X-Git-Url: https://gerrit.akraino.org/r/gitweb?a=blobdiff_plain;f=src%2Ftype3_AndroidCloud%2Fanbox-master%2Fandroid%2Faudio%2Faudio_hw.cpp;fp=src%2Ftype3_AndroidCloud%2Fanbox-master%2Fandroid%2Faudio%2Faudio_hw.cpp;h=f44e738268b1689bbfbc97ab919e04436352adce;hb=e26c1ec581be598521517829adba8c8dd23a768f;hp=0000000000000000000000000000000000000000;hpb=6699c1aea74eeb0eb400e6299079f0c7576f716f;p=iec.git diff --git a/src/type3_AndroidCloud/anbox-master/android/audio/audio_hw.cpp b/src/type3_AndroidCloud/anbox-master/android/audio/audio_hw.cpp new file mode 100644 index 0000000..f44e738 --- /dev/null +++ b/src/type3_AndroidCloud/anbox-master/android/audio/audio_hw.cpp @@ -0,0 +1,676 @@ +/* + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include +#include + +#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, + }, +};