1 files changed, 1770 insertions, 0 deletions

diff --git a/audio/primary/audio_hw.c b/audio/primary/audio_hw.c
new file mode 100644
index 0000000..95f0f06
--- /dev/null
+++ b/audio/primary/audio_hw.c
@@ -0,0 +1,1770 @@
+/*
+ * Copyright (C) 2013 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 "audio_hw_primary"
+//#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_voice_stream {
+    struct j6_audio_device *dev;
+    struct pcm *pcm_in;
+    struct pcm *pcm_out;
+    struct pcm_config in_config;
+    struct pcm_config out_config;
+    struct resampler_itfe *resampler;
+    struct resampler_buffer_provider buf_provider;
+    struct buffer_remix *remix;
+    pthread_t thread;
+    int16_t *in_buffer;
+    int16_t *out_buffer;
+    size_t in_frames;
+    size_t out_frames;
+    size_t in_frame_size;
+    size_t out_frame_size;
+    char *name;
+};
+
+struct j6_voice {
+    struct j6_voice_stream ul;
+    struct j6_voice_stream dl;
+};
+
+struct j6_audio_device {
+    struct audio_hw_device device;
+    struct j6_stream_in *in;
+    struct j6_stream_out *out;
+    struct j6_voice voice;
+    struct audio_route *route;
+    audio_devices_t in_device;
+    audio_devices_t out_device;
+    pthread_mutex_t lock;
+    unsigned int card;
+    unsigned int in_port;
+    unsigned int out_port;
+    unsigned int bt_card;
+    unsigned int bt_port;
+    bool mic_mute;
+    bool in_call;
+    audio_mode_t mode;
+};
+
+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;
+};
+
+struct j6_stream_out {
+    struct audio_stream_out stream;
+    struct j6_audio_device *dev;
+    struct pcm_config config;
+    struct pcm *pcm;
+    struct timespec last;
+    pthread_mutex_t lock;
+    bool standby;
+    int64_t written; /* total frames written, not cleared when entering standby */
+};
+
+
+static const char *supported_media_cards[] = {
+    "dra7evm",
+    "VayuEVM",
+    "DRA7xxEVM",
+};
+
+static const char *supported_bt_cards[] = {
+    "DRA7xxWiLink",
+};
+
+#define SUPPORTED_IN_DEVICES           (AUDIO_DEVICE_IN_BUILTIN_MIC | \
+                                        AUDIO_DEVICE_IN_WIRED_HEADSET | \
+                                        AUDIO_DEVICE_IN_DEFAULT)
+#define SUPPORTED_OUT_DEVICES          (AUDIO_DEVICE_OUT_SPEAKER | \
+                                        AUDIO_DEVICE_OUT_WIRED_HEADSET | \
+                                        AUDIO_DEVICE_OUT_WIRED_HEADPHONE | \
+                                        AUDIO_DEVICE_OUT_DEFAULT)
+
+#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 PLAYBACK_SAMPLE_RATE            44100
+#define PLAYBACK_PERIOD_SIZE            256
+#define PLAYBACK_PERIOD_COUNT           4
+#define PLAYBACK_BUFFER_SIZE            (PLAYBACK_PERIOD_SIZE * PLAYBACK_PERIOD_COUNT)
+
+#define BT_SAMPLE_RATE                  8000
+#define BT_PERIOD_SIZE                  80
+#define BT_PERIOD_COUNT                 4
+#define BT_BUFFER_SIZE                  (BT_PERIOD_SIZE * BT_PERIOD_COUNT)
+
+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,
+};
+
+struct pcm_config pcm_config_playback = {
+    .channels        = 2,
+    .rate            = PLAYBACK_SAMPLE_RATE,
+    .format          = PCM_FORMAT_S16_LE,
+    .period_size     = PLAYBACK_PERIOD_SIZE,
+    .period_count    = PLAYBACK_PERIOD_COUNT,
+    .start_threshold = PLAYBACK_BUFFER_SIZE / 2,
+    .stop_threshold  = PLAYBACK_BUFFER_SIZE,
+    .avail_min       = PLAYBACK_PERIOD_SIZE,
+};
+
+struct pcm_config pcm_config_bt_in = {
+    .channels        = 1,
+    .rate            = BT_SAMPLE_RATE,
+    .format          = PCM_FORMAT_S16_LE,
+    .period_size     = BT_PERIOD_SIZE,
+    .period_count    = BT_PERIOD_COUNT,
+};
+
+struct pcm_config pcm_config_bt_out = {
+    .channels        = 1,
+    .rate            = BT_SAMPLE_RATE,
+    .format          = PCM_FORMAT_S16_LE,
+    .period_size     = BT_PERIOD_SIZE,
+    .period_count    = BT_PERIOD_COUNT,
+};
+
+static void do_out_standby(struct j6_stream_out *out);
+
+/* 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");
+}
+
+/* must be called with device lock held */
+static void select_output_device(struct j6_audio_device *adev)
+{
+    if (adev->out_device & ~SUPPORTED_OUT_DEVICES)
+        ALOGW("select_output_device() device(s) not supported, will use default devices");
+}
+
+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;
+    }
+}
+
+static int setup_voice_remix(struct j6_voice_stream *stream)
+{
+    ALOGVV("setup_voice_mono_to_stereo_input_remix() %s stream", stream->name);
+
+    struct buffer_remix *br = (struct buffer_remix *)calloc(1, sizeof(struct buffer_remix));
+    if (!br)
+        return -ENOMEM;
+
+    br->remix_func = duplicate_channels_from_mono;
+    br->sample_size = sizeof(int16_t);
+    br->in_chans = stream->in_config.channels;
+    br->out_chans = stream->out_config.channels;
+
+    if (br->in_chans == 1)
+        br->remix_func = duplicate_channels_from_mono;
+    else if (br->out_chans == 1)
+        br->remix_func = remove_channels_from_buf;
+    else
+        return -ENOTSUP;
+
+    stream->remix = br;
+
+    return 0;
+}
+
+static int voice_get_next_buffer(struct resampler_buffer_provider *buffer_provider,
+                                 struct resampler_buffer* buffer)
+{
+    struct j6_voice_stream *stream;
+    int ret;
+
+    if (buffer_provider == NULL || buffer == NULL) {
+        ALOGE("voice_get_next_buffer() invalid buffer/provider");
+        return -EINVAL;
+    }
+
+    stream = (struct j6_voice_stream *)((char *)buffer_provider -
+                     offsetof(struct j6_voice_stream, buf_provider));
+
+    if (stream->pcm_in == NULL) {
+        buffer->raw = NULL;
+        buffer->frame_count = 0;
+        return -ENODEV;
+    }
+
+    if (buffer->frame_count > stream->in_frames) {
+        ALOGW("voice_get_next_buffer() %s unexpected frame count %u, "
+              "buffer was allocated for %u frames",
+              stream->name, buffer->frame_count, stream->in_frames);
+        buffer->frame_count = stream->in_frames;
+    }
+
+    ret = pcm_read(stream->pcm_in, stream->in_buffer,
+                   buffer->frame_count * stream->in_frame_size);
+    if (ret) {
+        ALOGE("voice_get_next_buffer() failed to read %s: %s",
+              stream->name, pcm_get_error(stream->pcm_in));
+        buffer->raw = NULL;
+        buffer->frame_count = 0;
+        return ret;
+    }
+
+    buffer->i16 = stream->in_buffer;
+
+    return ret;
+}
+
+static void voice_release_buffer(struct resampler_buffer_provider *buffer_provider,
+                                 struct resampler_buffer* buffer)
+{
+    UNUSED(buffer_provider);
+    UNUSED(buffer);
+}
+
+static void *voice_thread_func(void *arg)
+{
+    struct j6_voice_stream *stream = (struct j6_voice_stream *)arg;
+    struct j6_audio_device *adev = stream->dev;
+    size_t frames = stream->out_frames;
+    uint32_t periods = 0;
+    uint32_t avail;
+    bool in_steady = false;
+    bool out_steady = false;
+    int ret = 0;
+
+    pcm_start(stream->pcm_in);
+
+    memset(stream->out_buffer, 0, stream->out_frames * stream->out_frame_size);
+
+    while (adev->in_call) {
+        if (out_steady) {
+            if (in_steady) {
+                stream->resampler->resample_from_provider(stream->resampler,
+                                                          stream->out_buffer,
+                                                          &frames);
+            } else {
+                avail = pcm_avail_update(stream->pcm_in);
+                if (avail > 0) {
+                    in_steady = true;
+                    continue;
+                }
+            }
+        } else if (++periods == stream->out_config.period_count) {
+            out_steady = true;
+        }
+
+        if (stream->remix)
+            stream->remix->remix_func(stream->remix, stream->out_buffer, frames);
+
+        ret = pcm_write(stream->pcm_out, stream->out_buffer,
+                        frames * stream->out_frame_size);
+        if (ret) {
+            ALOGE("voice_thread_func() failed to write %s: %s",
+                  stream->name, pcm_get_error(stream->pcm_out));
+            usleep((frames * 1000000) / stream->out_config.rate);
+        }
+    }
+
+    return (void*)ret;
+}
+
+static void voice_stream_exit(struct j6_voice_stream *stream)
+{
+    if (stream->resampler) {
+        release_resampler(stream->resampler);
+        stream->resampler = NULL;
+    }
+
+    if (stream->pcm_out) {
+        pcm_close(stream->pcm_out);
+        stream->pcm_out = NULL;
+    }
+
+    if (stream->pcm_in) {
+        pcm_close(stream->pcm_in);
+        stream->pcm_in = NULL;
+    }
+
+    if (stream->in_buffer) {
+        free(stream->in_buffer);
+        stream->in_buffer = NULL;
+        stream->in_frames = 0;
+    }
+
+    if (stream->out_buffer) {
+        free(stream->out_buffer);
+        stream->out_buffer = NULL;
+        stream->out_frames = 0;
+    }
+
+    if (stream->remix) {
+        free(stream->remix);
+        stream->remix = NULL;
+    }
+
+    if (stream->name) {
+        free(stream->name);
+        stream->name = NULL;
+    }
+}
+
+static int voice_stream_init(struct j6_voice_stream *stream,
+                             unsigned int in_card,
+                             unsigned int in_port,
+                             unsigned int out_card,
+                             unsigned int out_port)
+{
+    struct j6_audio_device *adev = stream->dev;
+    int frames;
+    int ret;
+
+    stream->buf_provider.get_next_buffer = voice_get_next_buffer;
+    stream->buf_provider.release_buffer = voice_release_buffer;
+    ret = create_resampler(stream->in_config.rate,
+                           stream->out_config.rate,
+                           stream->in_config.channels,
+                           RESAMPLER_QUALITY_DEFAULT,
+                           &stream->buf_provider,
+                           &stream->resampler);
+    if (ret) {
+        ALOGE("voice_stream_init() failed to create %s resampler %d", stream->name, ret);
+        return ret;
+    }
+
+    stream->pcm_in = pcm_open(in_card, in_port,
+                              PCM_IN | PCM_MONOTONIC,
+                              &stream->in_config);
+    stream->pcm_out = pcm_open(out_card, out_port,
+                               PCM_OUT | PCM_MONOTONIC,
+                               &stream->out_config);
+
+    if (!pcm_is_ready(stream->pcm_in) || !pcm_is_ready(stream->pcm_out)) {
+        ALOGE("voice_stream_init() failed to open pcm %s devices", stream->name);
+        voice_stream_exit(stream);
+        return -ENODEV;
+    }
+
+    /*
+     * Choose the largest frame size since mono-to-stereo and
+     * stereo-to-mono conversion happens in place
+     */
+    stream->in_frame_size = pcm_frames_to_bytes(stream->pcm_in, 1);
+    stream->out_frame_size = pcm_frames_to_bytes(stream->pcm_out, 1);
+    frames = (stream->in_frame_size > stream->out_frame_size) ?
+                stream->in_frame_size : stream->out_frame_size;
+
+    /* out_buffer will store the resampled data */
+    stream->out_frames = stream->out_config.period_size;
+    stream->out_buffer = malloc(stream->out_frames * frames);
+
+    /* in_buffer will store the frames recorded from the PCM device */
+    stream->in_frames = (stream->out_frames * stream->in_config.rate) / stream->out_config.rate +
+                        RESAMPLER_HEADROOM_FRAMES;
+    stream->in_buffer = malloc(stream->in_frames * frames);
+
+    if (!stream->in_buffer || !stream->out_buffer) {
+        ALOGE("voice_stream_init() failed to allocate %s buffers", stream->name);
+        voice_stream_exit(stream);
+        return -ENOMEM;
+    }
+
+    ret = setup_voice_remix(stream);
+    if (ret) {
+        ALOGE("voice_stream_init() failed to setup remix %d", ret);
+        voice_stream_exit(stream);
+    }
+
+    return ret;
+}
+
+static int enter_voice_call(struct j6_audio_device *adev)
+{
+    struct j6_voice *voice = &adev->voice;
+    int ret;
+
+    ALOGI("enter_voice_call() entering bluetooth voice call");
+
+    /* Let the primary output switch to a dummy sink */
+    if (adev->out)
+        do_out_standby(adev->out);
+
+    /* Uplink: Mic (44.1kHz) -> BT (8kHz) */
+    voice->ul.name = strdup("UL");
+    voice->ul.in_config = pcm_config_capture;
+    voice->ul.out_config = pcm_config_bt_out;
+    voice->ul.dev = adev;
+    ret = voice_stream_init(&voice->ul, adev->card, adev->in_port,
+                            adev->bt_card, adev->bt_port);
+    if (ret) {
+        ALOGE("enter_voice_call() failed to init uplink %d", ret);
+        return ret;
+    }
+
+    /* Downlink: BT (8kHz) -> HP/Spk (44.1kHz) */
+    voice->dl.name = strdup("DL");
+    voice->dl.in_config = pcm_config_bt_in;
+    voice->dl.out_config = pcm_config_playback;
+    voice->dl.dev = adev;
+    ret = voice_stream_init(&voice->dl, adev->bt_card, adev->bt_port,
+                            adev->card, adev->out_port);
+    if (ret) {
+        ALOGE("enter_voice_call() failed to init downlink %d", ret);
+        goto err_dl_init;
+    }
+
+    adev->in_call = true;
+
+    /* Create uplink thread: Mic -> BT */
+    ret = pthread_create(&voice->ul.thread, NULL, voice_thread_func, &voice->ul);
+    if (ret) {
+        ALOGE("enter_voice_call() failed to create uplink thread %d", ret);
+        adev->in_call = false;
+        goto err_ul_thread;
+    }
+
+    /* Create downlink thread: BT -> HP/Spk */
+    ret = pthread_create(&voice->dl.thread, NULL, voice_thread_func, &voice->dl);
+    if (ret) {
+        ALOGE("enter_voice_call() failed to create downlink thread %d", ret);
+        adev->in_call = false;
+        goto err_dl_thread;
+    }
+
+    return 0;
+
+ err_dl_thread:
+    pthread_join(voice->ul.thread, NULL);
+ err_ul_thread:
+    voice_stream_exit(&voice->ul);
+ err_dl_init:
+    voice_stream_exit(&voice->dl);
+
+    return ret;
+}
+
+static void leave_voice_call(struct j6_audio_device *adev)
+{
+    struct j6_voice *voice = &adev->voice;
+    struct j6_voice_stream *ul = &voice->ul;
+    struct j6_voice_stream *dl = &voice->dl;
+    void *ret;
+
+    ALOGI("leave_voice_call() leaving bluetooth voice call");
+
+    adev->in_call = false;
+
+    if (ul->pcm_out)
+        pcm_stop(ul->pcm_out);
+    if (dl->pcm_in)
+        pcm_stop(dl->pcm_in);
+
+    pthread_join(voice->dl.thread, &ret);
+    pthread_join(voice->ul.thread, &ret);
+
+    voice_stream_exit(&voice->dl);
+    voice_stream_exit(&voice->ul);
+
+    /* Let the primary output switch back to its ALSA PCM device */
+    if (adev->out)
+        do_out_standby(adev->out);
+}
+
+static uint32_t time_diff(struct timespec t1, struct timespec t0)
+{
+    struct timespec temp;
+
+    if ((t1.tv_nsec - t0.tv_nsec) < 0) {
+        temp.tv_sec = t1.tv_sec - t0.tv_sec-1;
+        temp.tv_nsec = 1000000000UL + t1.tv_nsec - t0.tv_nsec;
+    } else {
+        temp.tv_sec = t1.tv_sec - t0.tv_sec;
+        temp.tv_nsec = t1.tv_nsec - t0.tv_nsec;
+    }
+
+    return (temp.tv_sec * 1000000UL + temp.tv_nsec / 1000);
+}
+
+/* audio HAL functions */
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    uint32_t rate = PLAYBACK_SAMPLE_RATE;
+
+    UNUSED(stream);
+    ALOGVV("out_get_sample_rate() stream=%p rate=%u", stream, rate);
+
+    return rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    UNUSED(stream);
+    UNUSED(rate);
+    ALOGVV("out_set_sample_rate() stream=%p rate=%u", stream, rate);
+
+    return 0;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    const struct audio_stream_out *s = (const struct audio_stream_out *)stream;
+    uint32_t frames = ((PLAYBACK_PERIOD_SIZE + 15) / 16) * 16;
+    size_t bytes = frames * audio_stream_out_frame_size(s);
+
+    ALOGVV("out_get_buffer_size() stream=%p frames=%u bytes=%u", stream, frames, bytes);
+
+    return bytes;
+}
+
+static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
+{
+    audio_channel_mask_t channels = AUDIO_CHANNEL_OUT_STEREO;
+
+    UNUSED(stream);
+    ALOGVV("out_get_channels() stream=%p channels=%u", stream, popcount(channels));
+
+    return channels;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+    audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
+
+    UNUSED(stream);
+    ALOGVV("out_set_format() stream=%p format=0x%08x (%u bits/sample)",
+           stream, format, audio_bytes_per_sample(format) << 3);
+
+    return format;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    UNUSED(stream);
+    ALOGVV("out_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_out_standby(struct j6_stream_out *out)
+{
+    struct j6_audio_device *adev = out->dev;
+
+    if (!out->standby) {
+        if (adev->mode != AUDIO_MODE_IN_CALL) {
+            ALOGI("do_out_standby() close card %u port %u", adev->card, adev->out_port);
+            pcm_close(out->pcm);
+            out->pcm = NULL;
+        } else {
+            ALOGI("do_out_standby() close dummy card");
+        }
+        out->standby = true;
+    }
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    struct j6_stream_out *out = (struct j6_stream_out *)(stream);
+    struct j6_audio_device *adev = out->dev;
+
+    ALOGV("out_standby() stream=%p", out);
+    pthread_mutex_lock(&adev->lock);
+    pthread_mutex_lock(&out->lock);
+    do_out_standby(out);
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&adev->lock);
+
+    return 0;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    UNUSED(stream);
+    UNUSED(fd);
+
+    return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    struct j6_stream_out *out = (struct j6_stream_out *)(stream);
+    struct j6_audio_device *adev = out->dev;
+    struct str_parms *parms;
+    char value[32];
+    int ret;
+    uint32_t val = 0;
+
+    ALOGV("out_set_parameters() stream=%p parameter='%s'", out, kvpairs);
+
+    parms = str_parms_create_str(kvpairs);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        val = atoi(value);
+        pthread_mutex_lock(&adev->lock);
+        pthread_mutex_lock(&out->lock);
+        if (val != 0) {
+            if ((adev->out_device & AUDIO_DEVICE_OUT_ALL) != val)
+                do_out_standby(out);
+
+            /* set the active output device */
+            adev->out_device = val;
+            select_output_device(adev);
+        }
+        pthread_mutex_unlock(&out->lock);
+        pthread_mutex_unlock(&adev->lock);
+    }
+
+    return 0;
+}
+
+static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    UNUSED(stream);
+    UNUSED(keys);
+
+    return strdup("");
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    const struct j6_stream_out *out = (const struct j6_stream_out *)(stream);
+    uint32_t frames = PLAYBACK_BUFFER_SIZE;
+    uint32_t latency = (frames * 1000) / PLAYBACK_SAMPLE_RATE;
+
+    ALOGVV("out_get_latency() stream=%p latency=%u msecs", out, latency);
+
+    return latency;
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    UNUSED(stream);
+    UNUSED(left);
+    UNUSED(right);
+
+    return -ENOSYS;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
+                         size_t bytes)
+{
+    const struct audio_stream_out *s = (const struct audio_stream_out *)stream;
+    struct j6_stream_out *out = (struct j6_stream_out *)(stream);
+    struct j6_audio_device *adev = out->dev;
+    struct timespec now;
+    const size_t frame_size = audio_stream_out_frame_size(s);
+    const size_t frames = bytes / frame_size;
+    uint32_t rate = out->config.rate;
+    uint32_t write_usecs = frames * 1000000 / rate;
+    uint32_t diff_usecs;
+    int ret = 0;
+
+    ALOGVV("out_write() stream=%p buffer=%p size=%u/%u time=%u usecs",
+           out, buffer, frames, rate, write_usecs);
+
+    pthread_mutex_lock(&adev->lock);
+    pthread_mutex_lock(&out->lock);
+
+    if (out->standby) {
+        if (!adev->in_call) {
+            select_output_device(adev);
+
+            ALOGI("out_write() open card %u port %u", adev->card, adev->out_port);
+            out->pcm = pcm_open(adev->card, adev->out_port,
+                                PCM_OUT | PCM_MONOTONIC,
+                                &out->config);
+            if (!pcm_is_ready(out->pcm)) {
+                ALOGE("out_write() failed to open pcm out: %s", pcm_get_error(out->pcm));
+                pcm_close(out->pcm);
+                out->pcm = NULL;
+                ret = -ENODEV;
+            }
+        } else {
+            ALOGI("out_write() open dummy port");
+            clock_gettime(CLOCK_REALTIME, &out->last);
+        }
+
+        if (ret) {
+            usleep(write_usecs); /* limits the rate of error messages */
+            pthread_mutex_unlock(&out->lock);
+            pthread_mutex_unlock(&adev->lock);
+            return ret;
+        }
+
+        out->standby = false;
+    }
+
+    pthread_mutex_unlock(&adev->lock);
+
+    if (!adev->in_call) {
+        ret = pcm_write(out->pcm, buffer, bytes);
+        if (ret) {
+            ALOGE("out_write() failed to write audio data %d", ret);
+            usleep(write_usecs); /* limits the rate of error messages */
+        }
+    } else {
+        clock_gettime(CLOCK_REALTIME, &now);
+        diff_usecs = time_diff(now, out->last);
+        if (write_usecs > diff_usecs)
+            usleep(write_usecs - diff_usecs);
+
+        clock_gettime(CLOCK_REALTIME, &out->last);
+    }
+
+    out->written += frames;
+
+    pthread_mutex_unlock(&out->lock);
+
+    return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    UNUSED(stream);
+    UNUSED(dsp_frames);
+
+    return -EINVAL;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    UNUSED(stream);
+    UNUSED(effect);
+
+    return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    UNUSED(stream);
+    UNUSED(effect);
+
+    return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
+                                        int64_t *timestamp)
+{
+    UNUSED(stream);
+    UNUSED(timestamp);
+
+    return -EINVAL;
+}
+
+static int out_get_presentation_position(const struct audio_stream_out *stream,
+                                         uint64_t *frames, struct timespec *timestamp)
+{
+    struct j6_stream_out *out = (struct j6_stream_out *)(stream);
+    struct j6_audio_device *adev = out->dev;
+    int64_t signed_frames = -1;
+    size_t avail;
+    int ret = -1;
+
+    pthread_mutex_lock(&out->lock);
+
+    if (!adev->in_call) {
+        if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
+            signed_frames = out->written - pcm_get_buffer_size(out->pcm) + avail;
+        }
+    } else {
+        clock_gettime(CLOCK_MONOTONIC, timestamp);
+        signed_frames = out->written +
+            (time_diff(*timestamp, out->last) * out->config.rate) / 1000000;
+    }
+
+    /* It would be unusual for this value to be negative, but check just in case ... */
+    if (signed_frames >= 0) {
+        *frames = signed_frames;
+        ret = 0;
+    }
+
+    pthread_mutex_unlock(&out->lock);
+
+    return ret;
+}
+
+/** 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->out_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;
+
+        if  (in->resampler) {
+            in->resampler->resample_from_provider(in->resampler,
+                    (int16_t *)((char *)buffer + frames_wr * frame_size),
+                    &frames_rd);
+        } else {
+            struct resampler_buffer buf = {
+                    { raw : NULL, },
+                    frame_count : frames_rd,
+            };
+            get_next_buffer(&in->buf_provider, &buf);
+            if (buf.raw) {
+                memcpy((char *)buffer + frames_wr * frame_size,
+                       buf.raw,
+                       buf.frame_count * frame_size);
+                frames_rd = buf.frame_count;
+            }
+            release_buffer(&in->buf_provider, &buf);
+        }
+
+        /* in->read_status is updated by getNextBuffer() also called by
+         * in->resampler->resample_from_provider() */
+        if (in->read_status != 0)
+            return in->read_status;
+
+        frames_wr += frames_rd;
+    }
+
+    return frames_wr;
+}
+
+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;
+    }
+
+    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 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)
+{
+    struct j6_audio_device *adev = (struct j6_audio_device *)dev;
+    struct j6_stream_out *out;
+
+    UNUSED(handle);
+    UNUSED(devices);
+    UNUSED(address);
+
+    out = (struct j6_stream_out *)calloc(1, sizeof(struct j6_stream_out));
+    if (!out)
+        return -ENOMEM;
+
+    ALOGV("adev_open_output_stream() stream=%p rate=%u channels=%u "
+          "format=0x%08x flags=0x%08x",
+          out, config->sample_rate, popcount(config->channel_mask),
+          config->format, flags);
+
+    pthread_mutex_init(&out->lock, NULL);
+
+    out->stream.common.get_sample_rate = out_get_sample_rate;
+    out->stream.common.set_sample_rate = out_set_sample_rate;
+    out->stream.common.get_buffer_size = out_get_buffer_size;
+    out->stream.common.get_channels = out_get_channels;
+    out->stream.common.get_format = out_get_format;
+    out->stream.common.set_format = out_set_format;
+    out->stream.common.standby = out_standby;
+    out->stream.common.dump = out_dump;
+    out->stream.common.set_parameters = out_set_parameters;
+    out->stream.common.get_parameters = out_get_parameters;
+    out->stream.common.add_audio_effect = out_add_audio_effect;
+    out->stream.common.remove_audio_effect = out_remove_audio_effect;
+    out->stream.get_latency = out_get_latency;
+    out->stream.set_volume = out_set_volume;
+    out->stream.write = out_write;
+    out->stream.get_render_position = out_get_render_position;
+    out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
+    out->stream.get_presentation_position = out_get_presentation_position;
+
+    out->dev = adev;
+    out->standby = true;
+    out->config = pcm_config_playback;
+    out->written = 0;
+    adev->out = out;
+
+    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;
+
+    return 0;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct j6_audio_device *adev = (struct j6_audio_device *)dev;
+    struct j6_stream_out *out = (struct j6_stream_out *)(stream);
+
+    ALOGV("adev_close_output_stream() stream=%p", out);
+
+    out_standby(&stream->common);
+    out->dev = NULL;
+    adev->out = NULL;
+
+    free(stream);
+}
+
+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)
+{
+    struct j6_audio_device *adev = (struct j6_audio_device *)dev;
+    struct j6_stream_out *out = adev->out;
+    int ret = 0;
+
+    ALOGV("adev_set_mode() mode=0x%08x", mode);
+
+    pthread_mutex_lock(&adev->lock);
+    pthread_mutex_lock(&out->lock);
+
+    if (adev->mode == mode) {
+        ALOGV("adev_set_mode() already in mode=0x%08x", mode);
+        goto out;
+    }
+
+    if (mode == AUDIO_MODE_IN_CALL) {
+        ret = enter_voice_call(adev);
+        if (ret) {
+            ALOGE("adev_set_mode() failed to initialize voice call %d", ret);
+            goto out;
+        }
+    } else if (adev->mode == AUDIO_MODE_IN_CALL) {
+        leave_voice_call(adev);
+    }
+
+    adev->mode = mode;
+
+out:
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&adev->lock);
+
+    return ret;
+}
+
+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 buffer_size;
+    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;
+
+    ALOGV("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");
+        ret = setup_stereo_to_mono_input_remix(in);
+        if (ret) {
+            ALOGE("adev_open_input_stream() failed to setup remix %d", ret);
+            goto err1;
+        }
+    }
+
+    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 err2;
+        }
+    }
+
+    /*
+     * buffer size needs to be enough to allow stereo-to-mono remix
+     * and resample if needed
+     */
+    if (in->resampler || in->remix) {
+        buffer_size = in->config.period_size * in->hw_frame_size;
+        if (in->resampler)
+            buffer_size *= 2;
+        if (in->remix)
+            buffer_size *= 2;
+
+        in->buffer = malloc(buffer_size);
+        if (!in->buffer) {
+            ret = -ENOMEM;
+            goto err3;
+        }
+    }
+
+    *stream_in = &in->stream;
+
+    return 0;
+
+ err3:
+    release_resampler(in->resampler);
+ err2:
+    free(in->remix);
+ 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);
+
+    if (in->remix)
+        free(in->remix);
+
+    free(in->buffer);
+    free(in);
+    adev->in = NULL;
+}
+
+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->out_device = AUDIO_DEVICE_OUT_SPEAKER;
+    adev->card = 0;
+    adev->in_port = 0;
+    adev->out_port = 0;
+    ALOGI("Media card is hw:%d\n", adev->card);
+    adev->bt_card=2;    
+    adev->bt_port = 0;
+    ALOGI("Bluetooth SCO card is hw:%d\n", adev->bt_card);
+
+    adev->mic_mute = false;
+    adev->in_call = false;
+    adev->mode = AUDIO_MODE_NORMAL;
+
+    adev->route = audio_route_init(adev->card, NULL);
+    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 Audio HAL",
+        .author = "Texas Instruments Inc.",
+        .methods = &hal_module_methods,
+    },
+};