blob: 3fdf3f97780c0f7f430fe79a079daa1b99f8b817 [file] [log] [blame]
/*
* Copyright (C) 2016 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_hikey"
//#define LOG_NDEBUG 0
#include <errno.h>
#include <malloc.h>
#include <pthread.h>
#include <stdint.h>
#include <sys/time.h>
#include <stdlib.h>
#include <unistd.h>
#include <log/log.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include <sound/asound.h>
#include <tinyalsa/asoundlib.h>
#include <audio_utils/resampler.h>
#include <audio_utils/echo_reference.h>
#include <hardware/audio_effect.h>
#include <hardware/audio_alsaops.h>
#include <audio_effects/effect_aec.h>
#include <sys/ioctl.h>
#define CARD_OUT 0
#define PORT_CODEC 0
/* Minimum granularity - Arbitrary but small value */
#define CODEC_BASE_FRAME_COUNT 32
/* number of base blocks in a short period (low latency) */
#define PERIOD_MULTIPLIER 32 /* 21 ms */
/* number of frames per short period (low latency) */
#define PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * PERIOD_MULTIPLIER)
/* number of pseudo periods for low latency playback */
#define PLAYBACK_PERIOD_COUNT 4
#define PLAYBACK_PERIOD_START_THRESHOLD 2
#define CODEC_SAMPLING_RATE 48000
#define CHANNEL_STEREO 2
#define MIN_WRITE_SLEEP_US 5000
#ifdef ENABLE_XAF_DSP_DEVICE
#include "xaf-utils-test.h"
#include "audio/xa_vorbis_dec_api.h"
#include "audio/xa-audio-decoder-api.h"
#define NUM_COMP_IN_GRAPH 1
struct alsa_audio_device;
struct xaf_dsp_device {
void *p_adev;
void *p_decoder;
xaf_info_t comp_info;
/* ...playback format */
xaf_format_t pb_format;
xaf_comp_status dec_status;
int dec_info[4];
void *dec_inbuf[2];
int read_length;
xf_id_t dec_id;
int xaf_started;
mem_obj_t* mem_handle;
int num_comp;
int (*dec_setup)(void *p_comp, struct alsa_audio_device *audio_device);
int xafinitdone;
};
#endif
struct stub_stream_in {
struct audio_stream_in stream;
};
struct alsa_audio_device {
struct audio_hw_device hw_device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
int devices;
struct alsa_stream_in *active_input;
struct alsa_stream_out *active_output;
bool mic_mute;
#ifdef ENABLE_XAF_DSP_DEVICE
struct xaf_dsp_device dsp_device;
int hifi_dsp_fd;
#endif
};
struct alsa_stream_out {
struct audio_stream_out stream;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
struct pcm_config config;
struct pcm *pcm;
bool unavailable;
int standby;
struct alsa_audio_device *dev;
int write_threshold;
unsigned int written;
};
#ifdef ENABLE_XAF_DSP_DEVICE
static int pcm_setup(void *p_pcm, struct alsa_audio_device *audio_device)
{
int param[6];
param[0] = XA_CODEC_CONFIG_PARAM_SAMPLE_RATE;
param[1] = audio_device->dsp_device.pb_format.sample_rate;
param[2] = XA_CODEC_CONFIG_PARAM_CHANNELS;
param[3] = audio_device->dsp_device.pb_format.channels;
param[4] = XA_CODEC_CONFIG_PARAM_PCM_WIDTH;
param[5] = audio_device->dsp_device.pb_format.pcm_width;
XF_CHK_API(xaf_comp_set_config(p_pcm, 3, &param[0]));
return 0;
}
void xa_thread_exit_handler(int sig)
{
/* ...unused arg */
(void) sig;
pthread_exit(0);
}
/*xtensa audio device init*/
static int xa_device_init(struct alsa_audio_device *audio_device)
{
/* ...initialize playback format */
audio_device->dsp_device.p_adev = NULL;
audio_device->dsp_device.pb_format.sample_rate = 48000;
audio_device->dsp_device.pb_format.channels = 2;
audio_device->dsp_device.pb_format.pcm_width = 16;
audio_device->dsp_device.xafinitdone = 0;
audio_frmwk_buf_size = 0; //unused
audio_comp_buf_size = 0; //unused
audio_device->dsp_device.num_comp = NUM_COMP_IN_GRAPH;
struct sigaction actions;
memset(&actions, 0, sizeof(actions));
sigemptyset(&actions.sa_mask);
actions.sa_flags = 0;
actions.sa_handler = xa_thread_exit_handler;
sigaction(SIGUSR1,&actions,NULL);
/* ...initialize tracing facility */
audio_device->dsp_device.xaf_started =1;
audio_device->dsp_device.dec_id = "audio-decoder/pcm";
audio_device->dsp_device.dec_setup = pcm_setup;
audio_device->dsp_device.mem_handle = mem_init(); //initialize memory handler
XF_CHK_API(xaf_adev_open(&audio_device->dsp_device.p_adev, audio_frmwk_buf_size, audio_comp_buf_size, mem_malloc, mem_free));
/* ...create decoder component */
XF_CHK_API(xaf_comp_create(audio_device->dsp_device.p_adev, &audio_device->dsp_device.p_decoder, audio_device->dsp_device.dec_id, 1, 1, &audio_device->dsp_device.dec_inbuf[0], XAF_DECODER));
XF_CHK_API(audio_device->dsp_device.dec_setup(audio_device->dsp_device.p_decoder,audio_device));
/* ...start decoder component */
XF_CHK_API(xaf_comp_process(audio_device->dsp_device.p_adev, audio_device->dsp_device.p_decoder, NULL, 0, XAF_START_FLAG));
return 0;
}
static int xa_device_run(struct audio_stream_out *stream, const void *buffer, size_t frame_size, size_t out_frames, size_t bytes)
{
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
struct alsa_audio_device *adev = out->dev;
int ret=0;
void *p_comp=adev->dsp_device.p_decoder;
xaf_comp_status comp_status;
memcpy(adev->dsp_device.dec_inbuf[0],buffer,bytes);
adev->dsp_device.read_length=bytes;
if (adev->dsp_device.xafinitdone == 0) {
XF_CHK_API(xaf_comp_process(adev->dsp_device.p_adev, adev->dsp_device.p_decoder, adev->dsp_device.dec_inbuf[0], adev->dsp_device.read_length, XAF_INPUT_READY_FLAG));
XF_CHK_API(xaf_comp_get_status(adev->dsp_device.p_adev, adev->dsp_device.p_decoder, &adev->dsp_device.dec_status, &adev->dsp_device.comp_info));
ALOGE("PROXY:%s xaf_comp_get_status %d\n",__func__,adev->dsp_device.dec_status);
if (adev->dsp_device.dec_status == XAF_INIT_DONE) {
adev->dsp_device.xafinitdone = 1;
out->written += out_frames;
XF_CHK_API(xaf_comp_process(NULL, p_comp, NULL, 0, XAF_EXEC_FLAG));
}
} else {
XF_CHK_API(xaf_comp_process(NULL, adev->dsp_device.p_decoder, adev->dsp_device.dec_inbuf[0], adev->dsp_device.read_length, XAF_INPUT_READY_FLAG));
while (1) {
XF_CHK_API(xaf_comp_get_status(NULL, p_comp, &comp_status, &adev->dsp_device.comp_info));
if (comp_status == XAF_EXEC_DONE) break;
if (comp_status == XAF_NEED_INPUT) {
ALOGV("PROXY:%s loop:XAF_NEED_INPUT\n",__func__);
break;
}
if (comp_status == XAF_OUTPUT_READY) {
void *p_buf = (void *)adev->dsp_device.comp_info.buf;
int size = adev->dsp_device.comp_info.length;
ret = pcm_mmap_write(out->pcm, p_buf, size);
if (ret == 0) {
out->written += out_frames;
}
XF_CHK_API(xaf_comp_process(NULL, adev->dsp_device.p_decoder, (void *)adev->dsp_device.comp_info.buf, adev->dsp_device.comp_info.length, XAF_NEED_OUTPUT_FLAG));
}
}
}
return ret;
}
static int xa_device_close(struct alsa_audio_device *audio_device)
{
if (audio_device->dsp_device.xaf_started) {
xaf_comp_status comp_status;
audio_device->dsp_device.xaf_started=0;
while (1) {
XF_CHK_API(xaf_comp_get_status(NULL, audio_device->dsp_device.p_decoder, &comp_status, &audio_device->dsp_device.comp_info));
ALOGV("PROXY:comp_status:%d,audio_device->dsp_device.comp_info.length:%d\n",(int)comp_status,audio_device->dsp_device.comp_info.length);
if (comp_status == XAF_EXEC_DONE)
break;
if (comp_status == XAF_NEED_INPUT) {
XF_CHK_API(xaf_comp_process(NULL, audio_device->dsp_device.p_decoder, NULL, 0, XAF_INPUT_OVER_FLAG));
}
if (comp_status == XAF_OUTPUT_READY) {
XF_CHK_API(xaf_comp_process(NULL, audio_device->dsp_device.p_decoder, (void *)audio_device->dsp_device.comp_info.buf, audio_device->dsp_device.comp_info.length, XAF_NEED_OUTPUT_FLAG));
}
}
/* ...exec done, clean-up */
XF_CHK_API(xaf_comp_delete(audio_device->dsp_device.p_decoder));
XF_CHK_API(xaf_adev_close(audio_device->dsp_device.p_adev, 0 /*unused*/));
mem_exit();
XF_CHK_API(print_mem_mcps_info(audio_device->dsp_device.mem_handle, audio_device->dsp_device.num_comp));
}
return 0;
}
#endif
/* must be called with hw device and output stream mutexes locked */
static int start_output_stream(struct alsa_stream_out *out)
{
struct alsa_audio_device *adev = out->dev;
if (out->unavailable)
return -ENODEV;
/* default to low power: will be corrected in out_write if necessary before first write to
* tinyalsa.
*/
out->write_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE;
out->config.start_threshold = PLAYBACK_PERIOD_START_THRESHOLD * PERIOD_SIZE;
out->config.avail_min = PERIOD_SIZE;
out->pcm = pcm_open(CARD_OUT, PORT_CODEC, PCM_OUT | PCM_MMAP | PCM_NOIRQ | PCM_MONOTONIC, &out->config);
if (!pcm_is_ready(out->pcm)) {
ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm));
pcm_close(out->pcm);
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
adev->active_output = out;
return 0;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return out->config.rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("out_set_sample_rate: %d", 0);
return -ENOSYS;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
ALOGV("out_get_buffer_size: %d", 4096);
/* return the closest majoring multiple of 16 frames, as
* audioflinger expects audio buffers to be a multiple of 16 frames */
size_t size = PERIOD_SIZE;
size = ((size + 15) / 16) * 16;
return size * audio_stream_out_frame_size((struct audio_stream_out *)stream);
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
ALOGV("out_get_channels");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return audio_channel_out_mask_from_count(out->config.channels);
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
ALOGV("out_get_format");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return audio_format_from_pcm_format(out->config.format);
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
ALOGV("out_set_format: %d",format);
return -ENOSYS;
}
static int do_output_standby(struct alsa_stream_out *out)
{
struct alsa_audio_device *adev = out->dev;
if (!out->standby) {
pcm_close(out->pcm);
out->pcm = NULL;
adev->active_output = NULL;
out->standby = 1;
}
return 0;
}
static int out_standby(struct audio_stream *stream)
{
ALOGV("out_standby");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
int status;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
#ifdef ENABLE_XAF_DSP_DEVICE
xa_device_close(out->dev);
#endif
status = do_output_standby(out);
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
return status;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
ALOGV("out_dump");
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
ALOGV("out_set_parameters");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
struct alsa_audio_device *adev = out->dev;
struct str_parms *parms;
char value[32];
int val = 0;
int ret = -EINVAL;
if (kvpairs == NULL || kvpairs[0] == 0) {
return 0;
}
parms = str_parms_create_str(kvpairs);
if (str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)) >= 0) {
val = atoi(value);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (((adev->devices & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) {
adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
adev->devices |= val;
}
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
ret = 0;
}
str_parms_destroy(parms);
return ret;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
ALOGV("out_get_parameters");
return strdup("");
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
ALOGV("out_get_latency");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return (PERIOD_SIZE * PLAYBACK_PERIOD_COUNT * 1000) / out->config.rate;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
ALOGV("out_set_volume: Left:%f Right:%f", left, right);
return 0;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret;
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
struct alsa_audio_device *adev = out->dev;
size_t frame_size = audio_stream_out_frame_size(stream);
size_t out_frames = bytes / frame_size;
/* acquiring hw device mutex systematically is useful if a low priority thread is waiting
* on the output stream mutex - e.g. executing select_mode() while holding the hw device
* mutex
*/
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
#ifdef ENABLE_XAF_DSP_DEVICE
if (adev->hifi_dsp_fd >= 0) {
xa_device_init(adev);
}
#endif
ret = start_output_stream(out);
if (ret != 0) {
pthread_mutex_unlock(&adev->lock);
goto exit;
}
out->standby = 0;
}
pthread_mutex_unlock(&adev->lock);
#ifdef ENABLE_XAF_DSP_DEVICE
/*fallback to original audio processing*/
if (adev->dsp_device.p_adev != NULL) {
ret = xa_device_run(stream, buffer,frame_size, out_frames, bytes);
} else {
#endif
ret = pcm_mmap_write(out->pcm, buffer, out_frames * frame_size);
if (ret == 0) {
out->written += out_frames;
}
#ifdef ENABLE_XAF_DSP_DEVICE
}
#endif
exit:
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
usleep((int64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common));
}
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
*dsp_frames = 0;
ALOGV("out_get_render_position: dsp_frames: %p", dsp_frames);
return -EINVAL;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
int ret = -1;
if (out->pcm) {
unsigned int avail;
if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
int64_t signed_frames = out->written - kernel_buffer_size + avail;
if (signed_frames >= 0) {
*frames = signed_frames;
ret = 0;
}
}
}
return ret;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
ALOGV("out_add_audio_effect: %p", effect);
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
ALOGV("out_remove_audio_effect: %p", effect);
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
int64_t *timestamp)
{
*timestamp = 0;
ALOGV("out_get_next_write_timestamp: %ld", (long int)(*timestamp));
return -EINVAL;
}
/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
ALOGV("in_get_sample_rate");
return 8000;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("in_set_sample_rate: %d", rate);
return -ENOSYS;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
ALOGV("in_get_buffer_size: %d", 320);
return 320;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
ALOGV("in_get_channels: %d", AUDIO_CHANNEL_IN_MONO);
return AUDIO_CHANNEL_IN_MONO;
}
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)
{
return -ENOSYS;
}
static int in_standby(struct audio_stream *stream)
{
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 %zu", bytes);
/* XXX: fake timing for audio input */
usleep((int64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&stream->common));
memset(buffer, 0, 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,
const char *address __unused)
{
ALOGV("adev_open_output_stream...");
struct alsa_audio_device *ladev = (struct alsa_audio_device *)dev;
struct alsa_stream_out *out;
struct pcm_params *params;
int ret = 0;
params = pcm_params_get(CARD_OUT, PORT_CODEC, PCM_OUT);
if (!params)
return -ENOSYS;
out = (struct alsa_stream_out *)calloc(1, sizeof(struct alsa_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_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
out->config.channels = CHANNEL_STEREO;
out->config.rate = CODEC_SAMPLING_RATE;
out->config.format = PCM_FORMAT_S16_LE;
out->config.period_size = PERIOD_SIZE;
out->config.period_count = PLAYBACK_PERIOD_COUNT;
if (out->config.rate != config->sample_rate ||
audio_channel_count_from_out_mask(config->channel_mask) != CHANNEL_STEREO ||
out->config.format != pcm_format_from_audio_format(config->format) ) {
config->sample_rate = out->config.rate;
config->format = audio_format_from_pcm_format(out->config.format);
config->channel_mask = audio_channel_out_mask_from_count(CHANNEL_STEREO);
ret = -EINVAL;
}
ALOGI("adev_open_output_stream selects channels=%d rate=%d format=%d",
out->config.channels, out->config.rate, out->config.format);
out->dev = ladev;
out->standby = 1;
out->unavailable = false;
config->format = out_get_format(&out->stream.common);
config->channel_mask = out_get_channels(&out->stream.common);
config->sample_rate = out_get_sample_rate(&out->stream.common);
*stream_out = &out->stream;
/* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
ret = 0;
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
ALOGV("adev_close_output_stream...");
free(stream);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
ALOGV("adev_set_parameters");
return -ENOSYS;
}
static char * adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
ALOGV("adev_get_parameters");
return strdup("");
}
static int adev_init_check(const struct audio_hw_device *dev)
{
ALOGV("adev_init_check");
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
ALOGV("adev_set_voice_volume: %f", volume);
return -ENOSYS;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
ALOGV("adev_set_master_volume: %f", volume);
return -ENOSYS;
}
static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
{
ALOGV("adev_get_master_volume: %f", *volume);
return -ENOSYS;
}
static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
{
ALOGV("adev_set_master_mute: %d", muted);
return -ENOSYS;
}
static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
{
ALOGV("adev_get_master_mute: %d", *muted);
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
ALOGV("adev_set_mode: %d", mode);
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
ALOGV("adev_set_mic_mute: %d",state);
return -ENOSYS;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
ALOGV("adev_get_mic_mute");
return -ENOSYS;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
ALOGV("adev_get_input_buffer_size: %d", 320);
return 320;
}
static int adev_open_input_stream(struct audio_hw_device __unused *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 stub_stream_in *in;
ALOGV("adev_open_input_stream...");
in = (struct stub_stream_in *)calloc(1, sizeof(struct stub_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;
*stream_in = &in->stream;
return 0;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *in)
{
ALOGV("adev_close_input_stream...");
return;
}
static int adev_dump(const audio_hw_device_t *device, int fd)
{
ALOGV("adev_dump");
return 0;
}
static int adev_close(hw_device_t *device)
{
#ifdef ENABLE_XAF_DSP_DEVICE
struct alsa_audio_device *adev = (struct alsa_audio_device *)device;
#endif
ALOGV("adev_close");
#ifdef ENABLE_XAF_DSP_DEVICE
if (adev->hifi_dsp_fd >= 0)
close(adev->hifi_dsp_fd);
#endif
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct alsa_audio_device *adev;
ALOGV("adev_open: %s", name);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
adev = calloc(1, sizeof(struct alsa_audio_device));
if (!adev)
return -ENOMEM;
adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->hw_device.common.module = (struct hw_module_t *) module;
adev->hw_device.common.close = adev_close;
adev->hw_device.init_check = adev_init_check;
adev->hw_device.set_voice_volume = adev_set_voice_volume;
adev->hw_device.set_master_volume = adev_set_master_volume;
adev->hw_device.get_master_volume = adev_get_master_volume;
adev->hw_device.set_master_mute = adev_set_master_mute;
adev->hw_device.get_master_mute = adev_get_master_mute;
adev->hw_device.set_mode = adev_set_mode;
adev->hw_device.set_mic_mute = adev_set_mic_mute;
adev->hw_device.get_mic_mute = adev_get_mic_mute;
adev->hw_device.set_parameters = adev_set_parameters;
adev->hw_device.get_parameters = adev_get_parameters;
adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
adev->hw_device.open_output_stream = adev_open_output_stream;
adev->hw_device.close_output_stream = adev_close_output_stream;
adev->hw_device.open_input_stream = adev_open_input_stream;
adev->hw_device.close_input_stream = adev_close_input_stream;
adev->hw_device.dump = adev_dump;
adev->devices = AUDIO_DEVICE_NONE;
*device = &adev->hw_device.common;
#ifdef ENABLE_XAF_DSP_DEVICE
adev->hifi_dsp_fd = open(HIFI_DSP_MISC_DRIVER, O_WRONLY, 0);
if (adev->hifi_dsp_fd < 0) {
ALOGW("hifi_dsp: Error opening device %d", errno);
} else {
ALOGI("hifi_dsp: Open device");
}
#endif
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 = "Hikey audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};