96,7 @@ PRODUCT_COPY_FILES += \
 else
 PRODUCT_COPY_FILES += \
         device/ti/jacinto6evm/audio/primary/mixer_paths.xml:$(TARGET_COPY_OUT_VENDOR)/etc/mixer_paths.xml \
+        device/ti/jacinto6evm/audio/primary/jamr3_mixer_paths.xml:$(TARGET_COPY_OUT_VENDOR)/etc/jamr3_mixer_paths.xml \
         device/ti/jacinto6evm/audio/primary/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/primary_audio_policy_configuration.xml \
         device/ti/jacinto6evm/audio/audio_policy_configuration.xml:$(TARGET_COPY_OUT_VENDOR)/etc/audio_policy_configuration.xml
 endif
@@ -161,10 +162,6 @@ PRODUCT_PACKAGES += audio.hdmi.jacinto6
 PRODUCT_PACKAGES += audio.a2dp.default
 # Remote submix
 PRODUCT_PACKAGES += audio.r_submix.default
-# JAMR3 Audio HAL module
-ifneq ($(APPE_AUDIO),true)
-PRODUCT_PACKAGES += audio.jamr3.jacinto6
-endif
 
 PRODUCT_PACKAGES += \
         tinymix \