anbox/android/audio/audio_hw.cpp
Simon Fels d729b8b9e4 Implement audio support
Audio playback still has a delay about 1-2 seconds but it's usable.

This also moves the platform policy class into its own namespace to be
not specific to just window management.
2017-01-12 07:02:43 +01:00

676 lines
20 KiB
C++

/*
* 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"
#define LOG_NDEBUG 0
#include <errno.h>
#include <fcntl.h>
#include <pthread.h>
#include <stdint.h>
#include <stdlib.h>
#include <sys/socket.h>
#include <sys/time.h>
#include <sys/un.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <unistd.h>
#include <cutils/log.h>
#include <cutils/str_parms.h>
#include <hardware/audio.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include "anbox/audio/client_info.h"
#define AUDIO_DEVICE_NAME "/dev/anbox_audio"
#define OUT_SAMPLING_RATE 44100
#define OUT_BUFFER_SIZE 4096
#define OUT_LATENCY_MS 20
#define IN_SAMPLING_RATE 8000
#define IN_BUFFER_SIZE 320
struct generic_audio_device {
struct audio_hw_device device;
pthread_mutex_t lock;
struct audio_stream_out *output;
struct audio_stream_in *input;
bool mic_mute;
};
struct generic_stream_out {
struct audio_stream_out stream;
struct generic_audio_device *dev;
audio_devices_t device;
int fd;
};
struct generic_stream_in {
struct audio_stream_in stream;
struct generic_audio_device *dev;
audio_devices_t device;
int fd;
};
static uint32_t out_get_sample_rate(const struct audio_stream *stream) {
return OUT_SAMPLING_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) {
return OUT_BUFFER_SIZE;
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) {
return AUDIO_CHANNEL_OUT_STEREO;
}
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) {
return -ENOSYS;
}
static int out_standby(struct audio_stream *stream) {
return 0;
}
static int out_dump(const struct audio_stream *stream, int fd) {
struct generic_stream_out *out = (struct generic_stream_out *)stream;
dprintf(fd,
"\tout_dump:\n"
"\t\tsample rate: %u\n"
"\t\tbuffer size: %u\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);
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;
long val;
char *end;
parms = str_parms_create_str(kvpairs);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
value, sizeof(value));
if (ret >= 0) {
errno = 0;
val = strtol(value, &end, 10);
if (errno == 0 && (end != NULL) && (*end == '\0') && ((int)val == val)) {
out->device = (int)val;
} else {
ret = -EINVAL;
}
}
str_parms_destroy(parms);
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;
char value[256];
struct str_parms *reply = str_parms_create();
int ret;
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, out->device);
str = strdup(str_parms_to_str(reply));
} else {
str = strdup(keys);
}
str_parms_destroy(query);
str_parms_destroy(reply);
return str;
}
static uint32_t out_get_latency(const struct audio_stream_out *stream) {
return OUT_LATENCY_MS;
}
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) {
struct generic_stream_out *out = (struct generic_stream_out *)stream;
struct generic_audio_device *adev = out->dev;
pthread_mutex_lock(&adev->lock);
if (out->fd >= 0)
bytes = write(out->fd, buffer, bytes);
pthread_mutex_unlock(&adev->lock);
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames) {
return -ENOSYS;
}
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 -ENOSYS;
}
static uint32_t in_get_sample_rate(const struct audio_stream *stream) {
return IN_SAMPLING_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) {
return IN_BUFFER_SIZE;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) {
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) {
struct generic_stream_in *in = (struct generic_stream_in *)stream;
dprintf(fd,
"\tin_dump:\n"
"\t\tsample rate: %u\n"
"\t\tbuffer size: %u\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);
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;
long val;
char *end;
parms = str_parms_create_str(kvpairs);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING,
value, sizeof(value));
if (ret >= 0) {
errno = 0;
val = strtol(value, &end, 10);
if ((errno == 0) && (end != NULL) && (*end == '\0') && ((int)val == val)) {
in->device = (int)val;
} else {
ret = -EINVAL;
}
}
str_parms_destroy(parms);
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;
char value[256];
struct str_parms *reply = str_parms_create();
int ret;
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);
str = strdup(str_parms_to_str(reply));
} else {
str = strdup(keys);
}
str_parms_destroy(query);
str_parms_destroy(reply);
return str;
}
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) {
struct generic_stream_in *in = (struct generic_stream_in *)stream;
struct generic_audio_device *adev = in->dev;
pthread_mutex_lock(&adev->lock);
if (in->fd >= 0)
bytes = read(in->fd, buffer, bytes);
if (adev->mic_mute && (bytes > 0)) {
memset(buffer, 0, bytes);
}
pthread_mutex_unlock(&adev->lock);
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 connect_audio_server(const anbox::audio::ClientInfo::Type &type) {
int fd = socket(AF_LOCAL, SOCK_STREAM, 0);
if (fd < 0)
return -errno;
struct sockaddr_un addr;
memset(&addr, 0, sizeof(addr));
addr.sun_family = AF_UNIX;
strncpy(addr.sun_path, AUDIO_DEVICE_NAME, sizeof(addr.sun_path));
if (connect(fd, (struct sockaddr *)&addr, sizeof(addr)) < 0) {
close(fd);
return -errno;
}
// We will send out client type information to the server and the
// server will either deny the request by closing the connection
// or by sending us the approved client details back.
anbox::audio::ClientInfo client_info{type};
if (::write(fd, &client_info, sizeof(client_info)) < 0) {
close(fd);
return -EIO;
}
auto bytes_read = ::read(fd, &client_info, sizeof(client_info));
if (bytes_read < 0) {
close(fd);
return -EIO;
}
// FIXME once we have real client details we need to check if we
// got everything we need or if anything is missing.
ALOGE("Successfully connected Anbox audio server");
return fd;
}
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, fd = 0;
pthread_mutex_lock(&adev->lock);
if (adev->output != NULL) {
ret = -ENOSYS;
goto error;
}
fd = connect_audio_server(anbox::audio::ClientInfo::Type::Playback);
if (fd < 0) {
ret = fd;
ALOGE("Failed to connect with Anbox audio servers (err %d)", ret);
goto error;
}
if ((config->format != AUDIO_FORMAT_PCM_16_BIT) ||
(config->channel_mask != AUDIO_CHANNEL_OUT_STEREO) ||
(config->sample_rate != OUT_SAMPLING_RATE)) {
ALOGE("Error opening output stream format %d, channel_mask %04x, sample_rate %u",
config->format, config->channel_mask, config->sample_rate);
config->format = AUDIO_FORMAT_PCM_16_BIT;
config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
config->sample_rate = OUT_SAMPLING_RATE;
ret = -EINVAL;
goto error;
}
out = (struct generic_stream_out *)calloc(1, sizeof(struct generic_stream_out));
out->fd = fd;
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->dev = adev;
out->device = devices;
adev->output = (struct audio_stream_out *)out;
*stream_out = &out->stream;
error:
pthread_mutex_unlock(&adev->lock);
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream) {
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
pthread_mutex_lock(&adev->lock);
if (stream == adev->output) {
free(stream);
adev->output = NULL;
}
pthread_mutex_unlock(&adev->lock);
}
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) {
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) {
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 IN_BUFFER_SIZE;
}
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, fd = 0;
pthread_mutex_lock(&adev->lock);
if (adev->input != NULL) {
ret = -ENOSYS;
goto error;
}
if ((config->format != AUDIO_FORMAT_PCM_16_BIT) ||
(config->channel_mask != AUDIO_CHANNEL_IN_MONO) ||
(config->sample_rate != IN_SAMPLING_RATE)) {
ALOGE("Error opening input stream format %d, channel_mask %04x, sample_rate %u",
config->format, config->channel_mask, config->sample_rate);
config->format = AUDIO_FORMAT_PCM_16_BIT;
config->channel_mask = AUDIO_CHANNEL_IN_MONO;
config->sample_rate = IN_SAMPLING_RATE;
ret = -EINVAL;
goto error;
}
fd = connect_audio_server(anbox::audio::ClientInfo::Type::Recording);
if (fd < 0) {
ret = fd;
ALOGE("Failed to connect with Anbox audio servers (err %d)", ret);
goto error;
}
in = (struct generic_stream_in *)calloc(1, sizeof(struct generic_stream_in));
in->fd = fd;
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;
in->dev = adev;
in->device = devices;
adev->input = (struct audio_stream_in *)in;
*stream_in = &in->stream;
error:
pthread_mutex_unlock(&adev->lock);
return ret;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream) {
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
pthread_mutex_lock(&adev->lock);
if (stream == adev->input) {
free(stream);
adev->input = NULL;
}
pthread_mutex_unlock(&adev->lock);
}
static int adev_dump(const audio_hw_device_t *dev, int fd) {
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
const size_t SIZE = 256;
char buffer[SIZE];
dprintf(fd,
"\nadev_dump:\n"
"\tmic_mute: %s\n"
"\toutput: %p\n"
"\tinput: %p\n\n",
adev->mic_mute ? "true" : "false",
adev->output,
adev->input);
if (adev->output != NULL)
out_dump((const struct audio_stream *)adev->output, fd);
if (adev->input != NULL)
in_dump((const struct audio_stream *)adev->input, fd);
return 0;
}
static int adev_close(hw_device_t *dev) {
struct generic_audio_device *adev = (struct generic_audio_device *)dev;
adev_close_output_stream((struct audio_hw_device *)dev, adev->output);
adev_close_input_stream((struct audio_hw_device *)dev, adev->input);
free(dev);
return 0;
}
static int adev_open(const hw_module_t *module, const char *name,
hw_device_t **device) {
struct generic_audio_device *adev;
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
adev = (struct generic_audio_device*) calloc(1, sizeof(struct generic_audio_device));
adev->device.common.tag = HARDWARE_DEVICE_TAG;
adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->device.common.module = (struct hw_module_t *)module;
adev->device.common.close = adev_close;
adev->device.init_check = adev_init_check;
adev->device.set_voice_volume = adev_set_voice_volume;
adev->device.set_master_volume = adev_set_master_volume;
adev->device.get_master_volume = adev_get_master_volume;
adev->device.set_master_mute = adev_set_master_mute;
adev->device.get_master_mute = adev_get_master_mute;
adev->device.set_mode = adev_set_mode;
adev->device.set_mic_mute = adev_set_mic_mute;
adev->device.get_mic_mute = adev_get_mic_mute;
adev->device.set_parameters = adev_set_parameters;
adev->device.get_parameters = adev_get_parameters;
adev->device.get_input_buffer_size = adev_get_input_buffer_size;
adev->device.open_output_stream = adev_open_output_stream;
adev->device.close_output_stream = adev_close_output_stream;
adev->device.open_input_stream = adev_open_input_stream;
adev->device.close_input_stream = adev_close_input_stream;
adev->device.dump = adev_dump;
*device = &adev->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 = "Anbox audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};