/*
* 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/MonoPipe.h>
#include <media/nbaio/MonoPipeReader.h>
#include <media/AudioBufferProvider.h>
#include <utils/String8.h>
#include <media/AudioParameter.h>
extern "C" {
namespace android {
#define MAX_PIPE_DEPTH_IN_FRAMES (1024*8)
// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to
// the duration of a record buffer at the current record sample rate (of the device, not of
// the recording itself). Here we have:
// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms
#define MAX_READ_ATTEMPTS 3
#define READ_ATTEMPT_SLEEP_MS 5 // 5ms 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;
bool output_standby;
bool input_standby;
submix_config config;
// Pipe variables: they handle the ring buffer that "pipes" audio:
// - from the submix virtual audio output == what needs to be played
// 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.
// A 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<MonoPipe> rsxSink;
sp<MonoPipeReader> rsxSource;
// device lock, also used to protect access to the audio pipe
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;
bool output_standby; // output standby state as seen from record thread
// wall clock when recording starts
struct timespec record_start_time;
// how many frames have been requested to be read
int64_t read_counter_frames;
};
/* 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)
{
ALOGI("out_standby()");
const struct submix_stream_out *out = reinterpret_cast<const struct submix_stream_out *>(stream);
pthread_mutex_lock(&out->dev->lock);
out->dev->output_standby = true;
pthread_mutex_unlock(&out->dev->lock);
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)
{
int exiting = -1;
AudioParameter parms = AudioParameter(String8(kvpairs));
// FIXME this is using hard-coded strings but in the future, this functionality will be
// converted to use audio HAL extensions required to support tunneling
if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) {
const struct submix_stream_out *out =
reinterpret_cast<const struct submix_stream_out *>(stream);
pthread_mutex_lock(&out->dev->lock);
MonoPipe* sink = out->dev->rsxSink.get();
if (sink != NULL) {
sink->incStrong(out);
} else {
pthread_mutex_unlock(&out->dev->lock);
return 0;
}
ALOGI("shutdown");
sink->shutdown(true);
sink->decStrong(out);
pthread_mutex_unlock(&out->dev->lock);
}
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_frames = 0;
struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream);
const size_t frame_size = audio_stream_frame_size(&stream->common);
const size_t frames = bytes / frame_size;
pthread_mutex_lock(&out->dev->lock);
out->dev->output_standby = false;
MonoPipe* sink = out->dev->rsxSink.get();
if (sink != NULL) {
if (sink->isShutdown()) {
pthread_mutex_unlock(&out->dev->lock);
// the pipe has already been shutdown, this buffer will be lost but we must
// simulate timing so we don't drain the output faster than realtime
usleep(frames * 1000000 / out_get_sample_rate(&stream->common));
return bytes;
}
sink->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);
written_frames = sink->write(buffer, frames);
if (written_frames < 0) {
if (written_frames == (ssize_t)NEGOTIATE) {
ALOGE("out_write() write to pipe returned NEGOTIATE");
pthread_mutex_lock(&out->dev->lock);
sink->decStrong(buffer);
pthread_mutex_unlock(&out->dev->lock);
written_frames = 0;
return 0;
} else {
// write() returned UNDERRUN or WOULD_BLOCK, retry
ALOGE("out_write() write to pipe returned unexpected %16lx", written_frames);
written_frames = sink->write(buffer, frames);
}
}
pthread_mutex_lock(&out->dev->lock);
sink->decStrong(buffer);
pthread_mutex_unlock(&out->dev->lock);
if (written_frames < 0) {
ALOGE("out_write() failed writing to pipe with %16lx", written_frames);
return 0;
} else {
ALOGV("out_write() wrote %lu bytes)", written_frames * frame_size);
return written_frames * frame_size;
}
}
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)
{
ALOGI("in_standby()");
const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
pthread_mutex_lock(&in->dev->lock);
in->dev->input_standby = true;
pthread_mutex_unlock(&in->dev->lock);
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)
{
//ALOGV("in_read bytes=%u", bytes);
ssize_t frames_read = -1977;
struct submix_stream_in *in = reinterpret_cast<struct submix_stream_in *>(stream);
const size_t frame_size = audio_stream_frame_size(&stream->common);
const size_t frames_to_read = bytes / frame_size;
pthread_mutex_lock(&in->dev->lock);
const bool output_standby_transition = (in->output_standby != in->dev->output_standby);
in->output_standby = in->dev->output_standby;
if (in->dev->input_standby || output_standby_transition) {
in->dev->input_standby = false;
// keep track of when we exit input standby (== first read == start "real recording")
// or when we start recording silence, and reset projected time
int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time);
if (rc == 0) {
in->read_counter_frames = 0;
}
}
in->read_counter_frames += frames_to_read;
MonoPipeReader* source = in->dev->rsxSource.get();
if (source != NULL) {
source->incStrong(buffer);
} else {
ALOGE("no audio pipe yet we're trying to read!");
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);
// read the data from the pipe (it's non blocking)
size_t remaining_frames = frames_to_read;
int attempts = 0;
char* buff = (char*)buffer;
while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
attempts++;
frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS);
if (frames_read > 0) {
remaining_frames -= frames_read;
buff += frames_read * frame_size;
//ALOGV(" in_read (att=%d) got %ld frames, remaining=%u",
// attempts, frames_read, remaining_frames);
} else {
//ALOGE(" in_read read returned %ld", frames_read);
usleep(READ_ATTEMPT_SLEEP_MS * 1000);
}
}
// done using the source
pthread_mutex_lock(&in->dev->lock);
source->decStrong(buffer);
pthread_mutex_unlock(&in->dev->lock);
if (remaining_frames > 0) {
ALOGV(" remaining_frames = %d", remaining_frames);
memset(((char*)buffer)+ bytes - (remaining_frames * frame_size), 0,
remaining_frames * frame_size);
}
// compute how much we need to sleep after reading the data by comparing the wall clock with
// the projected time at which we should return.
struct timespec time_after_read;// wall clock after reading from the pipe
struct timespec record_duration;// observed record duration
int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read);
const uint32_t sample_rate = in_get_sample_rate(&stream->common);
if (rc == 0) {
// for how long have we been recording?
record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec;
record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec;
if (record_duration.tv_nsec < 0) {
record_duration.tv_sec--;
record_duration.tv_nsec += 1000000000;
}
// read_counter_frames contains the number of frames that have been read since the beginning
// of recording (including this call): it's converted to usec and compared to how long we've
// been recording for, which gives us how long we must wait to sync the projected recording
// time, and the observed recording time
long projected_vs_observed_offset_us =
((int64_t)(in->read_counter_frames
- (record_duration.tv_sec*sample_rate)))
* 1000000 / sample_rate
- (record_duration.tv_nsec / 1000);
ALOGV(" record duration %5lds %3ldms, will wait: %7ldus",
record_duration.tv_sec, record_duration.tv_nsec/1000000,
projected_vs_observed_offset_us);
if (projected_vs_observed_offset_us > 0) {
usleep(projected_vs_observed_offset_us);
}
}
ALOGV("in_read returns %d", bytes);
return bytes;
}
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 MonoPipe with optional blocking set to true.
MonoPipe* sink = new MonoPipe(MAX_PIPE_DEPTH_IN_FRAMES, format, true/*writeCanBlock*/);
ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
MonoPipeReader* source = new MonoPipeReader(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;
in->read_counter_frames = 0;
in->output_standby = rsxadev->output_standby;
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);
MonoPipe* sink = rsxadev->rsxSink.get();
if (sink != NULL) {
ALOGI("shutdown");
sink->shutdown(true);
}
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 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_2_0;
rsxadev->device.common.module = (struct hw_module_t *) module;
rsxadev->device.common.close = adev_close;
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;
rsxadev->input_standby = true;
rsxadev->output_standby = true;
*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"