Merge "Add Audio HAL"
diff --git a/audio/Android.mk b/audio/Android.mk
new file mode 100644
index 0000000..e909f2f
--- /dev/null
+++ b/audio/Android.mk
@@ -0,0 +1,37 @@
+# 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.
+
+LOCAL_PATH := $(call my-dir)
+
+# The default audio HAL module, which is a stub, that is loaded if no other
+# device specific modules are present. The exact load order can be seen in
+# libhardware/hardware.c
+#
+# The format of the name is audio.<type>.<hardware/etc>.so where the only
+# required type is 'primary'. Other possibilites are 'a2dp', 'usb', etc.
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := audio.primary.hikey
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_SRC_FILES := audio_hw.c
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa
+LOCAL_CFLAGS := -Wno-unused-parameter
+LOCAL_C_INCLUDES += \
+ external/tinyalsa/include \
+ external/expat/lib \
+ system/media/audio_utils/include \
+ system/media/audio_effects/include
+
+include $(BUILD_SHARED_LIBRARY)
+
diff --git a/audio/audio_hw.c b/audio/audio_hw.c
new file mode 100644
index 0000000..33c569e
--- /dev/null
+++ b/audio/audio_hw.c
@@ -0,0 +1,696 @@
+/*
+ * 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 <cutils/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>
+
+
+#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
+
+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;
+};
+
+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;
+};
+
+
+/* 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);
+ struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
+
+ /* 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);
+ 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 *str;
+ char value[32];
+ int ret, val = 0;
+
+ parms = str_parms_create_str(kvpairs);
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+ if (ret >= 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);
+ }
+
+ 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;
+ int kernel_frames;
+
+ /* 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) {
+ ret = start_output_stream(out);
+ if (ret != 0) {
+ pthread_mutex_unlock(&adev->lock);
+ goto exit;
+ }
+ out->standby = 0;
+ }
+
+ pthread_mutex_unlock(&adev->lock);
+
+ ret = pcm_mmap_write(out->pcm, buffer, out_frames * frame_size);
+ if (ret == 0) {
+ out->written += out_frames;
+ }
+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 *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)
+{
+ ALOGV("adev_open_input_stream...");
+
+ struct stub_audio_device *ladev = (struct stub_audio_device *)dev;
+ struct stub_stream_in *in;
+ int ret;
+
+ 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)
+{
+ ALOGV("adev_close");
+ free(device);
+ return 0;
+}
+
+static int adev_open(const hw_module_t* module, const char* name,
+ hw_device_t** device)
+{
+ ALOGV("adev_open: %s", name);
+
+ struct alsa_audio_device *adev;
+ int ret;
+
+ 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;
+
+ 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,
+ },
+};
diff --git a/device.mk b/device.mk
index 50429f7..ba42933 100644
--- a/device.mk
+++ b/device.mk
@@ -61,6 +61,9 @@
# Build gralloc for hikey
PRODUCT_PACKAGES += gralloc.hikey
+# Build Audio Hal for hikey
+PRODUCT_PACKAGES += audio.primary.hikey
+
# Set zygote config
PRODUCT_DEFAULT_PROPERTY_OVERRIDES += ro.zygote=zygote64_32
PRODUCT_COPY_FILES += system/core/rootdir/init.zygote64_32.rc:root/init.zygote64_32.rc