Squashed commit for Google

BZ: 214804

Based on save/r44c_fugu_before_reset_bz214804

Change-Id: I9f38c7048bea5f554512519121d91927110cb6bc
Reviewed-on: https://android.intel.com/224231
Reviewed-by: Imberton, Guilhem <guilhem.imberton@intel.com>
Tested-by: Imberton, Guilhem <guilhem.imberton@intel.com>

2 files changed, 1023 insertions, 0 deletions

diff --git a/hdmi/Android.mk b/hdmi/Android.mk
new file mode 100644
index 0000000..754b936
--- /dev/null
+++ b/hdmi/Android.mk
@@ -0,0 +1,43 @@
+# Copyright (C) 2012 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+#      http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+LOCAL_PATH := $(call my-dir)
+
+# Build only when BOARD_USES_ALSA_AUDIO is set
+ifeq ($(BOARD_USES_ALSA_AUDIO), true)
+ifeq ($(BOARD_USES_TINY_ALSA_AUDIO), true)
+
+include $(CLEAR_VARS)
+
+LOCAL_PRELINK_MODULE := false
+
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+LOCAL_CFLAGS := -D_POSIX_C_SOURCE=200809
+
+LOCAL_SRC_FILES := \
+    tinyaudio_hw.c
+
+LOCAL_C_INCLUDES += \
+    external/tinyalsa/include
+
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa
+
+LOCAL_MODULE := audio.hdmi.$(TARGET_BOARD_PLATFORM)
+LOCAL_MODULE_TAGS := optional
+LOCAL_CFLAGS += -Werror
+
+include $(BUILD_SHARED_LIBRARY)
+
+endif #BOARD_USES_TINY_ALSA_AUDIO
+endif #BOARD_USES_ALSA_AUDIO
diff --git a/hdmi/tinyaudio_hw.c b/hdmi/tinyaudio_hw.c
new file mode 100644
index 0000000..9c8fc1b
--- /dev/null
+++ b/hdmi/tinyaudio_hw.c
@@ -0,0 +1,980 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0/
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/*
+ * This code has modified by Intel Corporation
+ */
+
+/*
+ * Copyright (c) 2014, Intel Corporation
+ *
+ * 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
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#define LOG_TAG "tiny_hdmi_audio_hw"
+//#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/time.h>
+#include <stdlib.h>
+
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <hardware/audio.h>
+
+#include <sound/asound.h>
+#include <tinyalsa/asoundlib.h>
+
+#define UNUSED_PARAMETER(x)        (void)(x)
+
+#define DEFAULT_CARD               0
+#define DEFAULT_DEVICE             0
+
+/*this is used to avoid starvation*/
+#define LATENCY_TO_BUFFER_SIZE_RATIO 2
+
+/*Playback Channel Map*/
+#define CHANNEL_MAP_REQUEST      2
+
+/*global - keep track of the active device.
+This is needed since we are supporting more
+than one profile for HDMI. The Flinger
+assumes we can suport multiple streams
+at the same time. This makes sure only one stream
+is active at a time.*/
+struct pcm * activePcm = NULL;
+/*TODO - move active channel inside activepcm*/
+static unsigned int activeChannel;
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+#define STRING_TO_ENUM(string) { #string, string }
+
+struct channel_list {
+    const char *name;
+    uint32_t value;
+};
+
+const struct channel_list channel_list_table[] = {
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+    STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+struct pcm_config pcm_config_default = {
+    .channels = 2,
+    .rate = 44100,
+    .period_size = 1024,
+    .period_count = 4,
+    .format = PCM_FORMAT_S24_LE,
+};
+
+#define CHANNEL_MASK_MAX 3
+struct audio_device {
+    struct audio_hw_device hw_device;
+
+    pthread_mutex_t lock;
+    int card;
+    int device;
+    bool standby;
+    int sink_sup_channels;
+    audio_channel_mask_t sup_channel_masks[CHANNEL_MASK_MAX];
+};
+
+struct stream_out {
+    struct audio_stream_out stream;
+
+    pthread_mutex_t lock;
+    struct pcm *pcm;
+    bool standby;
+
+/* PCM Stream Configurations */
+    struct pcm_config pcm_config;
+    uint32_t   channel_mask;
+
+ /* ALSA PCM Configurations */
+    uint32_t   sample_rate;
+    uint32_t   buffer_size;
+    uint32_t   channels;
+    uint32_t   latency;
+
+    struct audio_device *dev;
+};
+
+/**
+ * NOTE: when multiple mutexes have to be acquired, always respect the
+ * following order: hw device > out stream
+ */
+
+/* Helper functions */
+
+// This function return the card number associated with the card ID (name)
+// passed as argument
+static int get_card_number_by_name(const char* name)
+{
+    char id_filepath[PATH_MAX] = {0};
+    char number_filepath[PATH_MAX] = {0};
+    ssize_t written;
+
+    snprintf(id_filepath, sizeof(id_filepath), "/proc/asound/%s", name);
+
+    written = readlink(id_filepath, number_filepath, sizeof(number_filepath));
+    if (written < 0) {
+        ALOGE("Sound card %s does not exist - setting default", name);
+        return DEFAULT_CARD;
+    } else if (written >= (ssize_t)sizeof(id_filepath)) {
+        ALOGE("Sound card %s name is too long - setting default", name);
+        return DEFAULT_CARD;
+    }
+
+    // We are assured, because of the check in the previous elseif, that this
+    // buffer is null-terminated.  So this call is safe.
+    // 4 == strlen("card")
+    return atoi(number_filepath + 4);
+}
+
+static enum pcm_format Get_SinkSupported_format()
+{
+   /*TODO : query sink supported formats*/
+   return PCM_FORMAT_S24_LE;
+}
+
+static int format_to_bits(enum pcm_format pcmformat)
+{
+  switch (pcmformat) {
+    case PCM_FORMAT_S32_LE:
+         return 32;
+    case PCM_FORMAT_S24_LE:
+         return 24;
+    default:
+    case PCM_FORMAT_S16_LE:
+         return 16;
+  };
+}
+
+static int make_sinkcompliant_buffers(void* input, void *output, int ipbytes)
+{
+  int i = 0,outbytes = 0;
+  enum pcm_format in_pcmformat;
+  enum pcm_format out_pcmformat;
+  int *src = (int*)input;
+  int *dst = (int*)output;
+
+  /*by default android currently support only
+    16 bit signed PCM*/
+  in_pcmformat = PCM_FORMAT_S16_LE;
+  out_pcmformat = Get_SinkSupported_format();
+
+  switch (out_pcmformat) {
+    default:
+    case PCM_FORMAT_S24_LE:
+    {
+       ALOGV("convert 16 to 24 bits for %d",ipbytes);
+       /*convert 16 bit input to 24 bit output
+         in a 32 bit sample*/
+       if(0 == ipbytes)
+          break;
+
+       for(i = 0; i < (ipbytes/4); i++){
+          int x = (int)((int*)src)[i];
+          dst[i*2] = ((int)( x & 0x0000FFFF)) << 8;
+          // trying to sign exdent
+          dst[i*2] = dst[i*2] << 8;
+          dst[i*2] = dst[i*2] >> 8;
+          //shift to middle
+          dst[i*2 + 1] = (int)(( x & 0xFFFF0000) >> 8);
+          dst[i*2 + 1] = dst[i*2 + 1] << 8;
+          dst[i*2 + 1] = dst[i*2 + 1] >> 8;
+        }
+        outbytes=ipbytes * 2;
+
+    }//case
+  };//switch
+
+  return outbytes;
+}
+
+/* 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 hdmicard = 0;
+
+    ALOGV("%s enter",__func__);
+
+    if ((adev->card < 0) || (adev->device < 0)){
+        /*this will be updated once the hot plug intent
+          sends these information.*/
+        adev->card = DEFAULT_CARD;
+        adev->device = DEFAULT_DEVICE;
+        ALOGV("%s : Setting default card/ device %d,%d",__func__,adev->card,adev->device);
+    }
+
+    ALOGV("%s enter %d,%d,%d,%d,%d",__func__,
+          out->pcm_config.channels,
+          out->pcm_config.rate,
+          out->pcm_config.period_size,
+          out->pcm_config.period_count,
+          out->pcm_config.format);
+
+    out->pcm_config.start_threshold = 0;
+    out->pcm_config.stop_threshold = 0;
+    out->pcm_config.silence_threshold = 0;
+
+    if(activePcm){
+      ALOGV("Closing already open tiny alsa stream running state %d",(int)(activePcm));
+      pcm_close(activePcm);
+      activePcm = NULL;
+    }
+
+    /*TODO - this needs to be updated once the device connect intent sends
+      card, device id*/
+    adev->card = get_card_number_by_name("IntelHDMI");
+    ALOGD("%s: HDMI card number = %d, device = %d",__func__,adev->card,adev->device);
+
+    out->pcm = pcm_open(adev->card, adev->device, PCM_OUT, &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);
+        activePcm = NULL;
+        return -ENOMEM;
+    }
+
+    activePcm = out->pcm;
+    activeChannel = out->pcm_config.channels;
+
+    ALOGV("Initialized PCM device for channels %d handle = %d",out->pcm_config.channels, (int)activePcm);
+    ALOGV("%s exit",__func__);
+    return 0;
+}
+
+/* API functions */
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    return out->pcm_config.rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(rate);
+
+    return 0;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    size_t buf_size;
+
+    if(out->channel_mask > 2){
+       buf_size = out->pcm_config.period_size *
+                  audio_stream_frame_size((struct audio_stream *)stream);
+    }
+    else{
+       buf_size = out->pcm_config.period_size *
+                  out->pcm_config.period_count *
+                  audio_stream_frame_size((struct audio_stream *)stream);
+
+       /*latency of audio flinger is based on this
+         buffer size. modifying the buffer size to avoid
+         starvation*/
+       buf_size/=LATENCY_TO_BUFFER_SIZE_RATIO;
+    }
+
+    ALOGV("%s : %d, period_size : %d, frame_size : %d",
+        __func__,
+        buf_size,
+        out->pcm_config.period_size,
+        audio_stream_frame_size((struct audio_stream *)stream));
+
+    return buf_size;
+
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    ALOGV("%s channel mask : %x",__func__,out->channel_mask);
+    return out->channel_mask;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+    UNUSED_PARAMETER(stream);
+
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(format);
+
+    return 0;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+
+    ALOGV("%s enter standby = %d",__func__,out->standby);
+
+    pthread_mutex_lock(&out->dev->lock);
+    pthread_mutex_lock(&out->lock);
+
+    if (!out->standby && activePcm) {
+        pcm_close(activePcm);
+        out->pcm = NULL;
+        out->standby = true;
+        activePcm = NULL;
+        ALOGV("%s PCM device closed",__func__);
+    }
+
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&out->dev->lock);
+
+    ALOGV("%s exit",__func__);
+    return 0;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(fd);
+    return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *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 routing = 0;
+    ALOGV("%s enter",__func__);
+
+    parms = str_parms_create_str(kvpairs);
+
+    pthread_mutex_lock(&adev->lock);
+
+    if (parms == NULL) {
+        ALOGE("couldn't extract string params from key value pairs");
+        pthread_mutex_unlock(&adev->lock);
+        return 0;
+    }
+
+    ret = str_parms_get_str(parms, "card", value, sizeof(value));
+    if (ret >= 0)
+        adev->card = atoi(value);
+
+    ret = str_parms_get_str(parms, "device", value, sizeof(value));
+    if (ret >= 0)
+        adev->device = atoi(value);
+
+    pthread_mutex_unlock(&adev->lock);
+    str_parms_destroy(parms);
+
+    ALOGV("%s exit",__func__);
+    return 0;
+}
+
+static int parse_channel_map()
+{
+    struct mixer *mixer;
+    int card = 0;
+    struct mixer_ctl *ctl;
+    enum mixer_ctl_type type;
+    unsigned int num_values;
+    unsigned int i,id;
+    int chcount=0, chmap=0;
+
+    card = get_card_number_by_name("IntelHDMI");
+    mixer = mixer_open(card);
+    if (!mixer) {
+        ALOGE("[EDID] Failed to open mixer\n");
+        goto chmap_error;
+    }
+
+    id = CHANNEL_MAP_REQUEST;
+    if (id >= mixer_get_num_ctls(mixer)) {
+        ALOGE("[EDID] Invalid request for channel map %d",id);
+        goto chmap_error;
+    }
+
+    ctl = mixer_get_ctl_by_name(mixer, "Playback Channel Map");
+
+    //ctl = mixer_get_ctl(mixer, id);
+
+    type = mixer_ctl_get_type(ctl);
+    num_values = mixer_ctl_get_num_values(ctl);
+
+    ALOGV("[EDID]id = %d",id);
+    ALOGV("[EDID]type = %d",type);
+    ALOGV("[EDID]count = %d",num_values);
+
+    for (i = 0; i < num_values; i++) {
+      switch (type)
+      {
+       case MIXER_CTL_TYPE_INT:
+            chmap = mixer_ctl_get_value(ctl, i);
+            ALOGD("[EDID]chmap = %d", chmap);
+            if(chmap > 0)  ++chcount;
+            break;
+       default:
+            printf(" unknown");
+            break;
+      };
+    }//for
+
+    ALOGD("[EDID]valid number of channels supported by sink = %d",chcount);
+
+    mixer_close(mixer);
+
+    return chcount;
+
+chmap_error:
+    mixer_close(mixer);
+    return 2;//stereo by default
+
+}
+
+static int out_read_edid(const struct stream_out *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    struct audio_device *adev = out->dev;
+
+    /**read the channel max param from the sink*/
+    adev->sink_sup_channels = parse_channel_map();
+
+    if(adev->sink_sup_channels == 8) {
+      adev->sup_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
+      adev->sup_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1;
+    }
+    else if((adev->sink_sup_channels == 6) || (adev->sink_sup_channels > 2)) {
+      adev->sup_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
+    }
+    else {
+      adev->sup_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
+    }
+
+    ALOGV("%s sink supports 0x%x max channels", __func__,adev->sink_sup_channels);
+    return 0;
+}
+
+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;
+    struct str_parms *params_in = str_parms_create_str(keys);
+    char *str = NULL;
+    char value[256] = {0};
+    int ret;
+    size_t i, j;
+    bool append = false;
+
+    struct str_parms *params_out = str_parms_create();
+
+    ALOGV("%s Entered %s", __func__,keys);
+
+    if (params_in) {
+        ret = str_parms_get_str(params_in, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
+        if (ret >= 0) {
+            /*read the channel support from sink*/
+            out_read_edid(out);
+
+            value[0] = '\0';
+            for (i = 0; i < CHANNEL_MASK_MAX; i++) {
+               for (j = 0; j < ARRAY_SIZE(channel_list_table); j++) {
+                   if (channel_list_table[j].value == adev->sup_channel_masks[i]) {
+                      if (append) {
+                          strcat(value, "|");
+                      }
+                      strcat(value, channel_list_table[j].name);
+                      append = true;
+                      break;
+                   }
+               }
+            }
+        }
+    }
+    if (params_out) {
+        str_parms_add_str(params_out, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
+        str = str_parms_to_str(params_out);
+    } else {
+        str = strdup(keys);
+    }
+
+    ALOGV("%s AUDIO_PARAMETER_STREAM_SUP_CHANNELS %s", __func__,str);
+    if (params_in) {
+        str_parms_destroy(params_in);
+    }
+    if (params_out) {
+        str_parms_destroy(params_out);
+    }
+
+    return str;
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    struct stream_out *out = (struct stream_out *)stream;
+    return (out->pcm_config.period_size * out->pcm_config.period_count * 1000) /
+            out_get_sample_rate(&stream->common);
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(left);
+    UNUSED_PARAMETER(right);
+
+    return -ENOSYS;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
+                         size_t bytes)
+{
+    int ret = 0;
+    struct stream_out *out = (struct stream_out *)stream;
+    int32_t* dstbuff = NULL;
+    int outbytes = 0;
+
+    ALOGV("%s enter for bytes = %d channels = %d",__func__,bytes, out->pcm_config.channels);
+
+    pthread_mutex_lock(&out->dev->lock);
+    pthread_mutex_lock(&out->lock);
+
+    if(activePcm == NULL) {
+       ALOGV("%s: previous stream closed- open again",__func__);
+       out->standby = true;
+    }
+
+    if (out->standby) {
+        ret = start_output_stream(out);
+        if (ret != 0) {
+            goto err;
+        }
+        out->standby = false;
+    }
+
+    if((!out->pcm) || (activeChannel != out->pcm_config.channels)){
+       ALOGD("%s: null handle to write - device already closed",__func__);
+       goto err;
+    }
+
+    if(Get_SinkSupported_format() == out->pcm_config.format){
+
+       /*16 bit data will be converted to 24 bit over 32 bit data type
+       hence the multiplier 2*/
+       dstbuff = (int32_t*)malloc(bytes* 2);
+       if (!dstbuff) {
+           pthread_mutex_unlock(&out->lock);
+           pthread_mutex_unlock(&out->dev->lock);
+           ALOGE("%s : memory allocation failed",__func__);
+           return -ENOMEM;
+       }
+
+       memset(dstbuff,0,bytes * 2);
+
+       outbytes = make_sinkcompliant_buffers((void*)buffer, (void*)dstbuff,bytes);
+     } //if()for conversion
+
+    if(dstbuff){
+      ret = pcm_write(out->pcm, (void *)dstbuff, outbytes);
+    }
+    else
+      ret = pcm_write(out->pcm, (void *)buffer, bytes);
+
+    ALOGV("pcm_write: %s done for %d input bytes, output bytes = %d ", pcm_get_error(out->pcm),bytes,outbytes);
+
+    free(dstbuff);
+
+err:
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&out->dev->lock);
+
+   if(ret !=0){
+    uint64_t duration_ms = ((bytes * 1000)/
+                            (audio_stream_frame_size(&stream->common)) /
+                            (out_get_sample_rate(&stream->common)));
+    ALOGV("%s : silence written", __func__);
+    usleep(duration_ms * 1000);
+   }
+
+    ALOGV("%s exit",__func__);
+    return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(dsp_frames);
+
+    return -EINVAL;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(effect);
+
+    return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(effect);
+
+    return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
+                                        int64_t *timestamp)
+{
+    UNUSED_PARAMETER(stream);
+    UNUSED_PARAMETER(timestamp);
+
+    return -EINVAL;
+}
+
+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)
+{
+    UNUSED_PARAMETER(devices);
+    UNUSED_PARAMETER(handle);
+
+    struct audio_device *adev = (struct audio_device *)dev;
+    struct stream_out *out;
+    int ret;
+    ALOGV("%s enter",__func__);
+
+    out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
+    if (!out)
+        return -ENOMEM;
+
+
+    out->dev = adev;
+    out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    adev->sup_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
+
+    if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        ALOGV("%s: HDMI Multichannel",__func__);
+        if (config->sample_rate == 0)
+            config->sample_rate = pcm_config_default.rate;
+        if (config->channel_mask == 0){
+            /*read the channel support from sink*/
+            out_read_edid(out);
+            if(config->channel_mask == 0)
+               config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+        }
+    } else {
+        ALOGV("%s: HDMI Stereo",__func__);
+        if (config->sample_rate == 0)
+            config->sample_rate = pcm_config_default.rate;
+        if (config->channel_mask == 0)
+            config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    }
+
+    out->channel_mask                      = config->channel_mask;
+
+    out->pcm_config.channels               = popcount(config->channel_mask);
+    out->pcm_config.rate                   = config->sample_rate;
+    out->pcm_config.period_size            = pcm_config_default.period_size;
+    out->pcm_config.period_count           = pcm_config_default.period_count;
+    out->pcm_config.format                 = pcm_config_default.format;
+
+    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;
+
+    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);
+
+    out->standby = true;
+
+    adev->card = -1;
+    adev->device = -1;
+
+    pthread_mutex_lock(&out->dev->lock);
+    pthread_mutex_lock(&out->lock);
+
+    if(activePcm){
+      ALOGV("Closing already open tiny alsa stream %d",(int)out->pcm);
+      pcm_close(activePcm);
+      activePcm = NULL;
+    }
+    ret = start_output_stream(out);
+    if (ret != 0) {
+        ALOGV("%s: stream start failed", __func__);
+        goto err_open;
+    }
+
+    out->standby = false;
+
+    *stream_out = &out->stream;
+
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&out->dev->lock);
+
+    ALOGV("%s exit",__func__);
+    return 0;
+
+err_open:
+    ALOGE("%s exit with error",__func__);
+    pthread_mutex_unlock(&out->lock);
+    pthread_mutex_unlock(&out->dev->lock);
+    free(out);
+    *stream_out = NULL;
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    UNUSED_PARAMETER(dev);
+
+    struct stream_out *out = (struct stream_out *)stream;
+
+    ALOGV("%s enter",__func__);
+    out->standby = false;
+    out_standby(&stream->common);
+    free(stream);
+    ALOGV("%s exit",__func__);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(kvpairs);
+
+    return 0;
+}
+
+static char * adev_get_parameters(const struct audio_hw_device *dev,
+                                  const char *keys)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(keys);
+
+    return strdup("");
+}
+
+static int adev_init_check(const struct audio_hw_device *dev)
+{
+    UNUSED_PARAMETER(dev);
+
+    return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(volume);
+
+    return -ENOSYS;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(volume);
+
+    return -ENOSYS;
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(mode);
+
+    return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(state);
+
+    return -ENOSYS;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(state);
+
+    return -ENOSYS;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         const struct audio_config *config)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(config);
+
+    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_stream_in **stream_in)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(handle);
+    UNUSED_PARAMETER(devices);
+    UNUSED_PARAMETER(config);
+    UNUSED_PARAMETER(stream_in);
+
+    return -ENOSYS;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                                   struct audio_stream_in *stream)
+{
+    UNUSED_PARAMETER(dev);
+    UNUSED_PARAMETER(stream);
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+    UNUSED_PARAMETER(device);
+    UNUSED_PARAMETER(fd);
+
+    return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+    struct audio_device *adev = (struct audio_device *)device;
+
+    free(device);
+    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("%s enter",__func__);
+
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    adev = calloc(1, sizeof(struct audio_device));
+    if (!adev)
+        return -ENOMEM;
+
+    adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
+    adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
+    adev->hw_device.common.module = (struct hw_module_t *) module;
+    adev->hw_device.common.close = adev_close;
+
+    adev->hw_device.init_check = adev_init_check;
+    adev->hw_device.set_voice_volume = adev_set_voice_volume;
+    adev->hw_device.set_master_volume = adev_set_master_volume;
+    adev->hw_device.set_mode = adev_set_mode;
+    adev->hw_device.set_mic_mute = adev_set_mic_mute;
+    adev->hw_device.get_mic_mute = adev_get_mic_mute;
+    adev->hw_device.set_parameters = adev_set_parameters;
+    adev->hw_device.get_parameters = adev_get_parameters;
+    adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
+    adev->hw_device.open_output_stream = adev_open_output_stream;
+    adev->hw_device.close_output_stream = adev_close_output_stream;
+    adev->hw_device.open_input_stream = adev_open_input_stream;
+    adev->hw_device.close_input_stream = adev_close_input_stream;
+    adev->hw_device.dump = adev_dump;
+
+    *device = &adev->hw_device.common;
+
+    ALOGV("%s exit",__func__);
+
+    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 = "tiny_hdmi audio HW HAL",
+        .author = "The Android Open Source Project",
+        .methods = &hal_module_methods,
+    },
+};