diff --git a/modules/usbaudio/audio_hw.c b/modules/usbaudio/audio_hw.c
index 24a2d63..afe56b2 100644
--- a/modules/usbaudio/audio_hw.c
+++ b/modules/usbaudio/audio_hw.c
@@ -33,65 +33,270 @@
 
 #include <tinyalsa/asoundlib.h>
 
-struct pcm_config pcm_config = {
+/* This is the default configuration to hand to The Framework on the initial
+ * adev_open_output_stream(). Actual device attributes will be used on the subsequent
+ * adev_open_output_stream() after the card and device number have been set in out_set_parameters()
+ */
+#define OUT_PERIOD_SIZE 1024
+#define OUT_PERIOD_COUNT 4
+#define OUT_SAMPLING_RATE 44100
+
+struct pcm_config default_alsa_out_config = {
     .channels = 2,
-    .rate = 44100,
-    .period_size = 1024,
-    .period_count = 4,
+    .rate = OUT_SAMPLING_RATE,
+    .period_size = OUT_PERIOD_SIZE,
+    .period_count = OUT_PERIOD_COUNT,
     .format = PCM_FORMAT_S16_LE,
 };
 
+/*
+ * Input defaults.  See comment above.
+ */
+#define IN_PERIOD_SIZE 1024
+#define IN_PERIOD_COUNT 4
+#define IN_SAMPLING_RATE 44100
+
+struct pcm_config default_alsa_in_config = {
+    .channels = 2,
+    .rate = IN_SAMPLING_RATE,
+    .period_size = IN_PERIOD_SIZE,
+    .period_count = IN_PERIOD_COUNT,
+    .format = PCM_FORMAT_S16_LE,
+    .start_threshold = 1,
+    .stop_threshold = (IN_PERIOD_SIZE * IN_PERIOD_COUNT),
+};
+
 struct audio_device {
     struct audio_hw_device hw_device;
 
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
-    int card;
-    int device;
+
+    /* output */
+    int out_card;
+    int out_device;
+
+    /* input */
+    int in_card;
+    int in_device;
+
     bool standby;
 };
 
 struct stream_out {
     struct audio_stream_out stream;
 
+    pthread_mutex_t lock;               /* see note below on mutex acquisition order */
+    struct pcm *pcm;                    /* state of the stream */
+    bool standby;
+
+    struct audio_device *dev;           /* hardware information */
+
+    void * conversion_buffer;           /* any conversions are put into here
+                                         * they could come from here too if
+                                         * there was a previous conversion */
+    size_t conversion_buffer_size;      /* in bytes */
+};
+
+/*
+ * Output Configuration Cache
+ * FIXME(pmclean) This is not rentrant. Should probably be moved into the stream structure
+ * but that will involve changes in The Framework.
+ */
+static struct pcm_config cached_output_hardware_config;
+static bool output_hardware_config_is_cached = false;
+
+struct stream_in {
+    struct audio_stream_in stream;
+
     pthread_mutex_t lock; /* see note below on mutex acquisition order */
     struct pcm *pcm;
     bool standby;
 
+    struct pcm_config alsa_pcm_config;
+
     struct audio_device *dev;
+
+    struct audio_config hal_pcm_config;
+
+    unsigned int requested_rate;
+//    struct resampler_itfe *resampler;
+//    struct resampler_buffer_provider buf_provider;
+    int16_t *buffer;
+    size_t buffer_size;
+    size_t frames_in;
+    int read_status;
 };
 
+/*
+ * Utility
+ */
+/*
+ * Translates from ALSA format ID to ANDROID_AUDIO_CORE format ID
+ * (see master/system/core/include/core/audio.h)
+ * TODO(pmclean) Replace with audio_format_from_pcm_format() (in hardware/audio_alsaops.h).
+ *   post-integration.
+ */
+static audio_format_t alsa_to_fw_format_id(int alsa_fmt_id)
+{
+    switch (alsa_fmt_id) {
+    case PCM_FORMAT_S8:
+        return AUDIO_FORMAT_PCM_8_BIT;
+
+    case PCM_FORMAT_S24_3LE:
+        //TODO(pmclean) make sure this is the 'right' sort of 24-bit
+        return AUDIO_FORMAT_PCM_8_24_BIT;
+
+    case PCM_FORMAT_S32_LE:
+    case PCM_FORMAT_S24_LE:
+        return AUDIO_FORMAT_PCM_32_BIT;
+    }
+
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+/*
+ * Data Conversions
+ */
+/*
+ * Convert a buffer of PCM16LE samples to packed (3-byte) PCM24LE samples.
+ *   in_buff points to the buffer of PCM16 samples
+ *   num_in_samples size of input buffer in SAMPLES
+ *   out_buff points to the buffer to receive converted PCM24 LE samples.
+ *   returns the number of BYTES of output data.
+ * We are doing this since we *always* present to The Framework as A PCM16LE device, but need to
+ * support PCM24_3LE (24-bit, packed).
+ * NOTE: we're just filling the low-order byte of the PCM24LE samples with 0.
+ * TODO(pmclean, hung) Move this to a utilities module.
+ */
+static size_t convert_16_to_24_3(unsigned short * in_buff,
+                                 size_t num_in_samples,
+                                 unsigned char * out_buff) {
+    /*
+     * Move from back to front so that the conversion can be done in-place
+     * i.e. in_buff == out_buff
+     */
+    int in_buff_size_in_bytes = num_in_samples * 2;
+    /* we need 3 bytes in the output for every 2 bytes in the input */
+    int out_buff_size_in_bytes = ((3 * in_buff_size_in_bytes) / 2);
+    unsigned char* dst_ptr = out_buff + out_buff_size_in_bytes - 1;
+    int src_smpl_index;
+    unsigned char* src_ptr = ((unsigned char *)in_buff) + in_buff_size_in_bytes - 1;
+    for (src_smpl_index = 0; src_smpl_index < num_in_samples; src_smpl_index++) {
+        *dst_ptr-- = *src_ptr--; /* hi-byte */
+        *dst_ptr-- = *src_ptr--; /* low-byte */
+        *dst_ptr-- = 0;          /* zero-byte */
+    }
+
+    /* return number of *bytes* generated */
+    return out_buff_size_in_bytes;
+}
+
+/*
+ * Convert a buffer of 2-channel PCM16 samples to 4-channel PCM16 channels
+ *   in_buff points to the buffer of PCM16 samples
+ *   num_in_samples size of input buffer in SAMPLES
+ *   out_buff points to the buffer to receive converted PCM16 samples.
+ *   returns the number of BYTES of output data.
+ * NOTE channels 3 & 4 are filled with silence.
+ * We are doing this since we *always* present to The Framework as STEREO device, but need to
+ * support 4-channel devices.
+ * TODO(pmclean, hung) Move this to a utilities module.
+ */
+static size_t convert_2chan16_to_4chan16(unsigned short* in_buff,
+                                          size_t num_in_samples,
+                                          unsigned short* out_buff) {
+    /*
+     * Move from back to front so that the conversion can be done in-place
+     * i.e. in_buff == out_buff
+     */
+    int out_buff_size = num_in_samples * 2;
+    unsigned short* dst_ptr = out_buff + out_buff_size - 1;
+    int src_index;
+    unsigned short* src_ptr = in_buff + num_in_samples - 1;
+    for (src_index = 0; src_index < num_in_samples; src_index += 2) {
+        *dst_ptr-- = 0;          /* chan 4 */
+        *dst_ptr-- = 0;          /* chan 3 */
+        *dst_ptr-- = *src_ptr--; /* chan 2 */
+        *dst_ptr-- = *src_ptr--; /* chan 1 */
+    }
+
+    /* return number of *bytes* generated */
+    return out_buff_size * 2;
+}
+
+/*
+ * ALSA Utilities
+ */
+/*
+ * gets the ALSA bit-format flag from a bits-per-sample value.
+ * TODO(pmclean, hung) Move this to a utilities module.
+ */
+static int bits_to_alsa_format(int bits_per_sample, int default_format)
+{
+    enum pcm_format format;
+    for (format = PCM_FORMAT_S16_LE; format < PCM_FORMAT_MAX; format++) {
+        if (pcm_format_to_bits(format) == bits_per_sample) {
+            return  format;
+         }
+    }
+    return default_format;
+}
+
+/*
+ * Reads and decodes configuration info from the specified ALSA card/device
+ */
+static int read_alsa_device_config(int card, int device, int io_type, struct pcm_config * config)
+{
+    ALOGV("usb:audio_hw - read_alsa_device_config(card:%d device:%d)", card, device);
+
+    if (card < 0 || device < 0) {
+        return -EINVAL;
+    }
+
+    struct pcm_params * alsa_hw_params = pcm_params_get(card, device, io_type);
+    if (alsa_hw_params == NULL) {
+        return -EINVAL;
+    }
+
+    /*
+     * This Logging will be useful when testing new USB devices.
+     */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_SAMPLE_BITS min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS), pcm_params_get_max(alsa_hw_params, PCM_PARAM_SAMPLE_BITS)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_FRAME_BITS min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_FRAME_BITS), pcm_params_get_max(alsa_hw_params, PCM_PARAM_FRAME_BITS)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_CHANNELS min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS), pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_RATE min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE), pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_PERIOD_TIME min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIOD_TIME), pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIOD_TIME)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_PERIOD_SIZE min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIOD_SIZE), pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIOD_SIZE)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_PERIOD_BYTES min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIOD_BYTES), pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIOD_BYTES)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_PERIODS min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIODS), pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIODS)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_BUFFER_TIME min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_BUFFER_TIME), pcm_params_get_max(alsa_hw_params, PCM_PARAM_BUFFER_TIME)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_BUFFER_SIZE min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_BUFFER_SIZE), pcm_params_get_max(alsa_hw_params, PCM_PARAM_BUFFER_SIZE)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_BUFFER_BYTES min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_BUFFER_BYTES), pcm_params_get_max(alsa_hw_params, PCM_PARAM_BUFFER_BYTES)); */
+    /* ALOGV("usb:audio_hw - PCM_PARAM_TICK_TIME min:%d, max:%d", pcm_params_get_min(alsa_hw_params, PCM_PARAM_TICK_TIME), pcm_params_get_max(alsa_hw_params, PCM_PARAM_TICK_TIME)); */
+
+    config->channels = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+    config->rate = pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE);
+    config->period_size = pcm_params_get_max(alsa_hw_params, PCM_PARAM_PERIODS);
+    config->period_count = pcm_params_get_min(alsa_hw_params, PCM_PARAM_PERIODS);
+
+    int bits_per_sample = pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+    config->format = bits_to_alsa_format(bits_per_sample, PCM_FORMAT_S16_LE);
+
+    return 0;
+}
+
+/*
+ * HAl Functions
+ */
 /**
  * NOTE: when multiple mutexes have to be acquired, always respect the
  * following order: hw device > out stream
  */
 
 /* Helper functions */
-
-/* must be called with hw device and output stream mutexes locked */
-static int start_output_stream(struct stream_out *out)
-{
-    struct audio_device *adev = out->dev;
-    int i;
-
-    if ((adev->card < 0) || (adev->device < 0))
-        return -EINVAL;
-
-    out->pcm = pcm_open(adev->card, adev->device, PCM_OUT, &pcm_config);
-
-    if (out->pcm && !pcm_is_ready(out->pcm)) {
-        ALOGE("pcm_open() failed: %s", pcm_get_error(out->pcm));
-        pcm_close(out->pcm);
-        return -ENOMEM;
-    }
-
-    return 0;
-}
-
-/* API functions */
-
 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
 {
-    return pcm_config.rate;
+    return cached_output_hardware_config.rate;
 }
 
 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
@@ -101,17 +306,22 @@
 
 static size_t out_get_buffer_size(const struct audio_stream *stream)
 {
-    return pcm_config.period_size *
-           audio_stream_frame_size((struct audio_stream *)stream);
+    return cached_output_hardware_config.period_size * audio_stream_frame_size(stream);
 }
 
 static uint32_t out_get_channels(const struct audio_stream *stream)
 {
+    // Always Stero for now. We will do *some* conversions in this HAL.
+    // TODO(pmclean) When AudioPolicyManager & AudioFlinger supports arbitrary channels
+    // rewrite this to return the ACTUAL channel format
     return AUDIO_CHANNEL_OUT_STEREO;
 }
 
 static audio_format_t out_get_format(const struct audio_stream *stream)
 {
+    // Always return 16-bit PCM. We will do *some* conversions in this HAL.
+    // TODO(pmclean) When AudioPolicyManager & AudioFlinger supports arbitrary PCM formats
+    // rewrite this to return the ACTUAL data format
     return AUDIO_FORMAT_PCM_16_BIT;
 }
 
@@ -146,39 +356,122 @@
 
 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
 {
+    ALOGV("usb:audio_hw::out out_set_parameters() keys:%s", kvpairs);
+
     struct stream_out *out = (struct stream_out *)stream;
     struct audio_device *adev = out->dev;
     struct str_parms *parms;
     char value[32];
-    int ret;
+    int param_val;
     int routing = 0;
+    int ret_value = 0;
 
     parms = str_parms_create_str(kvpairs);
     pthread_mutex_lock(&adev->lock);
 
-    ret = str_parms_get_str(parms, "card", value, sizeof(value));
-    if (ret >= 0)
-        adev->card = atoi(value);
+    bool recache_device_params = false;
+    param_val = str_parms_get_str(parms, "card", value, sizeof(value));
+    if (param_val >= 0) {
+        adev->out_card = atoi(value);
+        recache_device_params = true;
+    }
 
-    ret = str_parms_get_str(parms, "device", value, sizeof(value));
-    if (ret >= 0)
-        adev->device = atoi(value);
+    param_val = str_parms_get_str(parms, "device", value, sizeof(value));
+    if (param_val >= 0) {
+        adev->out_device = atoi(value);
+        recache_device_params = true;
+    }
+
+    if (recache_device_params && adev->out_card >= 0 && adev->out_device >= 0) {
+        ret_value = read_alsa_device_config(adev->out_card, adev->out_device, PCM_OUT,
+                                            &(cached_output_hardware_config));
+        output_hardware_config_is_cached = (ret_value == 0);
+    }
 
     pthread_mutex_unlock(&adev->lock);
     str_parms_destroy(parms);
 
-    return 0;
+    return ret_value;
 }
 
-static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
-{
-    return strdup("");
+//TODO(pmclean) it seems like both out_get_parameters() and in_get_parameters()
+// could be written in terms of a get_device_parameters(io_type)
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys) {
+    struct stream_out *out = (struct stream_out *) stream;
+    struct audio_device *adev = out->dev;
+
+    unsigned min, max;
+
+    struct str_parms *query = str_parms_create_str(keys);
+    struct str_parms *result = str_parms_create();
+
+    int num_written = 0;
+    char buffer[256];
+    int buffer_size = sizeof(buffer) / sizeof(buffer[0]);
+    char* result_str = NULL;
+
+    struct pcm_params * alsa_hw_params = pcm_params_get(adev->out_card, adev->out_device, PCM_OUT);
+
+    // These keys are from hardware/libhardware/include/audio.h
+    // supported sample rates
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
+        // pcm_hw_params doesn't have a list of supported samples rates, just a min and a max, so
+        // if they are different, return a list containing those two values, otherwise just the one.
+        min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE);
+        max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE);
+        num_written = snprintf(buffer, buffer_size, "%d", min);
+        if (min != max) {
+            snprintf(buffer + num_written, buffer_size - num_written, "|%d",
+                     max);
+        }
+        str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
+                          buffer);
+    }  // AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES
+
+    // supported channel counts
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
+        // Similarly for output channels count
+        min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+        max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS);
+        num_written = snprintf(buffer, buffer_size, "%d", min);
+        if (min != max) {
+            snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+        }
+        str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, buffer);
+    }  // AUDIO_PARAMETER_STREAM_SUP_CHANNELS
+
+    // supported sample formats
+    if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+        // Similarly for output channels count
+        //TODO(pmclean): this is wrong.
+        min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+        max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+        num_written = snprintf(buffer, buffer_size, "%d", min);
+        if (min != max) {
+            snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+        }
+        str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS, buffer);
+    }  // AUDIO_PARAMETER_STREAM_SUP_FORMATS
+
+    result_str = str_parms_to_str(result);
+
+    // done with these...
+    str_parms_destroy(query);
+    str_parms_destroy(result);
+
+    return result_str;
 }
 
 static uint32_t out_get_latency(const struct audio_stream_out *stream)
 {
-    return (pcm_config.period_size * pcm_config.period_count * 1000) /
-            out_get_sample_rate(&stream->common);
+    struct stream_out *out = (struct stream_out *)stream;
+
+    //TODO(pmclean): Do we need a term here for the USB latency
+    // (as reported in the USB descriptors)?
+    uint32_t latency = (cached_output_hardware_config.period_size *
+        cached_output_hardware_config.period_count * 1000) / out_get_sample_rate(&stream->common);
+    return latency;
 }
 
 static int out_set_volume(struct audio_stream_out *stream, float left,
@@ -187,8 +480,41 @@
     return -ENOSYS;
 }
 
-static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
-                         size_t bytes)
+/* must be called with hw device and output stream mutexes locked */
+static int start_output_stream(struct stream_out *out)
+{
+    struct audio_device *adev = out->dev;
+    int return_val = 0;
+
+     ALOGV("usb:audio_hw::out start_output_stream(card:%d device:%d)",
+           adev->out_card, adev->out_device);
+
+    out->pcm = pcm_open(adev->out_card, adev->out_device, PCM_OUT, &cached_output_hardware_config);
+    if (out->pcm == NULL) {
+        return -ENOMEM;
+    }
+
+    if (out->pcm && !pcm_is_ready(out->pcm)) {
+        ALOGE("audio_hw audio_hw pcm_open() failed: %s", pcm_get_error(out->pcm));
+        pcm_close(out->pcm);
+        return -ENOMEM;
+    }
+
+    // Setup conversion buffer
+    size_t buffer_size = out_get_buffer_size(&(out->stream.common));
+
+    // computer maximum potential buffer size.
+    // * 2 for stereo -> quad conversion
+    // * 3/2 for 16bit -> 24 bit conversion
+    //TODO(pmclean) - remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
+    // (and do these conversions themselves)
+    out->conversion_buffer_size = (buffer_size * 3 * 2) / 2;
+    out->conversion_buffer = realloc(out->conversion_buffer, out->conversion_buffer_size);
+
+    return 0;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
 {
     int ret;
     struct stream_out *out = (struct stream_out *)stream;
@@ -203,7 +529,45 @@
         out->standby = false;
     }
 
-    pcm_write(out->pcm, (void *)buffer, bytes);
+    void * write_buff = buffer;
+    int num_write_buff_bytes = bytes;
+
+    /*
+     * Num Channels conversion
+     */
+    int num_device_channels = cached_output_hardware_config.channels;
+    int num_req_channels = 2; /* always, for now */
+    if (num_device_channels != num_req_channels && num_device_channels == 4) {
+        num_write_buff_bytes =
+                convert_2chan16_to_4chan16(write_buff, num_write_buff_bytes / 2,
+                                           out->conversion_buffer);
+        write_buff = out->conversion_buffer;
+    }
+
+    /*
+     *  16 vs 24-bit logic here
+     */
+    switch (cached_output_hardware_config.format) {
+    case PCM_FORMAT_S16_LE:
+        // the output format is the same as the input format, so just write it out
+        break;
+
+    case PCM_FORMAT_S24_3LE:
+        // 16-bit LE2 - 24-bit LE3
+        num_write_buff_bytes =
+                convert_16_to_24_3(write_buff, num_write_buff_bytes / 2, out->conversion_buffer);
+        write_buff = out->conversion_buffer;
+        break;
+
+    default:
+        // hmmmmm.....
+        ALOGV("usb:Unknown Format!!!");
+        break;
+    }
+
+    if (write_buff != NULL && num_write_buff_bytes != 0) {
+        pcm_write(out->pcm, write_buff, num_write_buff_bytes);
+    }
 
     pthread_mutex_unlock(&out->lock);
     pthread_mutex_unlock(&out->dev->lock);
@@ -250,14 +614,18 @@
                                    struct audio_config *config,
                                    struct audio_stream_out **stream_out)
 {
+    ALOGV("usb:audio_hw::out adev_open_output_stream() handle:0x%X, devices:0x%X, flags:0x%X",
+          handle, devices, flags);
+
     struct audio_device *adev = (struct audio_device *)dev;
+
     struct stream_out *out;
-    int ret;
 
     out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
     if (!out)
         return -ENOMEM;
 
+    // setup function pointers
     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;
@@ -278,30 +646,64 @@
 
     out->dev = adev;
 
-    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);
+    if (output_hardware_config_is_cached) {
+        config->sample_rate = cached_output_hardware_config.rate;
+
+        config->format = alsa_to_fw_format_id(cached_output_hardware_config.format);
+        if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+            // Always report PCM16 for now. AudioPolicyManagerBase/AudioFlinger dont' understand
+            // formats with more other format, so we won't get chosen (say with a 24bit DAC).
+            //TODO(pmclean) remove this when the above restriction is removed.
+            config->format = AUDIO_FORMAT_PCM_16_BIT;
+        }
+
+        config->channel_mask =
+                audio_channel_out_mask_from_count(cached_output_hardware_config.channels);
+        if (config->channel_mask != AUDIO_CHANNEL_OUT_STEREO) {
+            // Always report STEREO for now.  AudioPolicyManagerBase/AudioFlinger dont' understand
+            // formats with more channels, so we won't get chosen (say with a 4-channel DAC).
+            //TODO(pmclean) remove this when the above restriction is removed.
+            config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+        }
+    } else {
+        cached_output_hardware_config = default_alsa_out_config;
+
+        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);
+    }
+    ALOGV("usb:audio_hw  config->sample_rate:%d", config->sample_rate);
+    ALOGV("usb:audio_hw  config->format:0x%X", config->format);
+    ALOGV("usb:audio_hw  config->channel_mask:0x%X", config->channel_mask);
+
+    out->conversion_buffer = NULL;
+    out->conversion_buffer_size = 0;
 
     out->standby = true;
 
-    adev->card = -1;
-    adev->device = -1;
-
     *stream_out = &out->stream;
     return 0;
 
 err_open:
     free(out);
     *stream_out = NULL;
-    return ret;
+    return -ENOSYS;
 }
 
 static void adev_close_output_stream(struct audio_hw_device *dev,
                                      struct audio_stream_out *stream)
 {
+    ALOGV("usb:audio_hw::out adev_close_output_stream()");
     struct stream_out *out = (struct stream_out *)stream;
 
+    //TODO(pmclean) why are we doing this when stream get's freed at the end
+    // because it closes the pcm device
     out_standby(&stream->common);
+
+    free(out->conversion_buffer);
+    out->conversion_buffer = NULL;
+    out->conversion_buffer_size = 0;
+
     free(stream);
 }
 
@@ -352,13 +754,264 @@
     return 0;
 }
 
+/* Helper functions */
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    return in->alsa_pcm_config.rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return -ENOSYS;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    size_t buff_size =
+            in->alsa_pcm_config.period_size
+            * audio_stream_frame_size((struct audio_stream *)stream);
+    return buff_size;
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+    struct stream_in *in = (struct stream_in *)stream;
+    //TODO(pmclean) this should be done with a num_channels_to_alsa_channels()
+    return in->alsa_pcm_config.channels == 2
+            ? AUDIO_CHANNEL_IN_STEREO : AUDIO_CHANNEL_IN_MONO;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+    // just report 16-bit, pcm for now.
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+    ALOGV("-pcm-audio_hw::in in_standby() [Not Implemented]");
+    return 0;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+    return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+    ALOGV("Vaudio_hw::in in_set_parameters() keys:%s", kvpairs);
+
+    struct stream_in *in = (struct stream_in *)stream;
+    struct audio_device *adev = in->dev;
+    struct str_parms *parms;
+    char value[32];
+    int param_val;
+    int routing = 0;
+    int ret_value = 0;
+
+    parms = str_parms_create_str(kvpairs);
+    pthread_mutex_lock(&adev->lock);
+
+    // Card/Device
+    param_val = str_parms_get_str(parms, "card", value, sizeof(value));
+    if (param_val >= 0) {
+        adev->in_card = atoi(value);
+    }
+
+    param_val = str_parms_get_str(parms, "device", value, sizeof(value));
+    if (param_val >= 0) {
+        adev->in_device = atoi(value);
+    }
+
+    if (adev->in_card >= 0 && adev->in_device >= 0) {
+        ret_value = read_alsa_device_config(adev->in_card, adev->in_device, PCM_IN, &(in->alsa_pcm_config));
+    }
+
+    pthread_mutex_unlock(&adev->lock);
+    str_parms_destroy(parms);
+
+    return ret_value;
+}
+
+//TODO(pmclean) it seems like both out_get_parameters() and in_get_parameters()
+// could be written in terms of a get_device_parameters(io_type)
+
+static char * in_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+  ALOGV("usb:audio_hw::in in_get_parameters() keys:%s", keys);
+
+  struct stream_in *in = (struct stream_in *)stream;
+  struct audio_device *adev = in->dev;
+
+  struct pcm_params * alsa_hw_params = pcm_params_get(adev->in_card, adev->in_device, PCM_IN);
+  if (alsa_hw_params == NULL)
+      return strdup("");
+
+  struct str_parms *query = str_parms_create_str(keys);
+  struct str_parms *result = str_parms_create();
+
+  int num_written = 0;
+  char buffer[256];
+  int buffer_size = sizeof(buffer)/sizeof(buffer[0]);
+  char* result_str = NULL;
+
+  unsigned min, max;
+
+  // These keys are from hardware/libhardware/include/audio.h
+  // supported sample rates
+  if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
+    // pcm_hw_params doesn't have a list of supported samples rates, just a min and a max, so
+    // if they are different, return a list containing those two values, otherwise just the one.
+    min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_RATE);
+    max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_RATE);
+    num_written = snprintf(buffer, buffer_size, "%d", min);
+    if (min != max) {
+      snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+    }
+    str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SAMPLING_RATE, buffer);
+  } // AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES
+
+  // supported channel counts
+  if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
+    // Similarly for output channels count
+    min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_CHANNELS);
+    max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_CHANNELS);
+    num_written = snprintf(buffer, buffer_size, "%d", min);
+    if (min != max) {
+      snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+    }
+    str_parms_add_str(result, AUDIO_PARAMETER_STREAM_CHANNELS, buffer);
+  } // AUDIO_PARAMETER_STREAM_SUP_CHANNELS
+
+  // supported sample formats
+  if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+    //TODO(pmclean): this is wrong.
+    min = pcm_params_get_min(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+    max = pcm_params_get_max(alsa_hw_params, PCM_PARAM_SAMPLE_BITS);
+    num_written = snprintf(buffer, buffer_size, "%d", min);
+    if (min != max) {
+      snprintf(buffer + num_written, buffer_size - num_written, "|%d", max);
+    }
+    str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS, buffer);
+  } // AUDIO_PARAMETER_STREAM_SUP_FORMATS
+
+  result_str = str_parms_to_str(result);
+
+  // done with these...
+  str_parms_destroy(query);
+  str_parms_destroy(result);
+
+  return result_str;
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    return 0;
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain) {
+    return 0;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes) {
+    struct stream_in * in = (struct stream_in *)stream;
+
+    int err = pcm_read(in->pcm, buffer, bytes);
+
+    return err == 0 ? bytes : 0;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) {
+    return 0;
+}
+
 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_config *hal_config,
                                   struct audio_stream_in **stream_in)
 {
-    return -ENOSYS;
+    ALOGV("usb:audio_hw::in adev_open_input_stream() rate:%d, chanMask:0x%X, fmt:%d",
+          hal_config->sample_rate,
+          hal_config->channel_mask,
+          hal_config->format);
+
+    struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
+    if (in == NULL)
+        return -ENOMEM;
+
+    // setup function pointers
+    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;
+
+    struct audio_device *adev = (struct audio_device *)dev;
+    in->dev = adev;
+
+    in->standby = true;
+    in->requested_rate = hal_config->sample_rate;
+    in->alsa_pcm_config = default_alsa_in_config;
+
+    if (hal_config->sample_rate != 0)
+        in->alsa_pcm_config.rate = hal_config->sample_rate;
+
+    //TODO(pmclean) is this correct, or do we need to map from ALSA format?
+    // hal_config->format is an audio_format_t
+    // logical
+    // hal_config->format = default_alsa_in_config.format;
+    //TODO(pmclean) use audio_format_from_pcm_format() (in hardware/audio_alsaops.h)
+    switch (default_alsa_in_config.format) {
+    case PCM_FORMAT_S32_LE:
+        hal_config->format = AUDIO_FORMAT_PCM_32_BIT;
+        break;
+
+    case PCM_FORMAT_S8:
+        hal_config->format = AUDIO_FORMAT_PCM_8_BIT;
+        break;
+
+    case PCM_FORMAT_S24_LE:
+        hal_config->format = AUDIO_FORMAT_PCM_8_24_BIT;
+        break;
+
+    case PCM_FORMAT_S24_3LE:
+        hal_config->format = AUDIO_FORMAT_PCM_8_24_BIT;
+        break;
+
+    default:
+    case PCM_FORMAT_S16_LE:
+        hal_config->format = AUDIO_FORMAT_PCM_16_BIT;
+        break;
+    }
+
+    *stream_in = &in->stream;
+
+    return 0;
 }
 
 static void adev_close_input_stream(struct audio_hw_device *dev,
@@ -373,22 +1026,25 @@
 
 static int adev_close(hw_device_t *device)
 {
-    struct audio_device *adev = (struct audio_device *)device;
+    ALOGV("usb:audio_hw::adev_close()");
 
+    struct audio_device *adev = (struct audio_device *)device;
     free(device);
+
+    output_hardware_config_is_cached = false;
+
     return 0;
 }
 
 static int adev_open(const hw_module_t* module, const char* name,
                      hw_device_t** device)
 {
-    struct audio_device *adev;
-    int ret;
+    // ALOGV("usb:audio_hw::adev_open(%s)", name);
 
     if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
         return -EINVAL;
 
-    adev = calloc(1, sizeof(struct audio_device));
+    struct audio_device *adev = calloc(1, sizeof(struct audio_device));
     if (!adev)
         return -ENOMEM;