/*
* 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 "audio_hw_generic"
#include <assert.h>
#include <errno.h>
#include <inttypes.h>
#include <pthread.h>
#include <stdint.h>
#include <stdlib.h>
#include <sys/time.h>
#include <dlfcn.h>
#include <fcntl.h>
#include <unistd.h>
#include <log/log.h>
#include <cutils/str_parms.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include <tinyalsa/asoundlib.h>
#define PCM_CARD 0
#define PCM_DEVICE 0
#define OUT_PERIOD_MS 15
#define OUT_PERIOD_COUNT 4
#define IN_PERIOD_MS 15
#define IN_PERIOD_COUNT 4
struct generic_audio_device {
struct audio_hw_device device; // Constant after init
pthread_mutex_t lock;
bool mic_mute; // Proteced by this->lock
struct mixer* mixer; // Proteced by this->lock
};
/* If not NULL, this is a pointer to the fallback module.
* This really is the original goldfish audio device /dev/eac which we will use
* if no alsa devices are detected.
*/
static struct audio_module* sFallback;
static pthread_once_t sFallbackOnce = PTHREAD_ONCE_INIT;
static void fallback_init(void);
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state);
static int adev_get_microphones(const audio_hw_device_t *dev,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count);
typedef struct audio_vbuffer {
pthread_mutex_t lock;
uint8_t * data;
size_t frame_size;
size_t frame_count;
size_t head;
size_t tail;
size_t live;
} audio_vbuffer_t;
static int audio_vbuffer_init (audio_vbuffer_t * audio_vbuffer, size_t frame_count,
size_t frame_size) {
if (!audio_vbuffer) {
return -EINVAL;
}
audio_vbuffer->frame_size = frame_size;
audio_vbuffer->frame_count = frame_count;
size_t bytes = frame_count * frame_size;
audio_vbuffer->data = calloc(bytes, 1);
if (!audio_vbuffer->data) {
return -ENOMEM;
}
audio_vbuffer->head = 0;
audio_vbuffer->tail = 0;
audio_vbuffer->live = 0;
pthread_mutex_init (&audio_vbuffer->lock, (const pthread_mutexattr_t *) NULL);
return 0;
}
static int audio_vbuffer_destroy (audio_vbuffer_t * audio_vbuffer) {
if (!audio_vbuffer) {
return -EINVAL;
}
free(audio_vbuffer->data);
pthread_mutex_destroy(&audio_vbuffer->lock);
return 0;
}
static int audio_vbuffer_live (audio_vbuffer_t * audio_vbuffer) {
if (!audio_vbuffer) {
return -EINVAL;
}
pthread_mutex_lock (&audio_vbuffer->lock);
int live = audio_vbuffer->live;
pthread_mutex_unlock (&audio_vbuffer->lock);
return live;
}
#define MIN(a,b) (((a)<(b))?(a):(b))
static size_t audio_vbuffer_write (audio_vbuffer_t * audio_vbuffer, const void * buffer, size_t frame_count) {
size_t frames_written = 0;
pthread_mutex_lock (&audio_vbuffer->lock);
while (frame_count != 0) {
int frames = 0;
if (audio_vbuffer->live == 0 || audio_vbuffer->head > audio_vbuffer->tail) {
frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->head);
} else if (audio_vbuffer->head < audio_vbuffer->tail) {
frames = MIN(frame_count, audio_vbuffer->tail - (audio_vbuffer->head));
} else {
// Full
break;
}
memcpy(&audio_vbuffer->data[audio_vbuffer->head*audio_vbuffer->frame_size],
&((uint8_t*)buffer)[frames_written*audio_vbuffer->frame_size],
frames*audio_vbuffer->frame_size);
audio_vbuffer->live += frames;
frames_written += frames;
frame_count -= frames;
audio_vbuffer->head = (audio_vbuffer->head + frames) % audio_vbuffer->frame_count;
}
pthread_mutex_unlock (&audio_vbuffer->lock);
return frames_written;
}
static size_t audio_vbuffer_read (audio_vbuffer_t * audio_vbuffer, void * buffer, size_t frame_count) {
size_t frames_read = 0;
pthread_mutex_lock (&audio_vbuffer->lock);
while (frame_count != 0) {
int frames = 0;
if (audio_vbuffer->live == audio_vbuffer->frame_count ||
audio_vbuffer->tail > audio_vbuffer->head) {
frames = MIN(frame_count, audio_vbuffer->frame_count - audio_vbuffer->tail);
} else if (audio_vbuffer->tail < audio_vbuffer->head) {
frames = MIN(frame_count, audio_vbuffer->head - audio_vbuffer->tail);
} else {
break;
}
memcpy(&((uint8_t*)buffer)[frames_read*audio_vbuffer->frame_size],
&audio_vbuffer->data[audio_vbuffer->tail*audio_vbuffer->frame_size],
frames*audio_vbuffer->frame_size);
audio_vbuffer->live -= frames;
frames_read += frames;
frame_count -= frames;
audio_vbuffer->tail = (audio_vbuffer->tail + frames) % audio_vbuffer->frame_count;
}
pthread_mutex_unlock (&audio_vbuffer->lock);
return frames_read;
}
struct generic_stream_out {
struct audio_stream_out stream; // Constant after init
pthread_mutex_t lock;
struct generic_audio_device *dev; // Constant after init
audio_devices_t device; // Protected by this->lock
struct audio_config req_config; // Constant after init
struct pcm_config pcm_config; // Constant after init
audio_vbuffer_t buffer; // Constant after init
// Time & Position Keeping
bool standby; // Protected by this->lock
uint64_t underrun_position; // Protected by this->lock
struct timespec underrun_time; // Protected by this->lock
uint64_t last_write_time_us; // Protected by this->lock
uint64_t frames_total_buffered; // Protected by this->lock
uint64_t frames_written; // Protected by this->lock
uint64_t frames_rendered; // Protected by this->lock
// Worker
pthread_t worker_thread; // Constant after init
pthread_cond_t worker_wake; // Protected by this->lock
bool worker_standby; // Protected by this->lock
bool worker_exit; // Protected by this->lock
};
struct generic_stream_in {
struct audio_stream_in stream; // Constant after init
pthread_mutex_t lock;
struct generic_audio_device *dev; // Constant after init
audio_devices_t device; // Protected by this->lock
struct audio_config req_config; // Constant after init
struct pcm *pcm; // Protected by this->lock
struct pcm_config pcm_config; // Constant after init
int16_t *stereo_to_mono_buf; // Protected by this->lock
size_t stereo_to_mono_buf_size; // Protected by this->lock
audio_vbuffer_t buffer; // Protected by this->lock
// Time & Position Keeping
bool standby; // Protected by this->lock
int64_t standby_position; // Protected by this->lock
struct timespec standby_exit_time;// Protected by this->lock
int64_t standby_frames_read; // Protected by this->lock
// Worker
pthread_t worker_thread; // Constant after init
pthread_cond_t worker_wake; // Protected by this->lock
bool worker_standby; // Protected by this->lock
bool worker_exit; // Protected by this->lock
};
static struct pcm_config pcm_config_out = {
.channels = 2,
.rate = 0,
.period_size = 0,
.period_count = OUT_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = 0,
};
static struct pcm_config pcm_config_in = {
.channels = 2,
.rate = 0,
.period_size = 0,
.period_count = IN_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = 0,
.stop_threshold = INT_MAX,
};
static pthread_mutex_t adev_init_lock = PTHREAD_MUTEX_INITIALIZER;
static unsigned int audio_device_ref_count = 0;
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
return out->req_config.sample_rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return -ENOSYS;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
int size = out->pcm_config.period_size *
audio_stream_out_frame_size(&out->stream);
return size;
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
return out->req_config.channel_mask;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
return out->req_config.format;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
return -ENOSYS;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
pthread_mutex_lock(&out->lock);
dprintf(fd, "\tout_dump:\n"
"\t\tsample rate: %u\n"
"\t\tbuffer size: %zu\n"
"\t\tchannel mask: %08x\n"
"\t\tformat: %d\n"
"\t\tdevice: %08x\n"
"\t\taudio dev: %p\n\n",
out_get_sample_rate(stream),
out_get_buffer_size(stream),
out_get_channels(stream),
out_get_format(stream),
out->device,
out->dev);
pthread_mutex_unlock(&out->lock);
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
struct str_parms *parms;
char value[32];
int ret = -ENOSYS;
int success;
long val;
char *end;
if (kvpairs == NULL || kvpairs[0] == 0) {
return 0;
}
pthread_mutex_lock(&out->lock);
if (out->standby) {
parms = str_parms_create_str(kvpairs);
success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
value, sizeof(value));
if (success >= 0) {
errno = 0;
val = strtol(value, &end, 10);
if (errno == 0 && (end != NULL) && (*end == '\0') && ((int)val == val)) {
out->device = (int)val;
ret = 0;
}
}
// NO op for AUDIO_PARAMETER_DEVICE_CONNECT and AUDIO_PARAMETER_DEVICE_DISCONNECT
success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT,
value, sizeof(value));
if (success >= 0) {
ret = 0;
}
success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT,
value, sizeof(value));
if (success >= 0) {
ret = 0;
}
if (ret != 0) {
ALOGD("%s Unsupported parameter %s", __FUNCTION__, kvpairs);
}
str_parms_destroy(parms);
}
pthread_mutex_unlock(&out->lock);
return ret;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
struct str_parms *query = str_parms_create_str(keys);
char *str = NULL;
char value[256];
struct str_parms *reply = str_parms_create();
int ret;
bool get = false;
ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (ret >= 0) {
pthread_mutex_lock(&out->lock);
str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, out->device);
pthread_mutex_unlock(&out->lock);
get = true;
}
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
value[0] = 0;
strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
get = true;
}
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
value[0] = 0;
strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
get = true;
}
if (get) {
str = strdup(str_parms_to_str(reply));
}
else {
ALOGD("%s Unsupported paramter: %s", __FUNCTION__, keys);
}
str_parms_destroy(query);
str_parms_destroy(reply);
return str;
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
return (out->pcm_config.period_size * 1000) / out->pcm_config.rate;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
return -ENOSYS;
}
static void *out_write_worker(void * args)
{
struct generic_stream_out *out = (struct generic_stream_out *)args;
struct pcm *pcm = NULL;
uint8_t *buffer = NULL;
int buffer_frames;
int buffer_size;
bool restart = false;
bool shutdown = false;
while (true) {
pthread_mutex_lock(&out->lock);
while (out->worker_standby || restart) {
restart = false;
if (pcm) {
pcm_close(pcm); // Frees pcm
pcm = NULL;
free(buffer);
buffer=NULL;
}
if (out->worker_exit) {
break;
}
pthread_cond_wait(&out->worker_wake, &out->lock);
}
if (out->worker_exit) {
if (!out->worker_standby) {
ALOGE("Out worker not in standby before exiting");
}
shutdown = true;
}
while (!shutdown && audio_vbuffer_live(&out->buffer) == 0) {
pthread_cond_wait(&out->worker_wake, &out->lock);
}
if (shutdown) {
pthread_mutex_unlock(&out->lock);
break;
}
if (!pcm) {
pcm = pcm_open(PCM_CARD, PCM_DEVICE,
PCM_OUT | PCM_MONOTONIC, &out->pcm_config);
if (!pcm_is_ready(pcm)) {
ALOGE("pcm_open(out) failed: %s: channels %d format %d rate %d",
pcm_get_error(pcm),
out->pcm_config.channels,
out->pcm_config.format,
out->pcm_config.rate
);
pthread_mutex_unlock(&out->lock);
break;
}
buffer_frames = out->pcm_config.period_size;
buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
buffer = malloc(buffer_size);
if (!buffer) {
ALOGE("could not allocate write buffer");
pthread_mutex_unlock(&out->lock);
break;
}
}
int frames = audio_vbuffer_read(&out->buffer, buffer, buffer_frames);
pthread_mutex_unlock(&out->lock);
int ret = pcm_write(pcm, buffer, pcm_frames_to_bytes(pcm, frames));
if (ret != 0) {
ALOGE("pcm_write failed %s", pcm_get_error(pcm));
restart = true;
}
}
if (buffer) {
free(buffer);
}
return NULL;
}
// Call with in->lock held
static void get_current_output_position(struct generic_stream_out *out,
uint64_t * position,
struct timespec * timestamp) {
struct timespec curtime = { .tv_sec = 0, .tv_nsec = 0 };
clock_gettime(CLOCK_MONOTONIC, &curtime);
const int64_t now_us = (curtime.tv_sec * 1000000000LL + curtime.tv_nsec) / 1000;
if (timestamp) {
*timestamp = curtime;
}
int64_t position_since_underrun;
if (out->standby) {
position_since_underrun = 0;
} else {
const int64_t first_us = (out->underrun_time.tv_sec * 1000000000LL +
out->underrun_time.tv_nsec) / 1000;
position_since_underrun = (now_us - first_us) *
out_get_sample_rate(&out->stream.common) /
1000000;
if (position_since_underrun < 0) {
position_since_underrun = 0;
}
}
*position = out->underrun_position + position_since_underrun;
// The device will reuse the same output stream leading to periods of
// underrun.
if (*position > out->frames_written) {
ALOGW("Not supplying enough data to HAL, expected position %" PRIu64 " , only wrote "
"%" PRIu64,
*position, out->frames_written);
*position = out->frames_written;
out->underrun_position = *position;
out->underrun_time = curtime;
out->frames_total_buffered = 0;
}
}
static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
size_t bytes)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
const size_t frames = bytes / audio_stream_out_frame_size(stream);
pthread_mutex_lock(&out->lock);
if (out->worker_standby) {
out->worker_standby = false;
}
uint64_t current_position;
struct timespec current_time;
get_current_output_position(out, ¤t_position, ¤t_time);
const uint64_t now_us = (current_time.tv_sec * 1000000000LL +
current_time.tv_nsec) / 1000;
if (out->standby) {
out->standby = false;
out->underrun_time = current_time;
out->frames_rendered = 0;
out->frames_total_buffered = 0;
}
size_t frames_written = audio_vbuffer_write(&out->buffer, buffer, frames);
pthread_cond_signal(&out->worker_wake);
/* Implementation just consumes bytes if we start getting backed up */
out->frames_written += frames;
out->frames_rendered += frames;
out->frames_total_buffered += frames;
// We simulate the audio device blocking when it's write buffers become
// full.
// At the beginning or after an underrun, try to fill up the vbuffer.
// This will be throttled by the PlaybackThread
int frames_sleep = out->frames_total_buffered < out->buffer.frame_count ? 0 : frames;
uint64_t sleep_time_us = frames_sleep * 1000000LL /
out_get_sample_rate(&stream->common);
// If the write calls are delayed, subtract time off of the sleep to
// compensate
uint64_t time_since_last_write_us = now_us - out->last_write_time_us;
if (time_since_last_write_us < sleep_time_us) {
sleep_time_us -= time_since_last_write_us;
} else {
sleep_time_us = 0;
}
out->last_write_time_us = now_us + sleep_time_us;
pthread_mutex_unlock(&out->lock);
if (sleep_time_us > 0) {
usleep(sleep_time_us);
}
if (frames_written < frames) {
ALOGW("Hardware backing HAL too slow, could only write %zu of %zu frames", frames_written, frames);
}
/* Always consume all bytes */
return bytes;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
if (stream == NULL || frames == NULL || timestamp == NULL) {
return -EINVAL;
}
struct generic_stream_out *out = (struct generic_stream_out *)stream;
pthread_mutex_lock(&out->lock);
get_current_output_position(out, frames, timestamp);
pthread_mutex_unlock(&out->lock);
return 0;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
if (stream == NULL || dsp_frames == NULL) {
return -EINVAL;
}
struct generic_stream_out *out = (struct generic_stream_out *)stream;
pthread_mutex_lock(&out->lock);
*dsp_frames = out->frames_rendered;
pthread_mutex_unlock(&out->lock);
return 0;
}
// Must be called with out->lock held
static void do_out_standby(struct generic_stream_out *out)
{
int frames_sleep = 0;
uint64_t sleep_time_us = 0;
if (out->standby) {
return;
}
while (true) {
get_current_output_position(out, &out->underrun_position, NULL);
frames_sleep = out->frames_written - out->underrun_position;
if (frames_sleep == 0) {
break;
}
sleep_time_us = frames_sleep * 1000000LL /
out_get_sample_rate(&out->stream.common);
pthread_mutex_unlock(&out->lock);
usleep(sleep_time_us);
pthread_mutex_lock(&out->lock);
}
out->worker_standby = true;
out->standby = true;
}
static int out_standby(struct audio_stream *stream)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
pthread_mutex_lock(&out->lock);
do_out_standby(out);
pthread_mutex_unlock(&out->lock);
return 0;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
// out_add_audio_effect is a no op
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
// out_remove_audio_effect is a no op
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
int64_t *timestamp)
{
return -ENOSYS;
}
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
return in->req_config.sample_rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return -ENOSYS;
}
static int refine_output_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
{
static const uint32_t sample_rates [] = {8000,11025,16000,22050,24000,32000,
44100,48000};
static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
bool inval = false;
if (*format != AUDIO_FORMAT_PCM_16_BIT) {
*format = AUDIO_FORMAT_PCM_16_BIT;
inval = true;
}
int channel_count = popcount(*channel_mask);
if (channel_count != 1 && channel_count != 2) {
*channel_mask = AUDIO_CHANNEL_IN_STEREO;
inval = true;
}
int i;
for (i = 0; i < sample_rates_count; i++) {
if (*sample_rate < sample_rates[i]) {
*sample_rate = sample_rates[i];
inval=true;
break;
}
else if (*sample_rate == sample_rates[i]) {
break;
}
else if (i == sample_rates_count-1) {
// Cap it to the highest rate we support
*sample_rate = sample_rates[i];
inval=true;
}
}
if (inval) {
return -EINVAL;
}
return 0;
}
static int refine_input_parameters(uint32_t *sample_rate, audio_format_t *format, audio_channel_mask_t *channel_mask)
{
static const uint32_t sample_rates [] = {8000, 11025, 16000, 22050, 44100, 48000};
static const int sample_rates_count = sizeof(sample_rates)/sizeof(uint32_t);
bool inval = false;
// Only PCM_16_bit is supported. If this is changed, stereo to mono drop
// must be fixed in in_read
if (*format != AUDIO_FORMAT_PCM_16_BIT) {
*format = AUDIO_FORMAT_PCM_16_BIT;
inval = true;
}
int channel_count = popcount(*channel_mask);
if (channel_count != 1 && channel_count != 2) {
*channel_mask = AUDIO_CHANNEL_IN_STEREO;
inval = true;
}
int i;
for (i = 0; i < sample_rates_count; i++) {
if (*sample_rate < sample_rates[i]) {
*sample_rate = sample_rates[i];
inval=true;
break;
}
else if (*sample_rate == sample_rates[i]) {
break;
}
else if (i == sample_rates_count-1) {
// Cap it to the highest rate we support
*sample_rate = sample_rates[i];
inval=true;
}
}
if (inval) {
return -EINVAL;
}
return 0;
}
static int check_input_parameters(uint32_t sample_rate, audio_format_t format,
audio_channel_mask_t channel_mask)
{
return refine_input_parameters(&sample_rate, &format, &channel_mask);
}
static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format,
audio_channel_mask_t channel_mask)
{
size_t size;
int channel_count = popcount(channel_mask);
if (check_input_parameters(sample_rate, format, channel_mask) != 0)
return 0;
size = sample_rate*IN_PERIOD_MS/1000;
// Audioflinger expects audio buffers to be multiple of 16 frames
size = ((size + 15) / 16) * 16;
size *= sizeof(short) * channel_count;
return size;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
int size = get_input_buffer_size(in->req_config.sample_rate,
in->req_config.format,
in->req_config.channel_mask);
return size;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
return in->req_config.channel_mask;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
return in->req_config.format;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
return -ENOSYS;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
pthread_mutex_lock(&in->lock);
dprintf(fd, "\tin_dump:\n"
"\t\tsample rate: %u\n"
"\t\tbuffer size: %zu\n"
"\t\tchannel mask: %08x\n"
"\t\tformat: %d\n"
"\t\tdevice: %08x\n"
"\t\taudio dev: %p\n\n",
in_get_sample_rate(stream),
in_get_buffer_size(stream),
in_get_channels(stream),
in_get_format(stream),
in->device,
in->dev);
pthread_mutex_unlock(&in->lock);
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
struct str_parms *parms;
char value[32];
int ret = -ENOSYS;
int success;
long val;
char *end;
if (kvpairs == NULL || kvpairs[0] == 0) {
return 0;
}
pthread_mutex_lock(&in->lock);
if (in->standby) {
parms = str_parms_create_str(kvpairs);
success = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
value, sizeof(value));
if (success >= 0) {
errno = 0;
val = strtol(value, &end, 10);
if ((errno == 0) && (end != NULL) && (*end == '\0') && ((int)val == val)) {
in->device = (int)val;
ret = 0;
}
}
// NO op for AUDIO_PARAMETER_DEVICE_CONNECT and AUDIO_PARAMETER_DEVICE_DISCONNECT
success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT,
value, sizeof(value));
if (success >= 0) {
ret = 0;
}
success = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT,
value, sizeof(value));
if (success >= 0) {
ret = 0;
}
if (ret != 0) {
ALOGD("%s: Unsupported parameter %s", __FUNCTION__, kvpairs);
}
str_parms_destroy(parms);
}
pthread_mutex_unlock(&in->lock);
return ret;
}
static char * in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
struct str_parms *query = str_parms_create_str(keys);
char *str = NULL;
char value[256];
struct str_parms *reply = str_parms_create();
int ret;
bool get = false;
ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (ret >= 0) {
str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, in->device);
get = true;
}
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
value[0] = 0;
strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
get = true;
}
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_FORMAT)) {
value[0] = 0;
strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_FORMAT, value);
get = true;
}
if (get) {
str = strdup(str_parms_to_str(reply));
}
else {
ALOGD("%s Unsupported paramter: %s", __FUNCTION__, keys);
}
str_parms_destroy(query);
str_parms_destroy(reply);
return str;
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
// in_set_gain is a no op
return 0;
}
// Call with in->lock held
static void get_current_input_position(struct generic_stream_in *in,
int64_t * position,
struct timespec * timestamp) {
struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
clock_gettime(CLOCK_MONOTONIC, &t);
const int64_t now_us = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
if (timestamp) {
*timestamp = t;
}
int64_t position_since_standby;
if (in->standby) {
position_since_standby = 0;
} else {
const int64_t first_us = (in->standby_exit_time.tv_sec * 1000000000LL +
in->standby_exit_time.tv_nsec) / 1000;
position_since_standby = (now_us - first_us) *
in_get_sample_rate(&in->stream.common) /
1000000;
if (position_since_standby < 0) {
position_since_standby = 0;
}
}
*position = in->standby_position + position_since_standby;
}
// Must be called with in->lock held
static void do_in_standby(struct generic_stream_in *in)
{
if (in->standby) {
return;
}
in->worker_standby = true;
get_current_input_position(in, &in->standby_position, NULL);
in->standby = true;
}
static int in_standby(struct audio_stream *stream)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
pthread_mutex_lock(&in->lock);
do_in_standby(in);
pthread_mutex_unlock(&in->lock);
return 0;
}
static void *in_read_worker(void * args)
{
struct generic_stream_in *in = (struct generic_stream_in *)args;
struct pcm *pcm = NULL;
uint8_t *buffer = NULL;
size_t buffer_frames;
int buffer_size;
bool restart = false;
bool shutdown = false;
while (true) {
pthread_mutex_lock(&in->lock);
while (in->worker_standby || restart) {
restart = false;
if (pcm) {
pcm_close(pcm); // Frees pcm
pcm = NULL;
free(buffer);
buffer=NULL;
}
if (in->worker_exit) {
break;
}
pthread_cond_wait(&in->worker_wake, &in->lock);
}
if (in->worker_exit) {
if (!in->worker_standby) {
ALOGE("In worker not in standby before exiting");
}
shutdown = true;
}
if (shutdown) {
pthread_mutex_unlock(&in->lock);
break;
}
if (!pcm) {
pcm = pcm_open(PCM_CARD, PCM_DEVICE,
PCM_IN | PCM_MONOTONIC, &in->pcm_config);
if (!pcm_is_ready(pcm)) {
ALOGE("pcm_open(in) failed: %s: channels %d format %d rate %d",
pcm_get_error(pcm),
in->pcm_config.channels,
in->pcm_config.format,
in->pcm_config.rate
);
pthread_mutex_unlock(&in->lock);
break;
}
buffer_frames = in->pcm_config.period_size;
buffer_size = pcm_frames_to_bytes(pcm, buffer_frames);
buffer = malloc(buffer_size);
if (!buffer) {
ALOGE("could not allocate worker read buffer");
pthread_mutex_unlock(&in->lock);
break;
}
}
pthread_mutex_unlock(&in->lock);
int ret = pcm_read(pcm, buffer, pcm_frames_to_bytes(pcm, buffer_frames));
if (ret != 0) {
ALOGW("pcm_read failed %s", pcm_get_error(pcm));
restart = true;
continue;
}
pthread_mutex_lock(&in->lock);
size_t frames_written = audio_vbuffer_write(&in->buffer, buffer, buffer_frames);
pthread_mutex_unlock(&in->lock);
if (frames_written != buffer_frames) {
ALOGW("in_read_worker only could write %zu / %zu frames", frames_written, buffer_frames);
}
}
if (buffer) {
free(buffer);
}
return NULL;
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
struct generic_audio_device *adev = in->dev;
const size_t frames = bytes / audio_stream_in_frame_size(stream);
bool mic_mute = false;
size_t read_bytes = 0;
adev_get_mic_mute(&adev->device, &mic_mute);
pthread_mutex_lock(&in->lock);
if (in->worker_standby) {
in->worker_standby = false;
}
pthread_cond_signal(&in->worker_wake);
int64_t current_position;
struct timespec current_time;
get_current_input_position(in, ¤t_position, ¤t_time);
if (in->standby) {
in->standby = false;
in->standby_exit_time = current_time;
in->standby_frames_read = 0;
}
const int64_t frames_available = current_position - in->standby_position - in->standby_frames_read;
assert(frames_available >= 0);
const size_t frames_wait = ((uint64_t)frames_available > frames) ? 0 : frames - frames_available;
int64_t sleep_time_us = frames_wait * 1000000LL /
in_get_sample_rate(&stream->common);
pthread_mutex_unlock(&in->lock);
if (sleep_time_us > 0) {
usleep(sleep_time_us);
}
pthread_mutex_lock(&in->lock);
int read_frames = 0;
if (in->standby) {
ALOGW("Input put to sleep while read in progress");
goto exit;
}
in->standby_frames_read += frames;
if (popcount(in->req_config.channel_mask) == 1 &&
in->pcm_config.channels == 2) {
// Need to resample to mono
if (in->stereo_to_mono_buf_size < bytes*2) {
in->stereo_to_mono_buf = realloc(in->stereo_to_mono_buf,
bytes*2);
if (!in->stereo_to_mono_buf) {
ALOGE("Failed to allocate stereo_to_mono_buff");
goto exit;
}
}
read_frames = audio_vbuffer_read(&in->buffer, in->stereo_to_mono_buf, frames);
// Currently only pcm 16 is supported.
uint16_t *src = (uint16_t *)in->stereo_to_mono_buf;
uint16_t *dst = (uint16_t *)buffer;
size_t i;
// Resample stereo 16 to mono 16 by dropping one channel.
// The stereo stream is interleaved L-R-L-R
for (i = 0; i < frames; i++) {
*dst = *src;
src += 2;
dst += 1;
}
} else {
read_frames = audio_vbuffer_read(&in->buffer, buffer, frames);
}
exit:
read_bytes = read_frames*audio_stream_in_frame_size(stream);
if (mic_mute) {
read_bytes = 0;
}
if (read_bytes < bytes) {
memset (&((uint8_t *)buffer)[read_bytes], 0, bytes-read_bytes);
}
pthread_mutex_unlock(&in->lock);
return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
return 0;
}
static int in_get_capture_position(const struct audio_stream_in *stream,
int64_t *frames, int64_t *time)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
pthread_mutex_lock(&in->lock);
struct timespec current_time;
get_current_input_position(in, frames, ¤t_time);
*time = (current_time.tv_sec * 1000000000LL + current_time.tv_nsec);
pthread_mutex_unlock(&in->lock);
return 0;
}
static int in_get_active_microphones(const struct audio_stream_in *stream,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count)
{
return adev_get_microphones(NULL, mic_array, mic_count);
}
static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
// in_add_audio_effect is a no op
return 0;
}
static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
// in_add_audio_effect is a no op
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,
const char *address __unused)
{
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
struct generic_stream_out *out;
int ret = 0;
if (refine_output_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
ALOGE("Error opening output stream format %d, channel_mask %04x, sample_rate %u",
config->format, config->channel_mask, config->sample_rate);
ret = -EINVAL;
goto error;
}
out = (struct generic_stream_out *)calloc(1, sizeof(struct generic_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;
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
out->dev = adev;
out->device = devices;
memcpy(&out->req_config, config, sizeof(struct audio_config));
memcpy(&out->pcm_config, &pcm_config_out, sizeof(struct pcm_config));
out->pcm_config.rate = config->sample_rate;
out->pcm_config.period_size = out->pcm_config.rate*OUT_PERIOD_MS/1000;
out->standby = true;
out->underrun_position = 0;
out->underrun_time.tv_sec = 0;
out->underrun_time.tv_nsec = 0;
out->last_write_time_us = 0;
out->frames_total_buffered = 0;
out->frames_written = 0;
out->frames_rendered = 0;
ret = audio_vbuffer_init(&out->buffer,
out->pcm_config.period_size*out->pcm_config.period_count,
out->pcm_config.channels *
pcm_format_to_bits(out->pcm_config.format) >> 3);
if (ret == 0) {
pthread_cond_init(&out->worker_wake, NULL);
out->worker_standby = true;
out->worker_exit = false;
pthread_create(&out->worker_thread, NULL, out_write_worker, out);
}
*stream_out = &out->stream;
error:
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
struct generic_stream_out *out = (struct generic_stream_out *)stream;
pthread_mutex_lock(&out->lock);
do_out_standby(out);
out->worker_exit = true;
pthread_cond_signal(&out->worker_wake);
pthread_mutex_unlock(&out->lock);
pthread_join(out->worker_thread, NULL);
pthread_mutex_destroy(&out->lock);
audio_vbuffer_destroy(&out->buffer);
free(stream);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
return 0;
}
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)
{
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
// adev_set_voice_volume is a no op (simulates phones)
return 0;
}
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)
{
// adev_set_mode is a no op (simulates phones)
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
pthread_mutex_lock(&adev->lock);
adev->mic_mute = state;
pthread_mutex_unlock(&adev->lock);
return 0;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
pthread_mutex_lock(&adev->lock);
*state = adev->mic_mute;
pthread_mutex_unlock(&adev->lock);
return 0;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
return get_input_buffer_size(config->sample_rate, config->format, config->channel_mask);
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream)
{
struct generic_stream_in *in = (struct generic_stream_in *)stream;
pthread_mutex_lock(&in->lock);
do_in_standby(in);
in->worker_exit = true;
pthread_cond_signal(&in->worker_wake);
pthread_mutex_unlock(&in->lock);
pthread_join(in->worker_thread, NULL);
if (in->stereo_to_mono_buf != NULL) {
free(in->stereo_to_mono_buf);
in->stereo_to_mono_buf_size = 0;
}
pthread_mutex_destroy(&in->lock);
audio_vbuffer_destroy(&in->buffer);
free(stream);
}
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,
audio_input_flags_t flags __unused,
const char *address __unused,
audio_source_t source __unused)
{
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
struct generic_stream_in *in;
int ret = 0;
if (refine_input_parameters(&config->sample_rate, &config->format, &config->channel_mask)) {
ALOGE("Error opening input stream format %d, channel_mask %04x, sample_rate %u",
config->format, config->channel_mask, config->sample_rate);
ret = -EINVAL;
goto error;
}
in = (struct generic_stream_in *)calloc(1, sizeof(struct generic_stream_in));
if (!in) {
ret = -ENOMEM;
goto error;
}
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate; // no op
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; // no op
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; // no op
in->stream.common.remove_audio_effect = in_remove_audio_effect; // no op
in->stream.set_gain = in_set_gain; // no op
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost; // no op
in->stream.get_capture_position = in_get_capture_position;
in->stream.get_active_microphones = in_get_active_microphones;
pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
in->dev = adev;
in->device = devices;
memcpy(&in->req_config, config, sizeof(struct audio_config));
memcpy(&in->pcm_config, &pcm_config_in, sizeof(struct pcm_config));
in->pcm_config.rate = config->sample_rate;
in->pcm_config.period_size = in->pcm_config.rate*IN_PERIOD_MS/1000;
in->stereo_to_mono_buf = NULL;
in->stereo_to_mono_buf_size = 0;
in->standby = true;
in->standby_position = 0;
in->standby_exit_time.tv_sec = 0;
in->standby_exit_time.tv_nsec = 0;
in->standby_frames_read = 0;
ret = audio_vbuffer_init(&in->buffer,
in->pcm_config.period_size*in->pcm_config.period_count,
in->pcm_config.channels *
pcm_format_to_bits(in->pcm_config.format) >> 3);
if (ret == 0) {
pthread_cond_init(&in->worker_wake, NULL);
in->worker_standby = true;
in->worker_exit = false;
pthread_create(&in->worker_thread, NULL, in_read_worker, in);
}
*stream_in = &in->stream;
error:
return ret;
}
static int adev_dump(const audio_hw_device_t *dev, int fd)
{
return 0;
}
static int adev_get_microphones(const audio_hw_device_t *dev,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count)
{
if (mic_count == NULL) {
return -ENOSYS;
}
if (*mic_count == 0) {
*mic_count = 1;
return 0;
}
if (mic_array == NULL) {
return -ENOSYS;
}
strncpy(mic_array->device_id, "mic_goldfish", AUDIO_MICROPHONE_ID_MAX_LEN - 1);
mic_array->device = AUDIO_DEVICE_IN_BUILTIN_MIC;
strncpy(mic_array->address, AUDIO_BOTTOM_MICROPHONE_ADDRESS,
AUDIO_DEVICE_MAX_ADDRESS_LEN - 1);
memset(mic_array->channel_mapping, AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED,
sizeof(mic_array->channel_mapping));
mic_array->location = AUDIO_MICROPHONE_LOCATION_UNKNOWN;
mic_array->group = 0;
mic_array->index_in_the_group = 0;
mic_array->sensitivity = AUDIO_MICROPHONE_SENSITIVITY_UNKNOWN;
mic_array->max_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
mic_array->min_spl = AUDIO_MICROPHONE_SPL_UNKNOWN;
mic_array->directionality = AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN;
mic_array->num_frequency_responses = 0;
mic_array->geometric_location.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
mic_array->geometric_location.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
mic_array->geometric_location.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
mic_array->orientation.x = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
mic_array->orientation.y = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
mic_array->orientation.z = AUDIO_MICROPHONE_COORDINATE_UNKNOWN;
*mic_count = 1;
return 0;
}
static int adev_close(hw_device_t *dev)
{
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
int ret = 0;
if (!adev)
return 0;
pthread_mutex_lock(&adev_init_lock);
if (audio_device_ref_count == 0) {
ALOGE("adev_close called when ref_count 0");
ret = -EINVAL;
goto error;
}
if ((--audio_device_ref_count) == 0) {
if (adev->mixer) {
mixer_close(adev->mixer);
}
free(adev);
}
error:
pthread_mutex_unlock(&adev_init_lock);
return ret;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
static struct generic_audio_device *adev;
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
pthread_once(&sFallbackOnce, fallback_init);
if (sFallback != NULL) {
return sFallback->common.methods->open(&sFallback->common, name, device);
}
pthread_mutex_lock(&adev_init_lock);
if (audio_device_ref_count != 0) {
*device = &adev->device.common;
audio_device_ref_count++;
ALOGV("%s: returning existing instance of adev", __func__);
ALOGV("%s: exit", __func__);
goto unlock;
}
adev = calloc(1, sizeof(struct generic_audio_device));
pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
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; // no op
adev->device.set_voice_volume = adev_set_voice_volume; // no op
adev->device.set_master_volume = adev_set_master_volume; // no op
adev->device.get_master_volume = adev_get_master_volume; // no op
adev->device.set_master_mute = adev_set_master_mute; // no op
adev->device.get_master_mute = adev_get_master_mute; // no op
adev->device.set_mode = adev_set_mode; // no op
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; // no op
adev->device.get_parameters = adev_get_parameters; // no op
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->device.get_microphones = adev_get_microphones;
*device = &adev->device.common;
adev->mixer = mixer_open(PCM_CARD);
struct mixer_ctl *ctl;
// Set default mixer ctls
// Enable channels and set volume
for (int i = 0; i < (int)mixer_get_num_ctls(adev->mixer); i++) {
ctl = mixer_get_ctl(adev->mixer, i);
ALOGD("mixer %d name %s", i, mixer_ctl_get_name(ctl));
if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Volume") ||
!strcmp(mixer_ctl_get_name(ctl), "Capture Volume")) {
for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
ALOGD("set ctl %d to %d", z, 100);
mixer_ctl_set_percent(ctl, z, 100);
}
continue;
}
if (!strcmp(mixer_ctl_get_name(ctl), "Master Playback Switch") ||
!strcmp(mixer_ctl_get_name(ctl), "Capture Switch")) {
for (int z = 0; z < (int)mixer_ctl_get_num_values(ctl); z++) {
ALOGD("set ctl %d to %d", z, 1);
mixer_ctl_set_value(ctl, z, 1);
}
continue;
}
}
audio_device_ref_count++;
unlock:
pthread_mutex_unlock(&adev_init_lock);
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 = "Generic audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};
/* This function detects whether or not we should be using an alsa audio device
* or fall back to the legacy goldfish_audio driver.
*/
static void
fallback_init(void)
{
void* module;
FILE *fptr = fopen ("/proc/asound/pcm", "r");
if (fptr != NULL) {
// asound/pcm is empty if there are no devices
int c = fgetc(fptr);
fclose(fptr);
if (c != EOF) {
ALOGD("Emulator host-side ALSA audio emulation detected.");
return;
}
}
ALOGD("Emulator without host-side ALSA audio emulation detected.");
#if __LP64__
module = dlopen("/vendor/lib64/hw/audio.primary.goldfish_legacy.so",
RTLD_LAZY|RTLD_LOCAL);
#else
module = dlopen("/vendor/lib/hw/audio.primary.goldfish_legacy.so",
RTLD_LAZY|RTLD_LOCAL);
#endif
if (module != NULL) {
sFallback = (struct audio_module *)(dlsym(module, HAL_MODULE_INFO_SYM_AS_STR));
if (sFallback == NULL) {
dlclose(module);
}
}
if (sFallback == NULL) {
ALOGE("Could not find legacy fallback module!?");
}
}