--- /dev/null
+/*
+ * 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.
+ */
+
+#define LOG_TAG "r_submix"
+//#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 <media/nbaio/Pipe.h>
+#include <media/nbaio/PipeReader.h>
+#include <media/AudioBufferProvider.h>
+
+extern "C" {
+
+namespace android {
+
+#define MAX_PIPE_DEPTH_IN_FRAMES (1024*4)
+#define MAX_READ_ATTEMPTS 10
+#define READ_ATTEMPT_SLEEP_MS 10 // 10ms between two read attempts when pipe is empty
+#define DEFAULT_RATE_HZ 48000 // default sample rate
+
+struct submix_config {
+ audio_format_t format;
+ audio_channel_mask_t channel_mask;
+ unsigned int rate; // sample rate for the device
+ unsigned int period_size; // size of the audio pipe is period_size * period_count in frames
+ unsigned int period_count;
+};
+
+struct submix_audio_device {
+ struct audio_hw_device device;
+ submix_config config;
+ // Pipe variables: they handle the ring buffer that "pipes" audio:
+ // - from the submix virtual audio output == what needs to be played by
+ // the remotely, seen as an output for AudioFlinger
+ // - to the virtual audio source == what is captured by the component
+ // which "records" the submix / virtual audio source, and handles it as needed.
+ // An usecase example is one where the component capturing the audio is then sending it over
+ // Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a
+ // TV with Wifi Display capabilities), or to a wireless audio player.
+ sp<Pipe> rsxSink;
+ sp<PipeReader> rsxSource;
+
+ pthread_mutex_t lock;
+};
+
+struct submix_stream_out {
+ struct audio_stream_out stream;
+ struct submix_audio_device *dev;
+};
+
+struct submix_stream_in {
+ struct audio_stream_in stream;
+ struct submix_audio_device *dev;
+};
+
+
+/* audio HAL functions */
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+ const struct submix_stream_out *out =
+ reinterpret_cast<const struct submix_stream_out *>(stream);
+ uint32_t out_rate = out->dev->config.rate;
+ //ALOGV("out_get_sample_rate() returns %u", out_rate);
+ return out_rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+ if ((rate != 44100) && (rate != 48000)) {
+ ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
+ return -ENOSYS;
+ }
+ struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream);
+ //ALOGV("out_set_sample_rate(rate=%u)", rate);
+ out->dev->config.rate = rate;
+ return 0;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+ const struct submix_stream_out *out =
+ reinterpret_cast<const struct submix_stream_out *>(stream);
+ const struct submix_config& config_out = out->dev->config;
+ size_t buffer_size = config_out.period_size * popcount(config_out.channel_mask)
+ * sizeof(int16_t); // only PCM 16bit
+ //ALOGV("out_get_buffer_size() returns %u, period size=%u",
+ // buffer_size, config_out.period_size);
+ return buffer_size;
+}
+
+static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
+{
+ const struct submix_stream_out *out =
+ reinterpret_cast<const struct submix_stream_out *>(stream);
+ uint32_t channels = out->dev->config.channel_mask;
+ //ALOGV("out_get_channels() returns %08x", channels);
+ return channels;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+ return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+ if (format != AUDIO_FORMAT_PCM_16_BIT) {
+ return -ENOSYS;
+ } else {
+ return 0;
+ }
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+ // REMOTE_SUBMIX is a proxy / virtual audio device, so the notion of standby doesn't apply here
+ return 0;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+ return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ return 0;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+ return strdup("");
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+ const struct submix_stream_out *out =
+ reinterpret_cast<const struct submix_stream_out *>(stream);
+ const struct submix_config * config_out = &(out->dev->config);
+ uint32_t latency = (MAX_PIPE_DEPTH_IN_FRAMES * 1000) / config_out->rate;
+ ALOGV("out_get_latency() returns %u", latency);
+ return latency;
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+ float right)
+{
+ return -ENOSYS;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
+ size_t bytes)
+{
+ //ALOGV("out_write(bytes=%d)", bytes);
+ ssize_t written = 0;
+ struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream);
+
+ pthread_mutex_lock(&out->dev->lock);
+
+ Pipe* sink = out->dev->rsxSink.get();
+ if (sink != NULL) {
+ out->dev->rsxSink->incStrong(buffer);
+ } else {
+ pthread_mutex_unlock(&out->dev->lock);
+ ALOGE("out_write without a pipe!");
+ ALOG_ASSERT("out_write without a pipe!");
+ return 0;
+ }
+
+ pthread_mutex_unlock(&out->dev->lock);
+
+ const size_t frames = bytes / audio_stream_frame_size(&stream->common);
+ written = sink->write(buffer, frames);
+ if (written < 0) {
+ if (written == (ssize_t)NEGOTIATE) {
+ ALOGE("out_write() write to pipe returned NEGOTIATE");
+ written = 0;
+ } else {
+ // write() returned UNDERRUN or WOULD_BLOCK, retry
+ written = sink->write(buffer, frames);
+ }
+ }
+
+ pthread_mutex_lock(&out->dev->lock);
+
+ out->dev->rsxSink->decStrong(buffer);
+
+ pthread_mutex_unlock(&out->dev->lock);
+
+ if (written > 0) {
+ // fake timing for audio output, we can't return right after pushing the data in the pipe
+ // TODO who's doing the flow control here? the wifi display link, or the audio HAL?
+ usleep(written * 1000000 / out_get_sample_rate(&stream->common));
+ return written * audio_stream_frame_size(&stream->common);;
+ } else {
+ // error occurred, fake timing
+ usleep(frames * 1000000 / out_get_sample_rate(&stream->common));
+ ALOGE("out_write error=%16lx", written);
+ return 0;
+ }
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+ uint32_t *dsp_frames)
+{
+ return -EINVAL;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+ return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
+{
+ return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
+ int64_t *timestamp)
+{
+ return -EINVAL;
+}
+
+/** audio_stream_in implementation **/
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+ const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
+ ALOGV("in_get_sample_rate() returns %u", in->dev->config.rate);
+ return in->dev->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)
+{
+ const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
+ ALOGV("in_get_buffer_size() returns %u",
+ in->dev->config.period_size * audio_stream_frame_size(stream));
+ return in->dev->config.period_size * audio_stream_frame_size(stream);
+}
+
+static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
+{
+ return AUDIO_CHANNEL_IN_STEREO;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+ return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+ if (format != AUDIO_FORMAT_PCM_16_BIT) {
+ return -ENOSYS;
+ } else {
+ return 0;
+ }
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+ // REMOTE_SUBMIX is a proxy / virtual audio device, so the notion of standby doesn't apply here
+ 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)
+{
+ return 0;
+}
+
+static char * in_get_parameters(const struct audio_stream *stream,
+ const char *keys)
+{
+ return strdup("");
+}
+
+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)
+{
+ ssize_t frames_read = -1977;
+ const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
+ const size_t frame_size = audio_stream_frame_size(&stream->common);
+
+ pthread_mutex_lock(&in->dev->lock);
+
+ PipeReader* source = in->dev->rsxSource.get();
+ if (source != NULL) {
+ in->dev->rsxSource->incStrong(in);
+ } else {
+ pthread_mutex_unlock(&in->dev->lock);
+ usleep((bytes / frame_size) * 1000000 / in_get_sample_rate(&stream->common));
+ memset(buffer, 0, bytes);
+ return bytes;
+ }
+
+ pthread_mutex_unlock(&in->dev->lock);
+
+ int attempts = MAX_READ_ATTEMPTS;
+ size_t remaining_frames = bytes / frame_size;
+ char* buff = (char*)buffer;
+ while (attempts > 0) {
+ frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS);
+ if (frames_read > 0) {
+ //ALOGV("in_read frames=%ld size=%u", remaining_frames, frame_size);
+ remaining_frames -= frames_read;
+ buff += frames_read * frame_size;
+ if (remaining_frames == 0) {
+ // TODO simplify code by breaking out of loop
+
+ pthread_mutex_lock(&in->dev->lock);
+
+ in->dev->rsxSource->decStrong(in);
+
+ pthread_mutex_unlock(&in->dev->lock);
+
+ return bytes;
+ }
+ } else if (frames_read == 0) {
+ // TODO sleep should be tied to how much data is expected
+ usleep(READ_ATTEMPT_SLEEP_MS*1000);
+ attempts--;
+ } else { // frames_read is an error code
+ if (frames_read != (ssize_t)OVERRUN) {
+ attempts--;
+ }
+ // else OVERRUN: error has been signaled, ok to read, do not decrement counter
+ }
+ }
+
+ pthread_mutex_lock(&in->dev->lock);
+
+ in->dev->rsxSource->decStrong(in);
+
+ pthread_mutex_unlock(&in->dev->lock);
+
+ // TODO how to handle partial reads?
+
+ if (frames_read < 0) {
+ ALOGE("in_read error=%16lx", frames_read);
+ }
+ return 0;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+ return 0;
+}
+
+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 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)
+{
+ ALOGV("adev_open_output_stream()");
+ struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+ struct submix_stream_out *out;
+ int ret;
+
+ out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out));
+ if (!out) {
+ ret = -ENOMEM;
+ goto err_open;
+ }
+
+ pthread_mutex_lock(&rsxadev->lock);
+
+ 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->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ rsxadev->config.channel_mask = config->channel_mask;
+
+ if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) {
+ config->sample_rate = DEFAULT_RATE_HZ;
+ }
+ rsxadev->config.rate = config->sample_rate;
+
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ rsxadev->config.format = config->format;
+
+ rsxadev->config.period_size = 1024;
+ rsxadev->config.period_count = 4;
+ out->dev = rsxadev;
+
+ *stream_out = &out->stream;
+
+ // initialize pipe
+ {
+ ALOGV(" initializing pipe");
+ const NBAIO_Format format =
+ config->sample_rate == 48000 ? Format_SR48_C2_I16 : Format_SR44_1_C2_I16;
+ const NBAIO_Format offers[1] = {format};
+ size_t numCounterOffers = 0;
+ // creating a Pipe, not a MonoPipe with optional blocking set to true, so audio frames
+ // entering a full sink will overwrite the contents of the pipe.
+ Pipe* sink = new Pipe(MAX_PIPE_DEPTH_IN_FRAMES, format);
+ ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+ PipeReader* source = new PipeReader(*sink);
+ numCounterOffers = 0;
+ index = source->negotiate(offers, 1, NULL, numCounterOffers);
+ ALOG_ASSERT(index == 0);
+ rsxadev->rsxSink = sink;
+ rsxadev->rsxSource = source;
+ }
+
+ pthread_mutex_unlock(&rsxadev->lock);
+
+ return 0;
+
+err_open:
+ *stream_out = NULL;
+ return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+ struct audio_stream_out *stream)
+{
+ ALOGV("adev_close_output_stream()");
+ struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+
+ pthread_mutex_lock(&rsxadev->lock);
+
+ rsxadev->rsxSink.clear();
+ rsxadev->rsxSource.clear();
+ free(stream);
+
+ pthread_mutex_unlock(&rsxadev->lock);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+ return -ENOSYS;
+}
+
+static char * adev_get_parameters(const struct audio_hw_device *dev,
+ const char *keys)
+{
+ return strdup("");;
+}
+
+static int adev_init_check(const struct audio_hw_device *dev)
+{
+ ALOGI("adev_init_check()");
+ return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+ return -ENOSYS;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+ return -ENOSYS;
+}
+
+static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
+{
+ return -ENOSYS;
+}
+
+static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
+{
+ return -ENOSYS;
+}
+
+static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
+{
+ return -ENOSYS;
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
+{
+ return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+ return -ENOSYS;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+ return -ENOSYS;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+ const struct audio_config *config)
+{
+ //### TODO correlate this with pipe parameters
+ return 4096;
+}
+
+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)
+{
+ ALOGI("adev_open_input_stream()");
+
+ struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+ struct submix_stream_in *in;
+ int ret;
+
+ in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in));
+ if (!in) {
+ ret = -ENOMEM;
+ goto err_open;
+ }
+
+ pthread_mutex_lock(&rsxadev->lock);
+
+ 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;
+
+ config->channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ rsxadev->config.channel_mask = config->channel_mask;
+
+ if ((config->sample_rate != 48000) || (config->sample_rate != 44100)) {
+ config->sample_rate = DEFAULT_RATE_HZ;
+ }
+ rsxadev->config.rate = config->sample_rate;
+
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ rsxadev->config.format = config->format;
+
+ rsxadev->config.period_size = 1024;
+ rsxadev->config.period_count = 4;
+
+ *stream_in = &in->stream;
+
+ in->dev = rsxadev;
+
+ pthread_mutex_unlock(&rsxadev->lock);
+
+ return 0;
+
+err_open:
+ *stream_in = NULL;
+ return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+ struct audio_stream_in *stream)
+{
+ ALOGV("adev_close_input_stream()");
+ struct submix_audio_device *rsxadev = (struct submix_audio_device *)dev;
+
+ pthread_mutex_lock(&rsxadev->lock);
+
+ free(stream);
+
+ pthread_mutex_unlock(&rsxadev->lock);
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+ return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+ ALOGI("adev_close()");
+ free(device);
+ return 0;
+}
+
+static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev)
+{
+ ALOGI("adev_get_supported_devices() returns %08x",
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX |AUDIO_DEVICE_IN_REMOTE_SUBMIX);
+ return (/* OUT */
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX |
+ /* IN */
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX);
+}
+
+static int adev_open(const hw_module_t* module, const char* name,
+ hw_device_t** device)
+{
+ ALOGI("adev_open(name=%s)", name);
+ struct submix_audio_device *rsxadev;
+
+ if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+ return -EINVAL;
+
+ rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device));
+ if (!rsxadev)
+ return -ENOMEM;
+
+ rsxadev->device.common.tag = HARDWARE_DEVICE_TAG;
+ rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_1_0;
+ rsxadev->device.common.module = (struct hw_module_t *) module;
+ rsxadev->device.common.close = adev_close;
+
+ rsxadev->device.get_supported_devices = adev_get_supported_devices;
+ rsxadev->device.init_check = adev_init_check;
+ rsxadev->device.set_voice_volume = adev_set_voice_volume;
+ rsxadev->device.set_master_volume = adev_set_master_volume;
+ rsxadev->device.get_master_volume = adev_get_master_volume;
+ rsxadev->device.set_master_mute = adev_set_master_mute;
+ rsxadev->device.get_master_mute = adev_get_master_mute;
+ rsxadev->device.set_mode = adev_set_mode;
+ rsxadev->device.set_mic_mute = adev_set_mic_mute;
+ rsxadev->device.get_mic_mute = adev_get_mic_mute;
+ rsxadev->device.set_parameters = adev_set_parameters;
+ rsxadev->device.get_parameters = adev_get_parameters;
+ rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size;
+ rsxadev->device.open_output_stream = adev_open_output_stream;
+ rsxadev->device.close_output_stream = adev_close_output_stream;
+ rsxadev->device.open_input_stream = adev_open_input_stream;
+ rsxadev->device.close_input_stream = adev_close_input_stream;
+ rsxadev->device.dump = adev_dump;
+
+ *device = &rsxadev->device.common;
+
+ return 0;
+}
+
+static struct hw_module_methods_t hal_module_methods = {
+ /* open */ adev_open,
+};
+
+struct audio_module HAL_MODULE_INFO_SYM = {
+ /* common */ {
+ /* tag */ HARDWARE_MODULE_TAG,
+ /* module_api_version */ AUDIO_MODULE_API_VERSION_0_1,
+ /* hal_api_version */ HARDWARE_HAL_API_VERSION,
+ /* id */ AUDIO_HARDWARE_MODULE_ID,
+ /* name */ "Wifi Display audio HAL",
+ /* author */ "The Android Open Source Project",
+ /* methods */ &hal_module_methods,
+ /* dso */ NULL,
+ /* reserved */ { 0 },
+ },
+};
+
+} //namespace android
+
+} //extern "C"
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