/* * Copyright (C) 2017 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. */ /** * Derived from goldfish/audio/audio_hw.c * Changes made to adding support of AUDIO_DEVICE_OUT_BUS */ #define LOG_TAG "audio_hw_generic" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "audio_hw.h" #include "ext_pcm.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 #define PI 3.14159265 #define TWO_PI (2*PI) // 150 Hz #define DEFAULT_FREQUENCY 150 // Increase in changes to tone frequency #define TONE_FREQUENCY_INCREASE 20 // Max tone frequency to auto assign, don't want to generate too high of a pitch #define MAX_TONE_FREQUENCY 500 #define _bool_str(x) ((x)?"true":"false") static const char * const PROP_KEY_SIMULATE_MULTI_ZONE_AUDIO = "ro.aae.simulateMultiZoneAudio"; static const char * const AAE_PARAMETER_KEY_FOR_SELECTED_ZONE = "com.android.car.emulator.selected_zone"; #define PRIMARY_ZONE_ID 0 #define INVALID_ZONE_ID -1 // Note the primary zone goes to left speaker so route other zone to right speaker #define DEFAULT_ZONE_TO_LEFT_SPEAKER (PRIMARY_ZONE_ID + 1) static const char * const TONE_ADDRESS_KEYWORD = "_tone_"; static const char * const AUDIO_ZONE_KEYWORD = "_audio_zone_"; #define SIZE_OF_PARSE_BUFFER 32 static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state); 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 int get_int_value(const struct str_parms *str_parms, const char *key, int *return_value) { char value[SIZE_OF_PARSE_BUFFER]; int results = str_parms_get_str(str_parms, key, value, SIZE_OF_PARSE_BUFFER); if (results >= 0) { char *end = NULL; errno = 0; long val = strtol(value, &end, 10); if ((errno == 0) && (end != NULL) && (*end == '\0') && ((int) val == val)) { *return_value = val; } else { results = -EINVAL; } } return results; } 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 bool is_zone_selected_to_play(struct audio_hw_device *dev, int zone_id) { // play if current zone is enable or zone equal to primary zone bool is_selected_zone = true; if (zone_id != PRIMARY_ZONE_ID) { struct generic_audio_device *adev = (struct generic_audio_device *)dev; pthread_mutex_lock(&adev->lock); is_selected_zone = adev->last_zone_selected_to_play == zone_id; pthread_mutex_unlock(&adev->lock); } return is_selected_zone; } 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\taddress: %s\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\tamplitude ratio: %f\n" "\t\tenabled channels: %d\n" "\t\taudio dev: %p\n\n", out->bus_address, out_get_sample_rate(stream), out_get_buffer_size(stream), out_get_channels(stream), out_get_format(stream), out->device, out->amplitude_ratio, out->enabled_channels, 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; int ret = 0; pthread_mutex_lock(&out->lock); if (!out->standby) { //Do not support changing params while stream running ret = -ENOSYS; } else { parms = str_parms_create_str(kvpairs); int val = 0; ret = get_int_value(parms, AUDIO_PARAMETER_STREAM_ROUTING, &val); if (ret >= 0) { out->device = (int)val; ret = 0; } 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; 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) { pthread_mutex_lock(&out->lock); str_parms_add_int(reply, AUDIO_PARAMETER_STREAM_ROUTING, out->device); pthread_mutex_unlock(&out->lock); 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) { 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 int get_zone_id_from_address(const char *address) { int zone_id = INVALID_ZONE_ID; char *zone_start = strstr(address, AUDIO_ZONE_KEYWORD); if (zone_start) { char *end = NULL; zone_id = strtol(zone_start + strlen(AUDIO_ZONE_KEYWORD), &end, 10); if (end == NULL || zone_id < 0) { return INVALID_ZONE_ID; } } return zone_id; } static void *out_write_worker(void *args) { struct generic_stream_out *out = (struct generic_stream_out *)args; struct ext_pcm *ext_pcm = NULL; uint8_t *buffer = NULL; int buffer_frames; int buffer_size; bool restart = false; bool shutdown = false; int zone_id = PRIMARY_ZONE_ID; // If it is a audio zone keyword bus address then get zone id if (strstr(out->bus_address, AUDIO_ZONE_KEYWORD)) { zone_id = get_zone_id_from_address(out->bus_address); if (zone_id == INVALID_ZONE_ID) { ALOGE("%s Found invalid zone id, defaulting device %s to zone %d", __func__, out->bus_address, DEFAULT_ZONE_TO_LEFT_SPEAKER); zone_id = DEFAULT_ZONE_TO_LEFT_SPEAKER; } } ALOGD("Out worker:%s zone id %d", out->bus_address, zone_id); while (true) { pthread_mutex_lock(&out->lock); while (out->worker_standby || restart) { restart = false; if (ext_pcm) { ext_pcm_close(ext_pcm); // Frees pcm ext_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:%s not in standby before exiting", out->bus_address); } 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 (!ext_pcm) { ext_pcm = ext_pcm_open(PCM_CARD, PCM_DEVICE, PCM_OUT | PCM_MONOTONIC, &out->pcm_config); if (!ext_pcm_is_ready(ext_pcm)) { ALOGE("pcm_open(out) failed: %s: address %s channels %d format %d rate %d", ext_pcm_get_error(ext_pcm), out->bus_address, 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 = ext_pcm_frames_to_bytes(ext_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); if (is_zone_selected_to_play(out->dev, zone_id)) { int write_error = ext_pcm_write(ext_pcm, out->bus_address, buffer, ext_pcm_frames_to_bytes(ext_pcm, frames)); if (write_error) { ALOGE("pcm_write failed %s address %s", ext_pcm_get_error(ext_pcm), out->bus_address); restart = true; } else { ALOGV("pcm_write succeed address %s", out->bus_address); } } } 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; } } // Applies gain naively, assumes AUDIO_FORMAT_PCM_16_BIT and stereo output static void out_apply_gain(struct generic_stream_out *out, const void *buffer, size_t bytes) { int16_t *int16_buffer = (int16_t *)buffer; size_t int16_size = bytes / sizeof(int16_t); for (int i = 0; i < int16_size; i++) { if ((i % 2) && !(out->enabled_channels & RIGHT_CHANNEL)) { int16_buffer[i] = 0; } else if (!(i % 2) && !(out->enabled_channels & LEFT_CHANNEL)) { int16_buffer[i] = 0; } else { float multiplied = int16_buffer[i] * out->amplitude_ratio; if (multiplied > INT16_MAX) int16_buffer[i] = INT16_MAX; else if (multiplied < INT16_MIN) int16_buffer[i] = INT16_MIN; else int16_buffer[i] = (int16_t)multiplied; } } } 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; ALOGV("%s: to device %s", __func__, out->bus_address); 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 = frames; if (out->dev->master_mute) { ALOGV("%s: ignored due to master mute", __func__); } else { out_apply_gain(out, buffer, bytes); 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("%s Hardware backing HAL too slow, could only write %zu of %zu frames", __func__, 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) { int ret = -EINVAL; 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 size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format, audio_channel_mask_t channel_mask) { size_t size; size_t device_rate; int channel_count = popcount(channel_mask); if (refine_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; int ret = 0; pthread_mutex_lock(&in->lock); if (!in->standby) { ret = -ENOSYS; } else { parms = str_parms_create_str(kvpairs); int val = 0; ret = get_int_value(parms, AUDIO_PARAMETER_STREAM_ROUTING, &val); if (ret >= 0) { in->device = (int)val; ret = 0; } 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; 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) { // TODO(hwwang): support adjusting input gain 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; } // Generates pure tone for FM_TUNER and bus_device static int pseudo_pcm_read(void *data, unsigned int count, struct oscillator *oscillator) { unsigned int length = count / sizeof(int16_t); int16_t *sdata = (int16_t *)data; for (int index = 0; index < length; index++) { sdata[index] = (int16_t)(sin(oscillator->phase) * 4096); oscillator->phase += oscillator->phase_increment; oscillator->phase = oscillator->phase > TWO_PI ? oscillator->phase - TWO_PI : oscillator->phase; } return count; } 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; } 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 bool address_has_tone_keyword(char * address) { return strstr(address, TONE_ADDRESS_KEYWORD) != NULL; } static bool is_tone_generator_device(struct generic_stream_in *in) { return in->device == AUDIO_DEVICE_IN_FM_TUNER || ((in->device == AUDIO_DEVICE_IN_BUS) && address_has_tone_keyword(in->bus_address)); } 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); int ret = 0; 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; } // Tone generators fill the buffer via pseudo_pcm_read directly if (!is_tone_generator_device(in)) { 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 (is_tone_generator_device(in)) { int read_bytes = pseudo_pcm_read(buffer, bytes, &in->oscillator); read_frames = read_bytes / audio_stream_in_frame_size(stream); } else 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_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) { 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); } out->enabled_channels = BOTH_CHANNELS; if (address) { out->bus_address = calloc(strlen(address) + 1, sizeof(char)); strncpy(out->bus_address, address, strlen(address)); hashmapPut(adev->out_bus_stream_map, out->bus_address, out); /* TODO: read struct audio_gain from audio_policy_configuration */ out->gain_stage = (struct audio_gain) { .min_value = -3200, .max_value = 600, .step_value = 100, }; out->amplitude_ratio = 1.0; if (property_get_bool(PROP_KEY_SIMULATE_MULTI_ZONE_AUDIO, false)) { out->enabled_channels = strstr(out->bus_address, AUDIO_ZONE_KEYWORD) ? RIGHT_CHANNEL: LEFT_CHANNEL; ALOGD("%s Routing %s to %s channel", __func__, out->bus_address, out->enabled_channels == RIGHT_CHANNEL ? "Right" : "Left"); } } *stream_out = &out->stream; ALOGD("%s bus: %s", __func__, out->bus_address); error: 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; struct generic_stream_out *out = (struct generic_stream_out *)stream; ALOGD("%s bus:%s", __func__, out->bus_address); 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); if (out->bus_address) { hashmapRemove(adev->out_bus_stream_map, out->bus_address); free(out->bus_address); } free(stream); } static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) { struct generic_audio_device *adev = (struct generic_audio_device *)dev; pthread_mutex_lock(&adev->lock); struct str_parms *parms = str_parms_create_str(kvpairs); int value = 0; int results = get_int_value(parms, AAE_PARAMETER_KEY_FOR_SELECTED_ZONE, &value); if (results >= 0) { adev->last_zone_selected_to_play = value; results = 0; ALOGD("%s Changed play zone id to %d", __func__, adev->last_zone_selected_to_play); } str_parms_destroy(parms); pthread_mutex_unlock(&adev->lock); return results; } static char *adev_get_parameters(const struct audio_hw_device * dev, const char *keys) { return NULL; } 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) { ALOGD("%s: %s", __func__, _bool_str(muted)); struct generic_audio_device *adev = (struct generic_audio_device *)dev; pthread_mutex_lock(&adev->lock); adev->master_mute = muted; pthread_mutex_unlock(&adev->lock); return 0; } static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted) { struct generic_audio_device *adev = (struct generic_audio_device *)dev; pthread_mutex_lock(&adev->lock); *muted = adev->master_mute; pthread_mutex_unlock(&adev->lock); ALOGD("%s: %s", __func__, _bool_str(*muted)); return 0; } 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; } if (in->bus_address) { free(in->bus_address); } pthread_mutex_destroy(&in->lock); audio_vbuffer_destroy(&in->buffer); free(stream); } static void increase_next_tone_frequency(struct generic_audio_device *adev) { adev->next_tone_frequency_to_assign += TONE_FREQUENCY_INCREASE; if (adev->next_tone_frequency_to_assign > MAX_TONE_FREQUENCY) { adev->next_tone_frequency_to_assign = DEFAULT_FREQUENCY; } } static int create_or_fetch_tone_frequency(struct generic_audio_device *adev, char *address, int update_frequency) { int *frequency = hashmapGet(adev->in_bus_tone_frequency_map, address); if (frequency == NULL) { frequency = calloc(1, sizeof(int)); *frequency = update_frequency; hashmapPut(adev->in_bus_tone_frequency_map, strdup(address), frequency); ALOGD("%s assigned frequency %d to %s", __func__, *frequency, address); } return *frequency; } 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, audio_source_t source) { ALOGV("%s: audio_source_t: %d", __func__, source); 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; 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); } if (address) { in->bus_address = strdup(address); if (is_tone_generator_device(in)) { int update_frequency = adev->next_tone_frequency_to_assign; int frequency = create_or_fetch_tone_frequency(adev, address, update_frequency); if (update_frequency == frequency) { increase_next_tone_frequency(adev); } in->oscillator.phase = 0.0f; in->oscillator.phase_increment = (TWO_PI*(frequency)) / ((float) in_get_sample_rate(&in->stream.common)); } } *stream_in = &in->stream; error: return ret; } static int adev_dump(const audio_hw_device_t *dev, int fd) { return 0; } static int adev_set_audio_port_config(struct audio_hw_device *dev, const struct audio_port_config *config) { int ret = 0; struct generic_audio_device *adev = (struct generic_audio_device *)dev; const char *bus_address = config->ext.device.address; struct generic_stream_out *out = hashmapGet(adev->out_bus_stream_map, bus_address); if (out) { pthread_mutex_lock(&out->lock); int gainIndex = (config->gain.values[0] - out->gain_stage.min_value) / out->gain_stage.step_value; int totalSteps = (out->gain_stage.max_value - out->gain_stage.min_value) / out->gain_stage.step_value; int minDb = out->gain_stage.min_value / 100; int maxDb = out->gain_stage.max_value / 100; // curve: 10^((minDb + (maxDb - minDb) * gainIndex / totalSteps) / 20) out->amplitude_ratio = pow(10, (minDb + (maxDb - minDb) * (gainIndex / (float)totalSteps)) / 20); pthread_mutex_unlock(&out->lock); ALOGD("%s: set audio gain: %f on %s", __func__, out->amplitude_ratio, bus_address); } else { ALOGE("%s: can not find output stream by bus_address:%s", __func__, bus_address); ret = -EINVAL; } return ret; } static int adev_create_audio_patch(struct audio_hw_device *dev, unsigned int num_sources, const struct audio_port_config *sources, unsigned int num_sinks, const struct audio_port_config *sinks, audio_patch_handle_t *handle) { struct generic_audio_device *audio_dev = (struct generic_audio_device *)dev; for (int i = 0; i < num_sources; i++) { ALOGD("%s: source[%d] type=%d address=%s", __func__, i, sources[i].type, sources[i].type == AUDIO_PORT_TYPE_DEVICE ? sources[i].ext.device.address : ""); } for (int i = 0; i < num_sinks; i++) { ALOGD("%s: sink[%d] type=%d address=%s", __func__, i, sinks[i].type, sinks[i].type == AUDIO_PORT_TYPE_DEVICE ? sinks[i].ext.device.address : "N/A"); } if (num_sources == 1 && num_sinks == 1 && sources[0].type == AUDIO_PORT_TYPE_DEVICE && sinks[0].type == AUDIO_PORT_TYPE_DEVICE) { pthread_mutex_lock(&audio_dev->lock); audio_dev->last_patch_id += 1; pthread_mutex_unlock(&audio_dev->lock); *handle = audio_dev->last_patch_id; ALOGD("%s: handle: %d", __func__, *handle); } return 0; } static int adev_release_audio_patch(struct audio_hw_device *dev, audio_patch_handle_t handle) { ALOGD("%s: handle: %d", __func__, handle); 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); } if (adev->out_bus_stream_map) { hashmapFree(adev->out_bus_stream_map); } if (adev->in_bus_tone_frequency_map) { hashmapFree(adev->in_bus_tone_frequency_map); } free(adev); } error: pthread_mutex_unlock(&adev_init_lock); return ret; } /* copied from libcutils/str_parms.c */ static bool str_eq(void *key_a, void *key_b) { return !strcmp((const char *)key_a, (const char *)key_b); } /** * use djb hash unless we find it inadequate. * copied from libcutils/str_parms.c */ #ifdef __clang__ __attribute__((no_sanitize("integer"))) #endif static int str_hash_fn(void *str) { uint32_t hash = 5381; char *p; for (p = str; p && *p; p++) { hash = ((hash << 5) + hash) + *p; } return (int)hash; } 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_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_3_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; adev->device.get_master_mute = adev_get_master_mute; 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; // New in AUDIO_DEVICE_API_VERSION_3_0 adev->device.set_audio_port_config = adev_set_audio_port_config; adev->device.create_audio_patch = adev_create_audio_patch; adev->device.release_audio_patch = adev_release_audio_patch; *device = &adev->device.common; adev->mixer = mixer_open(PCM_CARD); ALOGD("%s Mixer name %s", __func__, mixer_get_name(adev->mixer)); 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; } } // Initialize the bus address to output stream map adev->out_bus_stream_map = hashmapCreate(5, str_hash_fn, str_eq); // Initialize the bus address to input stream map adev->in_bus_tone_frequency_map = hashmapCreate(5, str_hash_fn, str_eq); adev->next_tone_frequency_to_assign = DEFAULT_FREQUENCY; adev->last_zone_selected_to_play = DEFAULT_ZONE_TO_LEFT_SPEAKER; 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 car audio HW HAL", .author = "The Android Open Source Project", .methods = &hal_module_methods, }, };