/******************************************************************************
 *
 *  Copyright 2018 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.
 *
 ******************************************************************************/

/* Implements hal for bluedroid ha audio device */

#define LOG_TAG "bt_hearing_aid_hw"

#include "audio_hearing_aid_hw/include/audio_hearing_aid_hw.h"

#include <fcntl.h>
#include <hardware/audio.h>
#include <hardware/hardware.h>
#include <inttypes.h>
#include <log/log.h>
#include <stdint.h>
#include <sys/socket.h>
#include <sys/stat.h>
#include <sys/time.h>
#include <sys/un.h>
#include <system/audio.h>
#include <unistd.h>

#include <cerrno>
#include <mutex>

#include "osi/include/hash_map_utils.h"
#include "osi/include/osi.h"
#include "osi/include/socket_utils/sockets.h"

/*****************************************************************************
 *  Constants & Macros
 *****************************************************************************/

#define CTRL_CHAN_RETRY_COUNT 3
#define USEC_PER_SEC 1000000L
#define SOCK_SEND_TIMEOUT_MS 2000 /* Timeout for sending */
#define SOCK_RECV_TIMEOUT_MS 5000 /* Timeout for receiving */

// set WRITE_POLL_MS to 0 for blocking sockets, nonzero for polled non-blocking
// sockets
#define WRITE_POLL_MS 20

#define FNLOG() ALOGV("%s:%d %s: ", __FILE__, __LINE__, __func__)
#define DEBUG(fmt, args...) ALOGD("%s:%d %s: " fmt, __FILE__, __LINE__, __func__, ##args)
#define INFO(fmt, args...) ALOGI("%s:%d %s: " fmt, __FILE__, __LINE__, __func__, ##args)
#define WARN(fmt, args...) ALOGW("%s:%d %s: " fmt, __FILE__, __LINE__, __func__, ##args)
#define ERROR(fmt, args...) ALOGE("%s:%d %s: " fmt, __FILE__, __LINE__, __func__, ##args)

#define ASSERTC(cond, msg, val)                                                 \
  if (!(cond)) {                                                                \
    ERROR("### ASSERT : %s line %d %s (%d) ###", __FILE__, __LINE__, msg, val); \
  }

#define CASE_RETURN_STR(const) \
  case const:                  \
    return #const;

static const char* audio_ha_hw_dump_ctrl_event(tHEARING_AID_CTRL_CMD event) {
  switch (event) {
    CASE_RETURN_STR(HEARING_AID_CTRL_CMD_NONE)
    CASE_RETURN_STR(HEARING_AID_CTRL_CMD_CHECK_READY)
    CASE_RETURN_STR(HEARING_AID_CTRL_CMD_START)
    CASE_RETURN_STR(HEARING_AID_CTRL_CMD_STOP)
    CASE_RETURN_STR(HEARING_AID_CTRL_CMD_SUSPEND)
    CASE_RETURN_STR(HEARING_AID_CTRL_GET_INPUT_AUDIO_CONFIG)
    CASE_RETURN_STR(HEARING_AID_CTRL_GET_OUTPUT_AUDIO_CONFIG)
    CASE_RETURN_STR(HEARING_AID_CTRL_SET_OUTPUT_AUDIO_CONFIG)
    CASE_RETURN_STR(HEARING_AID_CTRL_CMD_OFFLOAD_START)
    default:
      break;
  }

  return "UNKNOWN HEARING_AID_CTRL_CMD";
}

/*****************************************************************************
 *  Local type definitions
 *****************************************************************************/

typedef enum {
  AUDIO_HA_STATE_STARTING,
  AUDIO_HA_STATE_STARTED,
  AUDIO_HA_STATE_STOPPING,
  AUDIO_HA_STATE_STOPPED,
  /* need explicit set param call to resume (suspend=false) */
  AUDIO_HA_STATE_SUSPENDED,
  AUDIO_HA_STATE_STANDBY /* allows write to autoresume */
} ha_state_t;

struct ha_stream_in;
struct ha_stream_out;

struct ha_audio_device {
  // Important: device must be first as an audio_hw_device* may be cast to
  // ha_audio_device* when the type is implicitly known.
  struct audio_hw_device device;
  std::recursive_mutex* mutex;  // See note below on mutex acquisition order.
  struct ha_stream_in* input;
  struct ha_stream_out* output;
};

struct ha_config {
  uint32_t rate;
  uint32_t channel_mask;
  bool is_stereo_to_mono;  // True if fetching Stereo and mixing into Mono
  int format;
};

/* move ctrl_fd outside output stream and keep open until HAL unloaded ? */

struct ha_stream_common {
  std::recursive_mutex* mutex;  // See note below on mutex acquisition order.
  int ctrl_fd;
  int audio_fd;
  size_t buffer_sz;
  struct ha_config cfg;
  ha_state_t state;
};

struct ha_stream_out {
  struct audio_stream_out stream;
  struct ha_stream_common common;
  uint64_t frames_presented;  // frames written, never reset
  uint64_t frames_rendered;   // frames written, reset on standby
};

struct ha_stream_in {
  struct audio_stream_in stream;
  struct ha_stream_common common;
};

/*
 * Mutex acquisition order:
 *
 * The ha_audio_device (adev) mutex must be acquired before
 * the ha_stream_common (out or in) mutex.
 *
 * This may differ from other audio HALs.
 */

/*****************************************************************************
 *  Static variables
 *****************************************************************************/

/*****************************************************************************
 *  Static functions
 *****************************************************************************/
static void hash_map_utils_dump_string_keys_string_values(
        std::unordered_map<std::string, std::string>& map) {
  for (const auto& ptr : map) {
    INFO("key: '%s' value: '%s'\n", ptr.first.c_str(), ptr.second.c_str());
  }
}

static size_t out_get_buffer_size(const struct audio_stream* stream);

/*****************************************************************************
 *  Externs
 *****************************************************************************/

/*****************************************************************************
 *  Functions
 *****************************************************************************/
static void ha_open_ctrl_path(struct ha_stream_common* common);

/*****************************************************************************
 *   Miscellaneous helper functions
 *****************************************************************************/

/* logs timestamp with microsec precision
   pprev is optional in case a dedicated diff is required */
static void ts_log(UNUSED_ATTR const char* tag, UNUSED_ATTR int val, struct timespec* pprev_opt) {
  struct timespec now;
  static struct timespec prev = {0, 0};
  unsigned long long now_us;
  unsigned long long diff_us;

  clock_gettime(CLOCK_MONOTONIC, &now);

  now_us = now.tv_sec * USEC_PER_SEC + now.tv_nsec / 1000;

  if (pprev_opt) {
    diff_us = (now.tv_sec - prev.tv_sec) * USEC_PER_SEC + (now.tv_nsec - prev.tv_nsec) / 1000;
    *pprev_opt = now;
    DEBUG("[%s] ts %08lld, *diff %08lld, val %d", tag, now_us, diff_us, val);
  } else {
    diff_us = (now.tv_sec - prev.tv_sec) * USEC_PER_SEC + (now.tv_nsec - prev.tv_nsec) / 1000;
    prev = now;
    DEBUG("[%s] ts %08lld, diff %08lld, val %d", tag, now_us, diff_us, val);
  }
}

static int calc_audiotime_usec(struct ha_config cfg, int bytes) {
  int chan_count = audio_channel_count_from_out_mask(cfg.channel_mask);
  int bytes_per_sample;

  switch (cfg.format) {
    case AUDIO_FORMAT_PCM_8_BIT:
      bytes_per_sample = 1;
      break;
    case AUDIO_FORMAT_PCM_16_BIT:
      bytes_per_sample = 2;
      break;
    case AUDIO_FORMAT_PCM_24_BIT_PACKED:
      bytes_per_sample = 3;
      break;
    case AUDIO_FORMAT_PCM_8_24_BIT:
      bytes_per_sample = 4;
      break;
    case AUDIO_FORMAT_PCM_32_BIT:
      bytes_per_sample = 4;
      break;
    default:
      ASSERTC(false, "unsupported sample format", cfg.format);
      bytes_per_sample = 2;
      break;
  }

  return (int)(((int64_t)bytes * (USEC_PER_SEC / (chan_count * bytes_per_sample))) / cfg.rate);
}

/*****************************************************************************
 *
 *   bluedroid stack adaptation
 *
 ****************************************************************************/

static int skt_connect(const char* path, size_t buffer_sz) {
  int ret;
  int skt_fd;
  int len;

  INFO("connect to %s (sz %zu)", path, buffer_sz);

  skt_fd = socket(AF_LOCAL, SOCK_STREAM, 0);

  if (osi_socket_local_client_connect(skt_fd, path, ANDROID_SOCKET_NAMESPACE_ABSTRACT,
                                      SOCK_STREAM) < 0) {
    ERROR("failed to connect (%s)", strerror(errno));
    close(skt_fd);
    return -1;
  }

  len = buffer_sz;
  ret = setsockopt(skt_fd, SOL_SOCKET, SO_SNDBUF, (char*)&len, (int)sizeof(len));
  if (ret < 0) {
    ERROR("setsockopt failed (%s)", strerror(errno));
  }

  ret = setsockopt(skt_fd, SOL_SOCKET, SO_RCVBUF, (char*)&len, (int)sizeof(len));
  if (ret < 0) {
    ERROR("setsockopt failed (%s)", strerror(errno));
  }

  /* Socket send/receive timeout value */
  struct timeval tv;
  tv.tv_sec = SOCK_SEND_TIMEOUT_MS / 1000;
  tv.tv_usec = (SOCK_SEND_TIMEOUT_MS % 1000) * 1000;

  ret = setsockopt(skt_fd, SOL_SOCKET, SO_SNDTIMEO, &tv, sizeof(tv));
  if (ret < 0) {
    ERROR("setsockopt failed (%s)", strerror(errno));
  }

  tv.tv_sec = SOCK_RECV_TIMEOUT_MS / 1000;
  tv.tv_usec = (SOCK_RECV_TIMEOUT_MS % 1000) * 1000;

  ret = setsockopt(skt_fd, SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv));
  if (ret < 0) {
    ERROR("setsockopt failed (%s)", strerror(errno));
  }

  INFO("connected to stack fd = %d", skt_fd);

  return skt_fd;
}

static int skt_read(int fd, void* p, size_t len) {
  ssize_t read;

  FNLOG();

  ts_log("skt_read recv", len, NULL);

  OSI_NO_INTR(read = recv(fd, p, len, MSG_NOSIGNAL));
  if (read == -1) {
    ERROR("read failed with errno=%d\n", errno);
  }

  return (int)read;
}

static int skt_write(int fd, const void* p, size_t len) {
  ssize_t sent;
  FNLOG();

  ts_log("skt_write", len, NULL);

  if (WRITE_POLL_MS == 0) {
    // do not poll, use blocking send
    OSI_NO_INTR(sent = send(fd, p, len, MSG_NOSIGNAL));
    if (sent == -1) {
      ERROR("write failed with error(%s)", strerror(errno));
    }

    return (int)sent;
  }

  // use non-blocking send, poll
  int ms_timeout = SOCK_SEND_TIMEOUT_MS;
  size_t count = 0;
  while (count < len) {
    OSI_NO_INTR(sent = send(fd, p, len - count, MSG_NOSIGNAL | MSG_DONTWAIT));
    if (sent == -1) {
      if (errno != EAGAIN && errno != EWOULDBLOCK) {
        ERROR("write failed with error(%s)", strerror(errno));
        return -1;
      }
      if (ms_timeout >= WRITE_POLL_MS) {
        usleep(WRITE_POLL_MS * 1000);
        ms_timeout -= WRITE_POLL_MS;
        continue;
      }
      WARN("write timeout exceeded, sent %zu bytes", count);
      return -1;
    }
    count += sent;
    p = (const uint8_t*)p + sent;
  }
  return (int)count;
}

static int skt_disconnect(int fd) {
  INFO("fd %d", fd);

  if (fd != AUDIO_SKT_DISCONNECTED) {
    shutdown(fd, SHUT_RDWR);
    close(fd);
  }
  return 0;
}

/*****************************************************************************
 *
 *  AUDIO CONTROL PATH
 *
 ****************************************************************************/

static int ha_ctrl_receive(struct ha_stream_common* common, void* buffer, size_t length) {
  ssize_t ret;
  int i;

  for (i = 0;; i++) {
    OSI_NO_INTR(ret = recv(common->ctrl_fd, buffer, length, MSG_NOSIGNAL));
    if (ret > 0) {
      break;
    }
    if (ret == 0) {
      ERROR("receive control data failed: peer closed");
      break;
    }
    if (errno != EWOULDBLOCK && errno != EAGAIN) {
      ERROR("receive control data failed: error(%s)", strerror(errno));
      break;
    }
    if (i == (CTRL_CHAN_RETRY_COUNT - 1)) {
      ERROR("receive control data failed: max retry count");
      break;
    }
    INFO("receive control data failed (%s), retrying", strerror(errno));
  }
  if (ret <= 0) {
    skt_disconnect(common->ctrl_fd);
    common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
  }
  return ret;
}

// Sends control info for stream |common|. The data to send is stored in
// |buffer| and has size |length|.
// On success, returns the number of octets sent, otherwise -1.
static int ha_ctrl_send(struct ha_stream_common* common, const void* buffer, size_t length) {
  ssize_t sent;
  size_t remaining = length;
  int i;

  if (length == 0) {
    return 0;  // Nothing to do
  }

  for (i = 0;; i++) {
    OSI_NO_INTR(sent = send(common->ctrl_fd, buffer, remaining, MSG_NOSIGNAL));
    if (sent == static_cast<ssize_t>(remaining)) {
      remaining = 0;
      break;
    }
    if (sent > 0) {
      buffer = (static_cast<const char*>(buffer) + sent);
      remaining -= sent;
      continue;
    }
    if (sent < 0) {
      if (errno != EWOULDBLOCK && errno != EAGAIN) {
        ERROR("send control data failed: error(%s)", strerror(errno));
        break;
      }
      INFO("send control data failed (%s), retrying", strerror(errno));
    }
    if (i >= (CTRL_CHAN_RETRY_COUNT - 1)) {
      ERROR("send control data failed: max retry count");
      break;
    }
  }
  if (remaining > 0) {
    skt_disconnect(common->ctrl_fd);
    common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
    return -1;
  }
  return length;
}

static int ha_command(struct ha_stream_common* common, tHEARING_AID_CTRL_CMD cmd) {
  char ack;

  DEBUG("HEARING_AID COMMAND %s", audio_ha_hw_dump_ctrl_event(cmd));

  if (common->ctrl_fd == AUDIO_SKT_DISCONNECTED) {
    INFO("starting up or recovering from previous error");
    ha_open_ctrl_path(common);
    if (common->ctrl_fd == AUDIO_SKT_DISCONNECTED) {
      ERROR("failure to open ctrl path");
      return -1;
    }
  }

  /* send command */
  ssize_t sent;
  OSI_NO_INTR(sent = send(common->ctrl_fd, &cmd, 1, MSG_NOSIGNAL));
  if (sent == -1) {
    ERROR("cmd failed (%s)", strerror(errno));
    skt_disconnect(common->ctrl_fd);
    common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
    return -1;
  }

  /* wait for ack byte */
  if (ha_ctrl_receive(common, &ack, 1) < 0) {
    ERROR("HEARING_AID COMMAND %s: no ACK", audio_ha_hw_dump_ctrl_event(cmd));
    return -1;
  }

  DEBUG("HEARING_AID COMMAND %s DONE STATUS %d", audio_ha_hw_dump_ctrl_event(cmd), ack);

  if (ack == HEARING_AID_CTRL_ACK_INCALL_FAILURE) {
    return ack;
  }
  if (ack != HEARING_AID_CTRL_ACK_SUCCESS) {
    ERROR("HEARING_AID COMMAND %s error %d", audio_ha_hw_dump_ctrl_event(cmd), ack);
    return -1;
  }

  return 0;
}

static int check_ha_ready(struct ha_stream_common* common) {
  if (ha_command(common, HEARING_AID_CTRL_CMD_CHECK_READY) < 0) {
    ERROR("check ha ready failed");
    return -1;
  }
  return 0;
}

static int ha_read_input_audio_config(struct ha_stream_common* common) {
  tHA_SAMPLE_RATE sample_rate;
  tHA_CHANNEL_COUNT channel_count;

  if (ha_command(common, HEARING_AID_CTRL_GET_INPUT_AUDIO_CONFIG) < 0) {
    ERROR("get ha input audio config failed");
    return -1;
  }

  if (ha_ctrl_receive(common, &sample_rate, sizeof(tHA_SAMPLE_RATE)) < 0) {
    return -1;
  }
  if (ha_ctrl_receive(common, &channel_count, sizeof(tHA_CHANNEL_COUNT)) < 0) {
    return -1;
  }

  switch (sample_rate) {
    case 16000:
    case 24000:
    case 44100:
    case 48000:
      common->cfg.rate = sample_rate;
      break;
    default:
      ERROR("Invalid sample rate: %" PRIu32, sample_rate);
      return -1;
  }

  switch (channel_count) {
    case 1:
      common->cfg.channel_mask = AUDIO_CHANNEL_IN_MONO;
      break;
    case 2:
      common->cfg.channel_mask = AUDIO_CHANNEL_IN_STEREO;
      break;
    default:
      ERROR("Invalid channel count: %" PRIu32, channel_count);
      return -1;
  }

  // TODO: For now input audio format is always hard-coded as PCM 16-bit
  common->cfg.format = AUDIO_FORMAT_PCM_16_BIT;

  INFO("got input audio config %d %d", common->cfg.format, common->cfg.rate);

  return 0;
}

static int ha_read_output_audio_config(struct ha_stream_common* common,
                                       btav_a2dp_codec_config_t* codec_config,
                                       btav_a2dp_codec_config_t* codec_capability,
                                       bool update_stream_config) {
  struct ha_config stream_config;

  if (ha_command(common, HEARING_AID_CTRL_GET_OUTPUT_AUDIO_CONFIG) < 0) {
    ERROR("get ha output audio config failed");
    return -1;
  }

  // Receive the current codec config
  if (ha_ctrl_receive(common, &codec_config->sample_rate, sizeof(btav_a2dp_codec_sample_rate_t)) <
      0) {
    return -1;
  }
  if (ha_ctrl_receive(common, &codec_config->bits_per_sample,
                      sizeof(btav_a2dp_codec_bits_per_sample_t)) < 0) {
    return -1;
  }
  if (ha_ctrl_receive(common, &codec_config->channel_mode, sizeof(btav_a2dp_codec_channel_mode_t)) <
      0) {
    return -1;
  }

  // Receive the current codec capability
  if (ha_ctrl_receive(common, &codec_capability->sample_rate,
                      sizeof(btav_a2dp_codec_sample_rate_t)) < 0) {
    return -1;
  }
  if (ha_ctrl_receive(common, &codec_capability->bits_per_sample,
                      sizeof(btav_a2dp_codec_bits_per_sample_t)) < 0) {
    return -1;
  }
  if (ha_ctrl_receive(common, &codec_capability->channel_mode,
                      sizeof(btav_a2dp_codec_channel_mode_t)) < 0) {
    return -1;
  }

  // Check the codec config sample rate
  switch (codec_config->sample_rate) {
    case BTAV_A2DP_CODEC_SAMPLE_RATE_44100:
      stream_config.rate = 44100;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_48000:
      stream_config.rate = 48000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_88200:
      stream_config.rate = 88200;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_96000:
      stream_config.rate = 96000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_176400:
      stream_config.rate = 176400;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_192000:
      stream_config.rate = 192000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_16000:
      stream_config.rate = 16000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_24000:
      stream_config.rate = 24000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_NONE:
    default:
      ERROR("Invalid sample rate: 0x%x", codec_config->sample_rate);
      return -1;
  }

  // Check the codec config bits per sample
  switch (codec_config->bits_per_sample) {
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16:
      stream_config.format = AUDIO_FORMAT_PCM_16_BIT;
      break;
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24:
      stream_config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
      break;
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32:
      stream_config.format = AUDIO_FORMAT_PCM_32_BIT;
      break;
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE:
    default:
      ERROR("Invalid bits per sample: 0x%x", codec_config->bits_per_sample);
      return -1;
  }

  // Check the codec config channel mode
  switch (codec_config->channel_mode) {
    case BTAV_A2DP_CODEC_CHANNEL_MODE_MONO:
      stream_config.channel_mask = AUDIO_CHANNEL_OUT_MONO;
      stream_config.is_stereo_to_mono = true;
      break;
    case BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO:
      stream_config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
      stream_config.is_stereo_to_mono = false;
      break;
    case BTAV_A2DP_CODEC_CHANNEL_MODE_NONE:
    default:
      ERROR("Invalid channel mode: 0x%x", codec_config->channel_mode);
      return -1;
  }
  if (stream_config.is_stereo_to_mono) {
    stream_config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
  }

  // Update the output stream configuration
  if (update_stream_config) {
    common->cfg.rate = stream_config.rate;
    common->cfg.channel_mask = stream_config.channel_mask;
    common->cfg.is_stereo_to_mono = stream_config.is_stereo_to_mono;
    common->cfg.format = stream_config.format;
    common->buffer_sz = audio_ha_hw_stream_compute_buffer_size(
            codec_config->sample_rate, codec_config->bits_per_sample, codec_config->channel_mode);
    if (common->cfg.is_stereo_to_mono) {
      // We need to fetch twice as much data from the Audio framework
      common->buffer_sz *= 2;
    }
  }

  INFO("got output codec config (update_stream_config=%s): "
       "sample_rate=0x%x bits_per_sample=0x%x channel_mode=0x%x",
       update_stream_config ? "true" : "false", codec_config->sample_rate,
       codec_config->bits_per_sample, codec_config->channel_mode);

  INFO("got output codec capability: sample_rate=0x%x bits_per_sample=0x%x channel_mode=0x%x",
       codec_capability->sample_rate, codec_capability->bits_per_sample,
       codec_capability->channel_mode);

  return 0;
}

static int ha_write_output_audio_config(struct ha_stream_common* common) {
  btav_a2dp_codec_config_t codec_config;

  if (ha_command(common, HEARING_AID_CTRL_SET_OUTPUT_AUDIO_CONFIG) < 0) {
    ERROR("set ha output audio config failed");
    return -1;
  }

  codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_NONE;
  codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE;
  codec_config.channel_mode = BTAV_A2DP_CODEC_CHANNEL_MODE_NONE;

  switch (common->cfg.rate) {
    case 44100:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_44100;
      break;
    case 48000:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_48000;
      break;
    case 88200:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_88200;
      break;
    case 96000:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_96000;
      break;
    case 176400:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_176400;
      break;
    case 192000:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_192000;
      break;
    case 16000:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_16000;
      break;
    case 24000:
      codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_24000;
      break;
    default:
      ERROR("Invalid sample rate: %" PRIu32, common->cfg.rate);
      return -1;
  }

  switch (common->cfg.format) {
    case AUDIO_FORMAT_PCM_16_BIT:
      codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16;
      break;
    case AUDIO_FORMAT_PCM_24_BIT_PACKED:
      codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24;
      break;
    case AUDIO_FORMAT_PCM_32_BIT:
      codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32;
      break;
    case AUDIO_FORMAT_PCM_8_24_BIT:
      // All 24-bit audio is expected in AUDIO_FORMAT_PCM_24_BIT_PACKED format
      FALLTHROUGH_INTENDED; /* FALLTHROUGH */
    default:
      ERROR("Invalid audio format: 0x%x", common->cfg.format);
      return -1;
  }

  switch (common->cfg.channel_mask) {
    case AUDIO_CHANNEL_OUT_MONO:
      codec_config.channel_mode = BTAV_A2DP_CODEC_CHANNEL_MODE_MONO;
      break;
    case AUDIO_CHANNEL_OUT_STEREO:
      if (common->cfg.is_stereo_to_mono) {
        codec_config.channel_mode = BTAV_A2DP_CODEC_CHANNEL_MODE_MONO;
      } else {
        codec_config.channel_mode = BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO;
      }
      break;
    default:
      ERROR("Invalid channel mask: 0x%x", common->cfg.channel_mask);
      return -1;
  }

  // Send the current codec config that has been selected by us
  if (ha_ctrl_send(common, &codec_config.sample_rate, sizeof(btav_a2dp_codec_sample_rate_t)) < 0) {
    return -1;
  }
  if (ha_ctrl_send(common, &codec_config.bits_per_sample,
                   sizeof(btav_a2dp_codec_bits_per_sample_t)) < 0) {
    return -1;
  }
  if (ha_ctrl_send(common, &codec_config.channel_mode, sizeof(btav_a2dp_codec_channel_mode_t)) <
      0) {
    return -1;
  }

  INFO("sent output codec config: sample_rate=0x%x bits_per_sample=0x%x channel_mode=0x%x",
       codec_config.sample_rate, codec_config.bits_per_sample, codec_config.channel_mode);

  return 0;
}

static void ha_open_ctrl_path(struct ha_stream_common* common) {
  int i;

  if (common->ctrl_fd != AUDIO_SKT_DISCONNECTED) {
    return;  // already connected
  }

  /* retry logic to catch any timing variations on control channel */
  for (i = 0; i < CTRL_CHAN_RETRY_COUNT; i++) {
    /* connect control channel if not already connected */
    if ((common->ctrl_fd =
                 skt_connect(HEARING_AID_CTRL_PATH, AUDIO_STREAM_CONTROL_OUTPUT_BUFFER_SZ)) >= 0) {
      /* success, now check if stack is ready */
      if (check_ha_ready(common) == 0) {
        break;
      }

      ERROR("error : ha not ready, wait 250 ms and retry");
      usleep(250000);
      skt_disconnect(common->ctrl_fd);
      common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
    }

    /* ctrl channel not ready, wait a bit */
    usleep(250000);
  }
}

/*****************************************************************************
 *
 * AUDIO DATA PATH
 *
 ****************************************************************************/

static void ha_stream_common_init(struct ha_stream_common* common) {
  FNLOG();

  common->mutex = new std::recursive_mutex;

  common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
  common->audio_fd = AUDIO_SKT_DISCONNECTED;
  common->state = AUDIO_HA_STATE_STOPPED;

  /* manages max capacity of socket pipe */
  common->buffer_sz = AUDIO_STREAM_OUTPUT_BUFFER_SZ;
}

static void ha_stream_common_destroy(struct ha_stream_common* common) {
  FNLOG();

  delete common->mutex;
  common->mutex = NULL;
}

static int start_audio_datapath(struct ha_stream_common* common) {
  INFO("state %d", common->state);

  int oldstate = common->state;
  common->state = AUDIO_HA_STATE_STARTING;

  int ha_status = ha_command(common, HEARING_AID_CTRL_CMD_START);
  if (ha_status < 0) {
    ERROR("Audiopath start failed (status %d)", ha_status);
    goto error;
  } else if (ha_status == HEARING_AID_CTRL_ACK_INCALL_FAILURE) {
    ERROR("Audiopath start failed - in call, move to suspended");
    goto error;
  }

  /* connect socket if not yet connected */
  if (common->audio_fd == AUDIO_SKT_DISCONNECTED) {
    common->audio_fd = skt_connect(HEARING_AID_DATA_PATH, common->buffer_sz);
    if (common->audio_fd < 0) {
      ERROR("Audiopath start failed - error opening data socket");
      goto error;
    }
  }
  common->state = (ha_state_t)AUDIO_HA_STATE_STARTED;
  return 0;

error:
  common->state = (ha_state_t)oldstate;
  return -1;
}

static int stop_audio_datapath(struct ha_stream_common* common) {
  int oldstate = common->state;

  INFO("state %d", common->state);

  /* prevent any stray output writes from autostarting the stream
     while stopping audiopath */
  common->state = AUDIO_HA_STATE_STOPPING;

  if (ha_command(common, HEARING_AID_CTRL_CMD_STOP) < 0) {
    ERROR("audiopath stop failed");
    common->state = (ha_state_t)oldstate;
    return -1;
  }

  common->state = (ha_state_t)AUDIO_HA_STATE_STOPPED;

  /* disconnect audio path */
  skt_disconnect(common->audio_fd);
  common->audio_fd = AUDIO_SKT_DISCONNECTED;

  return 0;
}

static int suspend_audio_datapath(struct ha_stream_common* common, bool standby) {
  INFO("state %d", common->state);

  if (common->state == AUDIO_HA_STATE_STOPPING) {
    return -1;
  }

  if (ha_command(common, HEARING_AID_CTRL_CMD_SUSPEND) < 0) {
    return -1;
  }

  if (standby) {
    common->state = AUDIO_HA_STATE_STANDBY;
  } else {
    common->state = AUDIO_HA_STATE_SUSPENDED;
  }

  /* disconnect audio path */
  skt_disconnect(common->audio_fd);

  common->audio_fd = AUDIO_SKT_DISCONNECTED;

  return 0;
}

/*****************************************************************************
 *
 *  audio output callbacks
 *
 ****************************************************************************/

static ssize_t out_write(struct audio_stream_out* stream, const void* buffer, size_t bytes) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;
  int sent = -1;
  size_t write_bytes = bytes;

  DEBUG("write %zu bytes (fd %d)", bytes, out->common.audio_fd);

  std::unique_lock<std::recursive_mutex> lock(*out->common.mutex);
  if (out->common.state == AUDIO_HA_STATE_SUSPENDED ||
      out->common.state == AUDIO_HA_STATE_STOPPING) {
    DEBUG("stream suspended or closing");
    goto finish;
  }

  /* only allow autostarting if we are in stopped or standby */
  if ((out->common.state == AUDIO_HA_STATE_STOPPED) ||
      (out->common.state == AUDIO_HA_STATE_STANDBY)) {
    if (start_audio_datapath(&out->common) < 0) {
      goto finish;
    }
  } else if (out->common.state != AUDIO_HA_STATE_STARTED) {
    ERROR("stream not in stopped or standby");
    goto finish;
  }

  // Mix the stereo into mono if necessary
  if (out->common.cfg.is_stereo_to_mono) {
    const size_t frames = bytes / audio_stream_out_frame_size(stream);
    int16_t* src = (int16_t*)buffer;
    int16_t* dst = (int16_t*)buffer;
    for (size_t i = 0; i < frames; i++, dst++, src += 2) {
      *dst = (int16_t)(((int32_t)src[0] + (int32_t)src[1]) >> 1);
    }
    write_bytes /= 2;
    DEBUG("stereo-to-mono mixing: write %zu bytes (fd %d)", write_bytes, out->common.audio_fd);
  }

  lock.unlock();
  sent = skt_write(out->common.audio_fd, buffer, write_bytes);
  lock.lock();

  if (sent == -1) {
    skt_disconnect(out->common.audio_fd);
    out->common.audio_fd = AUDIO_SKT_DISCONNECTED;
    if ((out->common.state != AUDIO_HA_STATE_SUSPENDED) &&
        (out->common.state != AUDIO_HA_STATE_STOPPING)) {
      out->common.state = AUDIO_HA_STATE_STOPPED;
    } else {
      ERROR("write failed : stream suspended, avoid resetting state");
    }
    goto finish;
  }

finish:;
  const size_t frames = bytes / audio_stream_out_frame_size(stream);
  out->frames_rendered += frames;
  out->frames_presented += frames;
  lock.unlock();

  // If send didn't work out, sleep to emulate write delay.
  if (sent == -1) {
    const int us_delay = calc_audiotime_usec(out->common.cfg, bytes);
    DEBUG("emulate ha write delay (%d us)", us_delay);
    usleep(us_delay);
  }
  return bytes;
}

static uint32_t out_get_sample_rate(const struct audio_stream* stream) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  DEBUG("rate %" PRIu32, out->common.cfg.rate);

  return out->common.cfg.rate;
}

static int out_set_sample_rate(struct audio_stream* stream, uint32_t rate) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  DEBUG("out_set_sample_rate : %" PRIu32, rate);

  out->common.cfg.rate = rate;

  return 0;
}

static size_t out_get_buffer_size(const struct audio_stream* stream) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;
  // period_size is the AudioFlinger mixer buffer size.
  const size_t period_size = out->common.buffer_sz / AUDIO_STREAM_OUTPUT_BUFFER_PERIODS;

  DEBUG("socket buffer size: %zu  period size: %zu", out->common.buffer_sz, period_size);

  return period_size;
}

size_t audio_ha_hw_stream_compute_buffer_size(
        btav_a2dp_codec_sample_rate_t codec_sample_rate,
        btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample,
        btav_a2dp_codec_channel_mode_t codec_channel_mode) {
  size_t buffer_sz = AUDIO_STREAM_OUTPUT_BUFFER_SZ;  // Default value
  const uint64_t time_period_ms = 20;                // Conservative 20ms
  uint32_t sample_rate;
  uint32_t bits_per_sample;
  uint32_t number_of_channels;

  // Check the codec config sample rate
  switch (codec_sample_rate) {
    case BTAV_A2DP_CODEC_SAMPLE_RATE_44100:
      sample_rate = 44100;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_48000:
      sample_rate = 48000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_88200:
      sample_rate = 88200;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_96000:
      sample_rate = 96000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_176400:
      sample_rate = 176400;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_192000:
      sample_rate = 192000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_16000:
      sample_rate = 16000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_24000:
      sample_rate = 24000;
      break;
    case BTAV_A2DP_CODEC_SAMPLE_RATE_NONE:
    default:
      ERROR("Invalid sample rate: 0x%x", codec_sample_rate);
      return buffer_sz;
  }

  // Check the codec config bits per sample
  switch (codec_bits_per_sample) {
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16:
      bits_per_sample = 16;
      break;
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24:
      bits_per_sample = 24;
      break;
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32:
      bits_per_sample = 32;
      break;
    case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE:
    default:
      ERROR("Invalid bits per sample: 0x%x", codec_bits_per_sample);
      return buffer_sz;
  }

  // Check the codec config channel mode
  switch (codec_channel_mode) {
    case BTAV_A2DP_CODEC_CHANNEL_MODE_MONO:
      number_of_channels = 1;
      break;
    case BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO:
      number_of_channels = 2;
      break;
    case BTAV_A2DP_CODEC_CHANNEL_MODE_NONE:
    default:
      ERROR("Invalid channel mode: 0x%x", codec_channel_mode);
      return buffer_sz;
  }

  //
  // The buffer size is computed by using the following formula:
  //
  // AUDIO_STREAM_OUTPUT_BUFFER_SIZE =
  //    (TIME_PERIOD_MS * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS *
  //     SAMPLE_RATE_HZ * NUMBER_OF_CHANNELS * (BITS_PER_SAMPLE / 8)) / 1000
  //
  // AUDIO_STREAM_OUTPUT_BUFFER_PERIODS controls how the socket buffer is
  // divided for AudioFlinger data delivery. The AudioFlinger mixer delivers
  // data in chunks of
  // (AUDIO_STREAM_OUTPUT_BUFFER_SIZE / AUDIO_STREAM_OUTPUT_BUFFER_PERIODS) .
  // If the number of periods is 2, the socket buffer represents "double
  // buffering" of the AudioFlinger mixer buffer.
  //
  // Furthermore, the AudioFlinger expects the buffer size to be a multiple
  // of 16 frames.
  const size_t divisor =
          (AUDIO_STREAM_OUTPUT_BUFFER_PERIODS * 16 * number_of_channels * bits_per_sample) / 8;

  buffer_sz = (time_period_ms * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS * sample_rate *
               number_of_channels * (bits_per_sample / 8)) /
              1000;

  // Adjust the buffer size so it can be divided by the divisor
  const size_t remainder = buffer_sz % divisor;
  if (remainder != 0) {
    buffer_sz += divisor - remainder;
  }

  return buffer_sz;
}

static audio_channel_mask_t out_get_channels(const struct audio_stream* stream) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  DEBUG("channels 0x%" PRIx32, out->common.cfg.channel_mask);

  return (audio_channel_mask_t)out->common.cfg.channel_mask;
}

static audio_format_t out_get_format(const struct audio_stream* stream) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;
  DEBUG("format 0x%x", out->common.cfg.format);
  return (audio_format_t)out->common.cfg.format;
}

static int out_set_format(UNUSED_ATTR struct audio_stream* stream,
                          UNUSED_ATTR audio_format_t format) {
  DEBUG("setting format not yet supported (0x%x)", format);
  return -ENOSYS;
}

static int out_standby(struct audio_stream* stream) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;
  int retVal = 0;

  FNLOG();

  std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
  // Do nothing in SUSPENDED state.
  if (out->common.state != AUDIO_HA_STATE_SUSPENDED) {
    retVal = suspend_audio_datapath(&out->common, true);
  }
  out->frames_rendered = 0;  // rendered is reset, presented is not

  return retVal;
}

static int out_dump(UNUSED_ATTR const struct audio_stream* stream, UNUSED_ATTR int fd) {
  FNLOG();
  return 0;
}

static int out_set_parameters(struct audio_stream* stream, const char* kvpairs) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  INFO("state %d kvpairs %s", out->common.state, kvpairs);

  std::unordered_map<std::string, std::string> params =
          hash_map_utils_new_from_string_params(kvpairs);
  int status = 0;

  if (params.empty()) {
    return status;
  }

  std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);

  /* dump params */
  hash_map_utils_dump_string_keys_string_values(params);

  if (params[AUDIO_PARAMETER_KEY_CLOSING].compare("true") == 0) {
    DEBUG("stream closing, disallow any writes");
    out->common.state = AUDIO_HA_STATE_STOPPING;
  }

  if (params["HearingAidSuspended"].compare("true") == 0) {
    if (out->common.state == AUDIO_HA_STATE_STARTED) {
      status = suspend_audio_datapath(&out->common, false);
    }
  } else {
    /* Do not start the streaming automatically. If the phone was streaming
     * prior to being suspended, the next out_write shall trigger the
     * AVDTP start procedure */
    if (out->common.state == AUDIO_HA_STATE_SUSPENDED) {
      out->common.state = AUDIO_HA_STATE_STANDBY;
    }
    /* Irrespective of the state, return 0 */
  }

  return status;
}

static char* out_get_parameters(const struct audio_stream* stream, const char* keys) {
  FNLOG();

  btav_a2dp_codec_config_t codec_config;
  btav_a2dp_codec_config_t codec_capability;

  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  std::unordered_map<std::string, std::string> params = hash_map_utils_new_from_string_params(keys);
  std::unordered_map<std::string, std::string> return_params;

  if (params.empty()) {
    return strdup("");
  }

  std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);

  if (ha_read_output_audio_config(&out->common, &codec_config, &codec_capability,
                                  false /* update_stream_config */) < 0) {
    ERROR("ha_read_output_audio_config failed");
    goto done;
  }

  // Add the format
  if (params.find(AUDIO_PARAMETER_STREAM_SUP_FORMATS) != params.end()) {
    std::string param;
    if (codec_capability.bits_per_sample & BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16) {
      if (!param.empty()) {
        param += "|";
      }
      param += "AUDIO_FORMAT_PCM_16_BIT";
    }
    if (codec_capability.bits_per_sample & BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24) {
      if (!param.empty()) {
        param += "|";
      }
      param += "AUDIO_FORMAT_PCM_24_BIT_PACKED";
    }
    if (codec_capability.bits_per_sample & BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32) {
      if (!param.empty()) {
        param += "|";
      }
      param += "AUDIO_FORMAT_PCM_32_BIT";
    }
    if (param.empty()) {
      ERROR("Invalid codec capability bits_per_sample=0x%x", codec_capability.bits_per_sample);
      goto done;
    } else {
      return_params[AUDIO_PARAMETER_STREAM_SUP_FORMATS] = param;
    }
  }

  // Add the sample rate
  if (params.find(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES) != params.end()) {
    std::string param;
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_44100) {
      if (!param.empty()) {
        param += "|";
      }
      param += "44100";
    }
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_48000) {
      if (!param.empty()) {
        param += "|";
      }
      param += "48000";
    }
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_88200) {
      if (!param.empty()) {
        param += "|";
      }
      param += "88200";
    }
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_96000) {
      if (!param.empty()) {
        param += "|";
      }
      param += "96000";
    }
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_176400) {
      if (!param.empty()) {
        param += "|";
      }
      param += "176400";
    }
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_192000) {
      if (!param.empty()) {
        param += "|";
      }
      param += "192000";
    }
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_16000) {
      if (!param.empty()) {
        param += "|";
      }
      param += "16000";
    }
    if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_24000) {
      if (!param.empty()) {
        param += "|";
      }
      param += "24000";
    }
    if (param.empty()) {
      ERROR("Invalid codec capability sample_rate=0x%x", codec_capability.sample_rate);
      goto done;
    } else {
      return_params[AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES] = param;
    }
  }

  // Add the channel mask
  if (params.find(AUDIO_PARAMETER_STREAM_SUP_CHANNELS) != params.end()) {
    std::string param;
    if (codec_capability.channel_mode & BTAV_A2DP_CODEC_CHANNEL_MODE_MONO) {
      if (!param.empty()) {
        param += "|";
      }
      param += "AUDIO_CHANNEL_OUT_MONO";
    }
    if (codec_capability.channel_mode & BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO) {
      if (!param.empty()) {
        param += "|";
      }
      param += "AUDIO_CHANNEL_OUT_STEREO";
    }
    if (param.empty()) {
      ERROR("Invalid codec capability channel_mode=0x%x", codec_capability.channel_mode);
      goto done;
    } else {
      return_params[AUDIO_PARAMETER_STREAM_SUP_CHANNELS] = param;
    }
  }

done:
  std::string result;
  for (const auto& ptr : return_params) {
    result += ptr.first + "=" + ptr.second + ";";
  }

  INFO("get parameters result = %s", result.c_str());

  return strdup(result.c_str());
}

static uint32_t out_get_latency(const struct audio_stream_out* stream) {
  int latency_us;

  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  FNLOG();

  latency_us = ((out->common.buffer_sz * 1000) / audio_stream_out_frame_size(&out->stream) /
                out->common.cfg.rate) *
               1000;

  return (latency_us / 1000) + 200;
}

static int out_set_volume(UNUSED_ATTR struct audio_stream_out* stream, UNUSED_ATTR float left,
                          UNUSED_ATTR float right) {
  FNLOG();

  /* volume controlled in audioflinger mixer (digital) */

  return -ENOSYS;
}

static int out_get_presentation_position(const struct audio_stream_out* stream, uint64_t* frames,
                                         struct timespec* timestamp) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  FNLOG();
  if (stream == NULL || frames == NULL || timestamp == NULL) {
    return -EINVAL;
  }

  int ret = -EWOULDBLOCK;
  std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
  uint64_t latency_frames = (uint64_t)out_get_latency(stream) * out->common.cfg.rate / 1000;
  if (out->frames_presented >= latency_frames) {
    *frames = out->frames_presented - latency_frames;
    clock_gettime(CLOCK_MONOTONIC,
                  timestamp);  // could also be associated with out_write().
    ret = 0;
  }
  return ret;
}

static int out_get_render_position(const struct audio_stream_out* stream, uint32_t* dsp_frames) {
  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  FNLOG();
  if (stream == NULL || dsp_frames == NULL) {
    return -EINVAL;
  }

  std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
  uint64_t latency_frames = (uint64_t)out_get_latency(stream) * out->common.cfg.rate / 1000;
  if (out->frames_rendered >= latency_frames) {
    *dsp_frames = (uint32_t)(out->frames_rendered - latency_frames);
  } else {
    *dsp_frames = 0;
  }
  return 0;
}

static int out_add_audio_effect(UNUSED_ATTR const struct audio_stream* stream,
                                UNUSED_ATTR effect_handle_t effect) {
  FNLOG();
  return 0;
}

static int out_remove_audio_effect(UNUSED_ATTR const struct audio_stream* stream,
                                   UNUSED_ATTR effect_handle_t effect) {
  FNLOG();
  return 0;
}

/*
 * AUDIO INPUT STREAM
 */

static uint32_t in_get_sample_rate(const struct audio_stream* stream) {
  struct ha_stream_in* in = (struct ha_stream_in*)stream;

  FNLOG();
  return in->common.cfg.rate;
}

static int in_set_sample_rate(struct audio_stream* stream, uint32_t rate) {
  struct ha_stream_in* in = (struct ha_stream_in*)stream;

  FNLOG();

  if (in->common.cfg.rate > 0 && in->common.cfg.rate == rate) {
    return 0;
  } else {
    return -1;
  }
}

static size_t in_get_buffer_size(UNUSED_ATTR const struct audio_stream* stream) {
  FNLOG();
  return 320;
}

static audio_channel_mask_t in_get_channels(const struct audio_stream* stream) {
  struct ha_stream_in* in = (struct ha_stream_in*)stream;

  FNLOG();
  return (audio_channel_mask_t)in->common.cfg.channel_mask;
}

static audio_format_t in_get_format(UNUSED_ATTR const struct audio_stream* stream) {
  FNLOG();
  return AUDIO_FORMAT_PCM_16_BIT;
}

static int in_set_format(UNUSED_ATTR struct audio_stream* stream,
                         UNUSED_ATTR audio_format_t format) {
  FNLOG();
  if (format == AUDIO_FORMAT_PCM_16_BIT) {
    return 0;
  } else {
    return -1;
  }
}

static int in_standby(UNUSED_ATTR struct audio_stream* stream) {
  FNLOG();
  return 0;
}

static int in_dump(UNUSED_ATTR const struct audio_stream* stream, UNUSED_ATTR int fd) {
  FNLOG();
  return 0;
}

static int in_set_parameters(UNUSED_ATTR struct audio_stream* stream,
                             UNUSED_ATTR const char* kvpairs) {
  FNLOG();
  return 0;
}

static char* in_get_parameters(UNUSED_ATTR const struct audio_stream* stream,
                               UNUSED_ATTR const char* keys) {
  FNLOG();
  return strdup("");
}

static int in_set_gain(UNUSED_ATTR struct audio_stream_in* stream, UNUSED_ATTR float gain) {
  FNLOG();
  return 0;
}

static ssize_t in_read(struct audio_stream_in* stream, void* buffer, size_t bytes) {
  struct ha_stream_in* in = (struct ha_stream_in*)stream;
  int read;
  int us_delay;

  DEBUG("read %zu bytes, state: %d", bytes, in->common.state);

  std::unique_lock<std::recursive_mutex> lock(*in->common.mutex);
  if (in->common.state == AUDIO_HA_STATE_SUSPENDED || in->common.state == AUDIO_HA_STATE_STOPPING) {
    DEBUG("stream suspended");
    goto error;
  }

  /* only allow autostarting if we are in stopped or standby */
  if ((in->common.state == AUDIO_HA_STATE_STOPPED) ||
      (in->common.state == AUDIO_HA_STATE_STANDBY)) {
    if (start_audio_datapath(&in->common) < 0) {
      goto error;
    }
  } else if (in->common.state != AUDIO_HA_STATE_STARTED) {
    ERROR("stream not in stopped or standby");
    goto error;
  }

  lock.unlock();
  read = skt_read(in->common.audio_fd, buffer, bytes);
  lock.lock();
  if (read == -1) {
    skt_disconnect(in->common.audio_fd);
    in->common.audio_fd = AUDIO_SKT_DISCONNECTED;
    if ((in->common.state != AUDIO_HA_STATE_SUSPENDED) &&
        (in->common.state != AUDIO_HA_STATE_STOPPING)) {
      in->common.state = AUDIO_HA_STATE_STOPPED;
    } else {
      ERROR("read failed : stream suspended, avoid resetting state");
    }
    goto error;
  } else if (read == 0) {
    DEBUG("read time out - return zeros");
    memset(buffer, 0, bytes);
    read = bytes;
  }
  lock.unlock();

  DEBUG("read %d bytes out of %zu bytes", read, bytes);
  return read;

error:
  memset(buffer, 0, bytes);
  us_delay = calc_audiotime_usec(in->common.cfg, bytes);
  DEBUG("emulate ha read delay (%d us)", us_delay);

  usleep(us_delay);
  return bytes;
}

static uint32_t in_get_input_frames_lost(UNUSED_ATTR struct audio_stream_in* stream) {
  FNLOG();
  return 0;
}

static int in_add_audio_effect(UNUSED_ATTR const struct audio_stream* stream,
                               UNUSED_ATTR effect_handle_t effect) {
  FNLOG();
  return 0;
}

static int in_remove_audio_effect(UNUSED_ATTR const struct audio_stream* stream,
                                  UNUSED_ATTR effect_handle_t effect) {
  FNLOG();

  return 0;
}

static int adev_open_output_stream(struct audio_hw_device* dev,
                                   UNUSED_ATTR audio_io_handle_t handle,
                                   UNUSED_ATTR audio_devices_t devices,
                                   UNUSED_ATTR audio_output_flags_t flags,
                                   struct audio_config* config,
                                   struct audio_stream_out** stream_out,
                                   UNUSED_ATTR const char* address)

{
  struct ha_audio_device* ha_dev = (struct ha_audio_device*)dev;
  struct ha_stream_out* out;
  int ret = 0;

  INFO("opening output");
  // protect against adev->output and stream_out from being inconsistent
  std::lock_guard<std::recursive_mutex> lock(*ha_dev->mutex);
  out = (struct ha_stream_out*)calloc(1, sizeof(struct ha_stream_out));

  if (!out) {
    return -ENOMEM;
  }

  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_presentation_position = out_get_presentation_position;

  /* initialize ha specifics */
  ha_stream_common_init(&out->common);

  // Make sure we always have the feeding parameters configured
  btav_a2dp_codec_config_t codec_config;
  btav_a2dp_codec_config_t codec_capability;
  if (ha_read_output_audio_config(&out->common, &codec_config, &codec_capability,
                                  true /* update_stream_config */) < 0) {
    ERROR("ha_read_output_audio_config failed");
    ret = -1;
    goto err_open;
  }
  // ha_read_output_audio_config() opens the socket control path (or fails)

  /* set output config values */
  if (config != nullptr) {
    // Try to use the config parameters and send it to the remote side
    // TODO: Shall we use out_set_format() and similar?
    if (config->format != 0) {
      out->common.cfg.format = config->format;
    }
    if (config->sample_rate != 0) {
      out->common.cfg.rate = config->sample_rate;
    }
    if (config->channel_mask != 0) {
      out->common.cfg.channel_mask = config->channel_mask;
    }
    if ((out->common.cfg.format != 0) || (out->common.cfg.rate != 0) ||
        (out->common.cfg.channel_mask != 0)) {
      if (ha_write_output_audio_config(&out->common) < 0) {
        ERROR("ha_write_output_audio_config failed");
        ret = -1;
        goto err_open;
      }
      // Read again and make sure we use the same parameters as the remote side
      if (ha_read_output_audio_config(&out->common, &codec_config, &codec_capability,
                                      true /* update_stream_config */) < 0) {
        ERROR("ha_read_output_audio_config failed");
        ret = -1;
        goto err_open;
      }
    }
    config->format = out_get_format((const struct audio_stream*)&out->stream);
    config->sample_rate = out_get_sample_rate((const struct audio_stream*)&out->stream);
    config->channel_mask = out_get_channels((const struct audio_stream*)&out->stream);

    INFO("Output stream config: format=0x%x sample_rate=%d channel_mask=0x%x buffer_sz=%zu",
         config->format, config->sample_rate, config->channel_mask, out->common.buffer_sz);
  }
  *stream_out = &out->stream;
  ha_dev->output = out;

  DEBUG("success");
  /* Delay to ensure Headset is in proper state when START is initiated from
   * DUT immediately after the connection due to ongoing music playback. */
  usleep(250000);
  return 0;

err_open:
  ha_stream_common_destroy(&out->common);
  free(out);
  *stream_out = NULL;
  ha_dev->output = NULL;
  ERROR("failed");
  return ret;
}

static void adev_close_output_stream(struct audio_hw_device* dev, struct audio_stream_out* stream) {
  struct ha_audio_device* ha_dev = (struct ha_audio_device*)dev;
  struct ha_stream_out* out = (struct ha_stream_out*)stream;

  // prevent interference with adev_set_parameters.
  std::lock_guard<std::recursive_mutex> lock(*ha_dev->mutex);
  {
    std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
    const ha_state_t state = out->common.state;
    INFO("closing output (state %d)", (int)state);
    if ((state == AUDIO_HA_STATE_STARTED) || (state == AUDIO_HA_STATE_STOPPING)) {
      stop_audio_datapath(&out->common);
    }

    skt_disconnect(out->common.ctrl_fd);
    out->common.ctrl_fd = AUDIO_SKT_DISCONNECTED;
  }

  ha_stream_common_destroy(&out->common);
  free(stream);
  ha_dev->output = NULL;

  DEBUG("done");
}

static int adev_set_parameters(struct audio_hw_device* dev, const char* kvpairs) {
  struct ha_audio_device* ha_dev = (struct ha_audio_device*)dev;
  int retval = 0;

  // prevent interference with adev_close_output_stream
  std::lock_guard<std::recursive_mutex> lock(*ha_dev->mutex);
  struct ha_stream_out* out = ha_dev->output;

  if (out == NULL) {
    return retval;
  }

  INFO("state %d", out->common.state);

  retval = out->stream.common.set_parameters((struct audio_stream*)out, kvpairs);

  return retval;
}

static char* adev_get_parameters(UNUSED_ATTR const struct audio_hw_device* dev, const char* keys) {
  FNLOG();

  std::unordered_map<std::string, std::string> params = hash_map_utils_new_from_string_params(keys);
  hash_map_utils_dump_string_keys_string_values(params);

  return strdup("");
}

static int adev_init_check(UNUSED_ATTR const struct audio_hw_device* dev) {
  FNLOG();

  return 0;
}

static int adev_set_voice_volume(UNUSED_ATTR struct audio_hw_device* dev,
                                 UNUSED_ATTR float volume) {
  FNLOG();

  return -ENOSYS;
}

static int adev_set_master_volume(UNUSED_ATTR struct audio_hw_device* dev,
                                  UNUSED_ATTR float volume) {
  FNLOG();

  return -ENOSYS;
}

static int adev_set_mode(UNUSED_ATTR struct audio_hw_device* dev, UNUSED_ATTR audio_mode_t mode) {
  FNLOG();

  return 0;
}

static int adev_set_mic_mute(UNUSED_ATTR struct audio_hw_device* dev, UNUSED_ATTR bool state) {
  FNLOG();

  return -ENOSYS;
}

static int adev_get_mic_mute(UNUSED_ATTR const struct audio_hw_device* dev,
                             UNUSED_ATTR bool* state) {
  FNLOG();

  return -ENOSYS;
}

static size_t adev_get_input_buffer_size(UNUSED_ATTR const struct audio_hw_device* dev,
                                         UNUSED_ATTR const struct audio_config* config) {
  FNLOG();

  return 320;
}

static int adev_open_input_stream(struct audio_hw_device* dev, UNUSED_ATTR audio_io_handle_t handle,
                                  UNUSED_ATTR audio_devices_t devices,
                                  UNUSED_ATTR struct audio_config* config,
                                  struct audio_stream_in** stream_in,
                                  UNUSED_ATTR audio_input_flags_t flags,
                                  UNUSED_ATTR const char* address,
                                  UNUSED_ATTR audio_source_t source) {
  struct ha_audio_device* ha_dev = (struct ha_audio_device*)dev;
  struct ha_stream_in* in;
  int ret;

  FNLOG();

  // protect against adev->input and stream_in from being inconsistent
  std::lock_guard<std::recursive_mutex> lock(*ha_dev->mutex);
  in = (struct ha_stream_in*)calloc(1, sizeof(struct ha_stream_in));

  if (!in) {
    return -ENOMEM;
  }

  in->stream.common.get_sample_rate = in_get_sample_rate;
  in->stream.common.set_sample_rate = in_set_sample_rate;
  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;

  /* initialize ha specifics */
  ha_stream_common_init(&in->common);

  *stream_in = &in->stream;
  ha_dev->input = in;

  if (ha_read_input_audio_config(&in->common) < 0) {
    ERROR("ha_read_input_audio_config failed (%s)", strerror(errno));
    ret = -1;
    goto err_open;
  }
  // ha_read_input_audio_config() opens socket control path (or fails)

  DEBUG("success");
  return 0;

err_open:
  ha_stream_common_destroy(&in->common);
  free(in);
  *stream_in = NULL;
  ha_dev->input = NULL;
  ERROR("failed");
  return ret;
}

static void adev_close_input_stream(struct audio_hw_device* dev, struct audio_stream_in* stream) {
  struct ha_audio_device* ha_dev = (struct ha_audio_device*)dev;
  struct ha_stream_in* in = (struct ha_stream_in*)stream;

  std::lock_guard<std::recursive_mutex> lock(*ha_dev->mutex);
  {
    std::lock_guard<std::recursive_mutex> lock(*in->common.mutex);
    const ha_state_t state = in->common.state;
    INFO("closing input (state %d)", (int)state);

    if ((state == AUDIO_HA_STATE_STARTED) || (state == AUDIO_HA_STATE_STOPPING)) {
      stop_audio_datapath(&in->common);
    }

    skt_disconnect(in->common.ctrl_fd);
    in->common.ctrl_fd = AUDIO_SKT_DISCONNECTED;
  }
  ha_stream_common_destroy(&in->common);
  free(stream);
  ha_dev->input = NULL;

  DEBUG("done");
}

static int adev_dump(UNUSED_ATTR const audio_hw_device_t* device, UNUSED_ATTR int fd) {
  FNLOG();

  return 0;
}

static int adev_close(hw_device_t* device) {
  struct ha_audio_device* ha_dev = (struct ha_audio_device*)device;
  FNLOG();

  delete ha_dev->mutex;
  ha_dev->mutex = nullptr;
  free(device);
  return 0;
}

static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device) {
  struct ha_audio_device* adev;

  INFO(" adev_open in ha_hw module");
  FNLOG();

  if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) {
    ERROR("interface %s not matching [%s]", name, AUDIO_HARDWARE_INTERFACE);
    return -EINVAL;
  }

  adev = (struct ha_audio_device*)calloc(1, sizeof(struct ha_audio_device));

  if (!adev) {
    return -ENOMEM;
  }

  adev->mutex = new std::recursive_mutex;

  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.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->output = NULL;

  *device = &adev->device.common;

  return 0;
}

static struct hw_module_methods_t hal_module_methods = {
        .open = adev_open,
};

__attribute__((visibility("default"))) struct audio_module HAL_MODULE_INFO_SYM = {
        .common =
                {
                        .tag = HARDWARE_MODULE_TAG,
                        .version_major = 1,
                        .version_minor = 0,
                        .id = AUDIO_HARDWARE_MODULE_ID,
                        .name = "Hearing Aid Audio HW HAL",
                        .author = "The Android Open Source Project",
                        .methods = &hal_module_methods,
                },
};