Merge "kylin: Add audio, base on intel's" into m-brillo-dev-kylin
diff --git a/peripheral/audio/generic/Android.mk b/peripheral/audio/generic/Android.mk
new file mode 100644
index 0000000..f211787
--- /dev/null
+++ b/peripheral/audio/generic/Android.mk
@@ -0,0 +1,49 @@
+# 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.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_SRC_FILES := \
+ audio_hal.c
+LOCAL_C_INCLUDES += \
+ external/tinyalsa/include \
+ $(call include-path-for, audio-utils) \
+ $(call include-path-for, alsa-utils)
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioutils libalsautils
+LOCAL_MODULE_TAGS := optional
+# setting to build for primary audio or usb audio
+# set -DTARGET_AUDIO_PRIMARY to 1 for Primary (audio jack)
+# set -DTARGET_AUDIO_PRIMARY to 0 for USB audio
+LOCAL_CFLAGS := -Wno-unused-parameter -DTARGET_AUDIO_PRIMARY=1
+LOCAL_MODULE := audio.primary.$(TARGET_BOARD_PLATFORM)
+include $(BUILD_SHARED_LIBRARY)
+
+include $(CLEAR_VARS)
+LOCAL_MODULE_RELATIVE_PATH := hw
+LOCAL_SRC_FILES := \
+ audio_hal.c
+LOCAL_C_INCLUDES += \
+ external/tinyalsa/include \
+ $(call include-path-for, audio-utils) \
+ $(call include-path-for, alsa-utils)
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioutils libalsautils
+LOCAL_MODULE_TAGS := optional
+# setting to build for primary audio or usb audio
+# set -DTARGET_AUDIO_PRIMARY to 1 for Primary (audio jack)
+# set -DTARGET_AUDIO_PRIMARY to 0 for USB audio
+LOCAL_CFLAGS := -Wno-unused-parameter -DTARGET_AUDIO_PRIMARY=0
+LOCAL_MODULE := audio.usb.$(TARGET_BOARD_PLATFORM)
+include $(BUILD_SHARED_LIBRARY)
diff --git a/peripheral/audio/generic/audio_hal.c b/peripheral/audio/generic/audio_hal.c
new file mode 100644
index 0000000..8cc1639
--- /dev/null
+++ b/peripheral/audio/generic/audio_hal.c
@@ -0,0 +1,1272 @@
+/*
+ * 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 "modules.audio.audio_hal"
+/*#define LOG_NDEBUG 0*/
+
+#include <errno.h>
+#include <inttypes.h>
+#include <math.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <stdlib.h>
+#include <sys/time.h>
+
+#include <log/log.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+
+#include <hardware/audio.h>
+#include <hardware/audio_alsaops.h>
+#include <hardware/hardware.h>
+
+#include <system/audio.h>
+
+#include <tinyalsa/asoundlib.h>
+
+#include <audio_utils/channels.h>
+
+#include <dirent.h>
+#include <sys/ioctl.h>
+#include <fcntl.h>
+#include <sound/asound.h>
+
+
+#define PCM_DEV_STR "pcm"
+#if TARGET_AUDIO_PRIMARY
+#define AUDIO_STR "rt5616"
+#else
+#define AUDIO_STR "USB Audio"
+#endif
+#define MAX_PATH_LEN 30
+
+#define NBR_RETRIES 5
+#define RETRY_WAIT_USEC 20000
+
+/* FOR TESTING:
+ * Set k_force_channels to force the number of channels to present to AudioFlinger.
+ * 0 disables (this is default: present the device channels to AudioFlinger).
+ * 2 forces to legacy stereo mode.
+ *
+ * Others values can be tried (up to 8).
+ * TODO: AudioFlinger cannot support more than 8 active output channels
+ * at this time, so limiting logic needs to be put here or communicated from above.
+ */
+static const unsigned k_force_channels = 0;
+
+#include "alsa_device_profile.h"
+#include "alsa_device_proxy.h"
+#include "alsa_logging.h"
+
+#define DEFAULT_INPUT_BUFFER_SIZE_MS 20
+
+// stereo channel count
+#define FCC_2 2
+// fixed channel count of 8 limitation (for data processing in AudioFlinger)
+#define FCC_8 8
+
+struct audio_device {
+ struct audio_hw_device hw_device;
+
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+
+ /* output */
+ alsa_device_profile out_profile;
+
+ /* input */
+ alsa_device_profile in_profile;
+
+ bool mic_muted;
+
+ bool standby;
+#if TARGET_AUDIO_PRIMARY
+ unsigned int master_volume;
+#endif
+};
+
+struct stream_out {
+ struct audio_stream_out stream;
+
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
+ bool standby;
+
+ struct audio_device *dev; /* hardware information - only using this for the lock */
+
+ alsa_device_profile * profile; /* Points to the alsa_device_profile in the audio_device */
+ alsa_device_proxy proxy; /* state of the stream */
+
+ unsigned hal_channel_count; /* channel count exposed to AudioFlinger.
+ * This may differ from the device channel count when
+ * the device is not compatible with AudioFlinger
+ * capabilities, e.g. exposes too many channels or
+ * too few channels. */
+ audio_channel_mask_t hal_channel_mask; /* channel mask exposed to AudioFlinger. */
+
+ void * conversion_buffer; /* any conversions are put into here
+ * they could come from here too if
+ * there was a previous conversion */
+ size_t conversion_buffer_size; /* in bytes */
+};
+
+struct stream_in {
+ struct audio_stream_in stream;
+
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by capture thread */
+ bool standby;
+
+ struct audio_device *dev; /* hardware information - only using this for the lock */
+
+ alsa_device_profile * profile; /* Points to the alsa_device_profile in the audio_device */
+ alsa_device_proxy proxy; /* state of the stream */
+
+ unsigned hal_channel_count; /* channel count exposed to AudioFlinger.
+ * This may differ from the device channel count when
+ * the device is not compatible with AudioFlinger
+ * capabilities, e.g. exposes too many channels or
+ * too few channels. */
+ audio_channel_mask_t hal_channel_mask; /* channel mask exposed to AudioFlinger. */
+
+ /* We may need to read more data from the device in order to data reduce to 16bit, 4chan */
+ void * conversion_buffer; /* any conversions are put into here
+ * they could come from here too if
+ * there was a previous conversion */
+ size_t conversion_buffer_size; /* in bytes */
+};
+
+/*
+ * NOTE: when multiple mutexes have to be acquired, always take the
+ * stream_in or stream_out mutex first, followed by the audio_device mutex.
+ * stream pre_lock is always acquired before stream lock to prevent starvation of control thread by
+ * higher priority playback or capture thread.
+ */
+
+
+static int in_stream_card_number = -1, out_stream_card_number = -1;
+
+
+/*
+ * Examines a pcm-device file to see if its a Audio device and
+ * returns its card-number. If no match, returns -1.
+ */
+static int first_valid_sound_card(char *pcm_name, bool is_out_stream)
+{
+ int fd;
+ char pcm_dev_path[MAX_PATH_LEN];
+ struct snd_pcm_info info;
+ char type;
+ int pcm_name_length;
+
+ ALOGV("%s enter",__func__);
+
+ pcm_name_length = strlen(pcm_name);
+ if (pcm_name_length < 2) {
+ return -1;
+ }
+ type = is_out_stream ? 'p' : 'c';
+ /* If pcm out then filename must end with 0p/0c */
+ if ((pcm_name[pcm_name_length -2] != '0') && (pcm_name[pcm_name_length - 1] != type)) {
+ ALOGV("%s exit",__func__);
+ return -1;
+ }
+
+ snprintf(pcm_dev_path, sizeof(pcm_dev_path), "/dev/snd/%s", pcm_name);
+ fd = open(pcm_dev_path, O_RDONLY);
+
+ if (fd != -1) {
+ if (!(ioctl(fd, SNDRV_PCM_IOCTL_INFO, &info))) {
+ if (strstr(info.id, AUDIO_STR)) {
+ close(fd);
+ ALOGV("%s exit",__func__);
+ return info.card;
+ }
+ } else {
+ ALOGE("ioctl failed for file: %s", pcm_dev_path);
+ }
+
+ close(fd);
+ }
+
+ ALOGV("%s exit",__func__);
+ return -1;
+}
+
+/*
+ * Returns the number of the first valid Audio card
+ * If none is found, returns -1.
+ */
+static int get_first_sound_card(bool is_out_stream)
+{
+ DIR *dir;
+ struct dirent *de = NULL;
+ int card_nr;
+
+ ALOGV("%s enter",__func__);
+
+ dir = opendir("/dev/snd");
+ if (dir == NULL) {
+ ALOGE("Could not open directory /dev/snd");
+ ALOGV("%s exit",__func__);
+ return -1;
+ }
+
+ while ((de = readdir(dir))) {
+ if (strncmp(de->d_name, PCM_DEV_STR, sizeof(PCM_DEV_STR) - 1) == 0) {
+ if ((card_nr = first_valid_sound_card(de->d_name, is_out_stream)) != -1) {
+ closedir(dir);
+ ALOGV("%s exit",__func__);
+ return card_nr;
+ }
+ }
+ }
+
+ closedir(dir);
+ ALOGW("No card found in /dev/snd");
+ ALOGV("%s exit",__func__);
+ return -1;
+}
+
+static bool parse_card_device_params(bool is_out_stream, int *card, int *device)
+{
+ int try_time;
+ int found_card = -1;
+
+ if (is_out_stream) {
+ if (out_stream_card_number != -1) {
+ *card = out_stream_card_number;
+ *device = 0;
+ return true;
+ }
+ } else {
+ if (in_stream_card_number != -1) {
+ *card = in_stream_card_number;
+ *device = 0;
+ return true;
+ }
+ }
+
+ for (try_time = 0; try_time < NBR_RETRIES; try_time++) {
+ found_card = get_first_sound_card(is_out_stream);
+ if (found_card == -1)
+ usleep(RETRY_WAIT_USEC);
+ else
+ break;
+ }
+
+ if (found_card == -1) {
+ *card = -1;
+ *device = -1;
+ return false;
+ }
+
+ if (is_out_stream) {
+ out_stream_card_number = found_card;
+ } else {
+ in_stream_card_number = found_card;
+ }
+
+ *card = found_card;
+ *device = 0;
+
+ return true;
+}
+
+static char * device_get_parameters(alsa_device_profile * profile, const char * keys)
+{
+ if (profile->card < 0 || profile->device < 0) {
+ return strdup("");
+ }
+
+ struct str_parms *query = str_parms_create_str(keys);
+ struct str_parms *result = str_parms_create();
+
+ /* These keys are from hardware/libhardware/include/audio.h */
+ /* supported sample rates */
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
+ char* rates_list = profile_get_sample_rate_strs(profile);
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
+ rates_list);
+ free(rates_list);
+ }
+
+ /* supported channel counts */
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
+ char* channels_list = profile_get_channel_count_strs(profile);
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_CHANNELS,
+ channels_list);
+ free(channels_list);
+ }
+
+ /* supported sample formats */
+ if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
+ char * format_params = profile_get_format_strs(profile);
+ str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS,
+ format_params);
+ free(format_params);
+ }
+ str_parms_destroy(query);
+
+ char* result_str = str_parms_to_str(result);
+ str_parms_destroy(result);
+
+ ALOGV("device_get_parameters = %s", result_str);
+
+ return result_str;
+}
+
+void lock_input_stream(struct stream_in *in)
+{
+ pthread_mutex_lock(&in->pre_lock);
+ pthread_mutex_lock(&in->lock);
+ pthread_mutex_unlock(&in->pre_lock);
+}
+
+void lock_output_stream(struct stream_out *out)
+{
+ pthread_mutex_lock(&out->pre_lock);
+ pthread_mutex_lock(&out->lock);
+ pthread_mutex_unlock(&out->pre_lock);
+}
+
+/*
+ * HAl Functions
+ */
+/**
+ * NOTE: when multiple mutexes have to be acquired, always respect the
+ * following order: hw device > out stream
+ */
+
+/*
+ * OUT functions
+ */
+
+static uint32_t adjust_volume(const uint32_t volume)
+{
+ /*
+ * map [0, 100] to [0, 25]
+ */
+ return (int)(sqrt(volume) * 2.5f);
+}
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+ uint32_t rate = proxy_get_sample_rate(&((struct stream_out*)stream)->proxy);
+ ALOGV("out_get_sample_rate() = %d", rate);
+ return rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+ return 0;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+ const struct stream_out* out = (const struct stream_out*)stream;
+ size_t buffer_size =
+ proxy_get_period_size(&out->proxy) * audio_stream_out_frame_size(&(out->stream));
+ return buffer_size;
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+ const struct stream_out *out = (const struct stream_out*)stream;
+ return out->hal_channel_mask;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+ /* Note: The HAL doesn't do any FORMAT conversion at this time. It
+ * Relies on the framework to provide data in the specified format.
+ * This could change in the future.
+ */
+ alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
+ audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
+ return format;
+}
+
+static int out_set_format(struct audio_stream *stream, audio_format_t format)
+{
+ return 0;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ lock_output_stream(out);
+ if (!out->standby) {
+ pthread_mutex_lock(&out->dev->lock);
+ proxy_close(&out->proxy);
+ pthread_mutex_unlock(&out->dev->lock);
+ out->standby = true;
+ }
+ pthread_mutex_unlock(&out->lock);
+
+ return 0;
+}
+
+static int out_dump(const struct audio_stream *stream, int fd)
+{
+ return 0;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ ALOGV("out_set_parameters() keys:%s", kvpairs);
+
+ struct stream_out *out = (struct stream_out *)stream;
+
+ int routing = 0;
+ int ret_value = 0;
+ int card = -1;
+ int device = -1;
+
+ if (!parse_card_device_params(true, &card, &device)) {
+ // nothing to do
+ return ret_value;
+ }
+
+ lock_output_stream(out);
+ /* Lock the device because that is where the profile lives */
+ pthread_mutex_lock(&out->dev->lock);
+
+ if (!profile_is_cached_for(out->profile, card, device)) {
+ /* cannot read pcm device info if playback is active */
+ if (!out->standby)
+ ret_value = -ENOSYS;
+ else {
+ int saved_card = out->profile->card;
+ int saved_device = out->profile->device;
+ out->profile->card = card;
+ out->profile->device = device;
+ ret_value = profile_read_device_info(out->profile) ? 0 : -EINVAL;
+ if (ret_value != 0) {
+ out->profile->card = saved_card;
+ out->profile->device = saved_device;
+ }
+ }
+ }
+
+ pthread_mutex_unlock(&out->dev->lock);
+ pthread_mutex_unlock(&out->lock);
+
+ return ret_value;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ lock_output_stream(out);
+ pthread_mutex_lock(&out->dev->lock);
+
+ char * params_str = device_get_parameters(out->profile, keys);
+
+ pthread_mutex_unlock(&out->lock);
+ pthread_mutex_unlock(&out->dev->lock);
+
+ return params_str;
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+ alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
+ return proxy_get_latency(proxy);
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left, float right)
+{
+ return -ENOSYS;
+}
+
+/* must be called with hw device and output stream mutexes locked */
+static int start_output_stream(struct stream_out *out)
+{
+ ALOGV("start_output_stream(card:%d device:%d)", out->profile->card, out->profile->device);
+
+ return proxy_open(&out->proxy);
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
+{
+ int ret;
+ struct stream_out *out = (struct stream_out *)stream;
+
+ lock_output_stream(out);
+ if (out->standby) {
+ pthread_mutex_lock(&out->dev->lock);
+ ret = start_output_stream(out);
+ pthread_mutex_unlock(&out->dev->lock);
+ if (ret != 0) {
+ goto err;
+ }
+ out->standby = false;
+ }
+
+ alsa_device_proxy* proxy = &out->proxy;
+ const void * write_buff = buffer;
+ int num_write_buff_bytes = bytes;
+ const int num_device_channels = proxy_get_channel_count(proxy); /* what we told alsa */
+ const int num_req_channels = out->hal_channel_count; /* what we told AudioFlinger */
+ if (num_device_channels != num_req_channels) {
+ /* allocate buffer */
+ const size_t required_conversion_buffer_size =
+ bytes * num_device_channels / num_req_channels;
+ if (required_conversion_buffer_size > out->conversion_buffer_size) {
+ out->conversion_buffer_size = required_conversion_buffer_size;
+ out->conversion_buffer = realloc(out->conversion_buffer,
+ out->conversion_buffer_size);
+ }
+ /* convert data */
+ const audio_format_t audio_format = out_get_format(&(out->stream.common));
+ const unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
+ num_write_buff_bytes =
+ adjust_channels(write_buff, num_req_channels,
+ out->conversion_buffer, num_device_channels,
+ sample_size_in_bytes, num_write_buff_bytes);
+ write_buff = out->conversion_buffer;
+ }
+
+ if (write_buff != NULL && num_write_buff_bytes != 0) {
+ proxy_write(&out->proxy, write_buff, num_write_buff_bytes);
+ }
+
+ pthread_mutex_unlock(&out->lock);
+
+ return bytes;
+
+err:
+ pthread_mutex_unlock(&out->lock);
+ if (ret != 0) {
+ usleep(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)
+{
+ return -EINVAL;
+}
+
+static int out_get_presentation_position(const struct audio_stream_out *stream,
+ uint64_t *frames, struct timespec *timestamp)
+{
+ struct stream_out *out = (struct stream_out *)stream; // discard const qualifier
+ lock_output_stream(out);
+
+ const alsa_device_proxy *proxy = &out->proxy;
+ const int ret = proxy_get_presentation_position(proxy, frames, timestamp);
+
+ pthread_mutex_unlock(&out->lock);
+ ALOGV("out_get_presentation_position() status:%d frames:%llu",
+ ret, (unsigned long long)*frames);
+ return ret;
+}
+
+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 -EINVAL;
+}
+
+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() handle:0x%X, device:0x%X, flags:0x%X, addr:%s",
+ handle, devices, flags, address);
+
+ struct audio_device *adev = (struct audio_device *)dev;
+
+ struct stream_out *out;
+ out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
+ if (!out)
+ return -ENOMEM;
+
+ /* setup function pointers */
+ 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);
+ pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL);
+
+ out->dev = adev;
+ pthread_mutex_lock(&adev->lock);
+ out->profile = &adev->out_profile;
+
+ // build this to hand to the alsa_device_proxy
+ struct pcm_config proxy_config;
+ memset(&proxy_config, 0, sizeof(proxy_config));
+
+ /* Pull out the card/device pair */
+ parse_card_device_params(true, &(out->profile->card), &(out->profile->device));
+
+ profile_read_device_info(out->profile);
+
+ pthread_mutex_unlock(&adev->lock);
+
+ int ret = 0;
+
+ /* Rate */
+ if (config->sample_rate == 0) {
+ proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
+ } else if (profile_is_sample_rate_valid(out->profile, config->sample_rate)) {
+ proxy_config.rate = config->sample_rate;
+ } else {
+ ALOGE("%s: The requested sample rate (%d) is not valid", __func__, config->sample_rate);
+ proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
+ ret = -EINVAL;
+ }
+
+ /* Format */
+ if (config->format == AUDIO_FORMAT_DEFAULT) {
+ proxy_config.format = profile_get_default_format(out->profile);
+ config->format = audio_format_from_pcm_format(proxy_config.format);
+ } else {
+ enum pcm_format fmt = pcm_format_from_audio_format(config->format);
+ if (profile_is_format_valid(out->profile, fmt)) {
+ proxy_config.format = fmt;
+ } else {
+ ALOGE("%s: The requested format (0x%x) is not valid", __func__, config->format);
+ proxy_config.format = profile_get_default_format(out->profile);
+ config->format = audio_format_from_pcm_format(proxy_config.format);
+ ret = -EINVAL;
+ }
+ }
+
+ /* Channels */
+ unsigned proposed_channel_count = 0;
+ if (k_force_channels) {
+ proposed_channel_count = k_force_channels;
+ } else if (config->channel_mask == AUDIO_CHANNEL_NONE) {
+ proposed_channel_count = profile_get_default_channel_count(out->profile);
+ }
+ if (proposed_channel_count != 0) {
+ if (proposed_channel_count <= FCC_2) {
+ // use channel position mask for mono and stereo
+ config->channel_mask = audio_channel_out_mask_from_count(proposed_channel_count);
+ } else {
+ // use channel index mask for multichannel
+ config->channel_mask =
+ audio_channel_mask_for_index_assignment_from_count(proposed_channel_count);
+ }
+ out->hal_channel_count = proposed_channel_count;
+ } else {
+ out->hal_channel_count = audio_channel_count_from_out_mask(config->channel_mask);
+ }
+ /* we can expose any channel mask, and emulate internally based on channel count. */
+ out->hal_channel_mask = config->channel_mask;
+
+ /* no validity checks are needed as proxy_prepare() forces channel_count to be valid.
+ * and we emulate any channel count discrepancies in out_write(). */
+ proxy_config.channels = proposed_channel_count;
+
+ proxy_prepare(&out->proxy, out->profile, &proxy_config);
+
+ /* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
+ ret = 0;
+
+ out->conversion_buffer = NULL;
+ out->conversion_buffer_size = 0;
+
+ out->standby = true;
+
+ *stream_out = &out->stream;
+
+ return ret;
+
+err_open:
+ free(out);
+ *stream_out = NULL;
+ return -ENOSYS;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+ struct audio_stream_out *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ ALOGV("adev_close_output_stream(c:%d d:%d)", out->profile->card, out->profile->device);
+ /* Close the pcm device */
+ out_standby(&stream->common);
+
+ free(out->conversion_buffer);
+
+ out->conversion_buffer = NULL;
+ out->conversion_buffer_size = 0;
+
+ free(stream);
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+ const struct audio_config *config)
+{
+ /* TODO This needs to be calculated based on format/channels/rate */
+ return 320;
+}
+
+/*
+ * IN functions
+ */
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+ uint32_t rate = proxy_get_sample_rate(&((const struct stream_in *)stream)->proxy);
+ ALOGV("in_get_sample_rate() = %d", rate);
+ return rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+ ALOGV("in_set_sample_rate(%d) - NOPE", rate);
+ return -ENOSYS;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+ const struct stream_in * in = ((const struct stream_in*)stream);
+ return proxy_get_period_size(&in->proxy) * audio_stream_in_frame_size(&(in->stream));
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+ const struct stream_in *in = (const struct stream_in*)stream;
+ return in->hal_channel_mask;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+ alsa_device_proxy *proxy = &((struct stream_in*)stream)->proxy;
+ audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
+ return format;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+ ALOGV("in_set_format(%d) - NOPE", format);
+
+ return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ lock_input_stream(in);
+ if (!in->standby) {
+ pthread_mutex_lock(&in->dev->lock);
+ proxy_close(&in->proxy);
+ pthread_mutex_unlock(&in->dev->lock);
+ in->standby = true;
+ }
+
+ pthread_mutex_unlock(&in->lock);
+
+ 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)
+{
+ ALOGV("in_set_parameters() keys:%s", kvpairs);
+
+ struct stream_in *in = (struct stream_in *)stream;
+
+ char value[32];
+ int param_val;
+ int routing = 0;
+ int ret_value = 0;
+ int card = -1;
+ int device = -1;
+
+ if (!parse_card_device_params(false, &card, &device)) {
+ // nothing to do
+ return ret_value;
+ }
+
+ lock_input_stream(in);
+ pthread_mutex_lock(&in->dev->lock);
+
+ if (card >= 0 && device >= 0 && !profile_is_cached_for(in->profile, card, device)) {
+ /* cannot read pcm device info if playback is active */
+ if (!in->standby)
+ ret_value = -ENOSYS;
+ else {
+ int saved_card = in->profile->card;
+ int saved_device = in->profile->device;
+ in->profile->card = card;
+ in->profile->device = device;
+ ret_value = profile_read_device_info(in->profile) ? 0 : -EINVAL;
+ if (ret_value != 0) {
+ in->profile->card = saved_card;
+ in->profile->device = saved_device;
+ }
+ }
+ }
+
+ pthread_mutex_unlock(&in->dev->lock);
+ pthread_mutex_unlock(&in->lock);
+
+ return ret_value;
+}
+
+static char * in_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ lock_input_stream(in);
+ pthread_mutex_lock(&in->dev->lock);
+
+ char * params_str = device_get_parameters(in->profile, keys);
+
+ pthread_mutex_unlock(&in->dev->lock);
+ pthread_mutex_unlock(&in->lock);
+
+ return params_str;
+}
+
+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 in_set_gain(struct audio_stream_in *stream, float gain)
+{
+ return 0;
+}
+
+/* must be called with hw device and output stream mutexes locked */
+static int start_input_stream(struct stream_in *in)
+{
+ ALOGV("ustart_input_stream(card:%d device:%d)", in->profile->card, in->profile->device);
+
+ return proxy_open(&in->proxy);
+}
+
+/* TODO mutex stuff here (see out_write) */
+static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
+{
+ size_t num_read_buff_bytes = 0;
+ void * read_buff = buffer;
+ void * out_buff = buffer;
+ int ret = 0;
+
+ struct stream_in * in = (struct stream_in *)stream;
+
+ lock_input_stream(in);
+ if (in->standby) {
+ pthread_mutex_lock(&in->dev->lock);
+ ret = start_input_stream(in);
+ pthread_mutex_unlock(&in->dev->lock);
+ if (ret != 0) {
+ goto err;
+ }
+ in->standby = false;
+ }
+
+ alsa_device_profile * profile = in->profile;
+
+ /*
+ * OK, we need to figure out how much data to read to be able to output the requested
+ * number of bytes in the HAL format (16-bit, stereo).
+ */
+ num_read_buff_bytes = bytes;
+ int num_device_channels = proxy_get_channel_count(&in->proxy); /* what we told Alsa */
+ int num_req_channels = in->hal_channel_count; /* what we told AudioFlinger */
+
+ if (num_device_channels != num_req_channels) {
+ num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels;
+ }
+
+ /* Setup/Realloc the conversion buffer (if necessary). */
+ if (num_read_buff_bytes != bytes) {
+ if (num_read_buff_bytes > in->conversion_buffer_size) {
+ /*TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
+ (and do these conversions themselves) */
+ in->conversion_buffer_size = num_read_buff_bytes;
+ in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
+ }
+ read_buff = in->conversion_buffer;
+ }
+
+ ret = proxy_read(&in->proxy, read_buff, num_read_buff_bytes);
+ if (ret == 0) {
+ if (num_device_channels != num_req_channels) {
+ // ALOGV("chans dev:%d req:%d", num_device_channels, num_req_channels);
+
+ out_buff = buffer;
+ /* Num Channels conversion */
+ if (num_device_channels != num_req_channels) {
+ audio_format_t audio_format = in_get_format(&(in->stream.common));
+ unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
+
+ num_read_buff_bytes =
+ adjust_channels(read_buff, num_device_channels,
+ out_buff, num_req_channels,
+ sample_size_in_bytes, num_read_buff_bytes);
+ }
+ }
+
+ /* no need to acquire in->dev->lock to read mic_muted here as we don't change its state */
+ if (num_read_buff_bytes > 0 && in->dev->mic_muted)
+ memset(buffer, 0, num_read_buff_bytes);
+ } else {
+ num_read_buff_bytes = 0; // reset the value after headset is unplugged
+ }
+
+err:
+ pthread_mutex_unlock(&in->lock);
+
+ return num_read_buff_bytes;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+ return 0;
+}
+
+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("in adev_open_input_stream() rate:%" PRIu32 ", chanMask:0x%" PRIX32 ", fmt:%" PRIu8,
+ config->sample_rate, config->channel_mask, config->format);
+
+ struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
+ int ret = 0;
+
+ if (in == NULL)
+ return -ENOMEM;
+
+ /* setup function pointers */
+ 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;
+
+ pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
+ pthread_mutex_init(&in->pre_lock, (const pthread_mutexattr_t *) NULL);
+
+ in->dev = (struct audio_device *)dev;
+ pthread_mutex_lock(&in->dev->lock);
+
+ in->profile = &in->dev->in_profile;
+
+ struct pcm_config proxy_config;
+ memset(&proxy_config, 0, sizeof(proxy_config));
+
+ /* Pull out the card/device pair */
+ parse_card_device_params(false, &(in->profile->card), &(in->profile->device));
+
+ profile_read_device_info(in->profile);
+ pthread_mutex_unlock(&in->dev->lock);
+
+ /* Rate */
+ if (config->sample_rate == 0) {
+ proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
+ } else if (profile_is_sample_rate_valid(in->profile, config->sample_rate)) {
+ proxy_config.rate = config->sample_rate;
+ } else {
+ ALOGE("%s: The requested sample rate (%d) is not valid", __func__, config->sample_rate);
+ proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
+ ret = -EINVAL;
+ }
+
+ /* Format */
+ if (config->format == AUDIO_FORMAT_DEFAULT) {
+ proxy_config.format = profile_get_default_format(in->profile);
+ config->format = audio_format_from_pcm_format(proxy_config.format);
+ } else {
+ enum pcm_format fmt = pcm_format_from_audio_format(config->format);
+ if (profile_is_format_valid(in->profile, fmt)) {
+ proxy_config.format = fmt;
+ } else {
+ ALOGE("%s: The requested format (0x%x) is not valid", __func__, config->format);
+ proxy_config.format = profile_get_default_format(in->profile);
+ config->format = audio_format_from_pcm_format(proxy_config.format);
+ ret = -EINVAL;
+ }
+ }
+
+ /* Channels */
+ unsigned proposed_channel_count = 0;
+ if (k_force_channels) {
+ proposed_channel_count = k_force_channels;
+ } else if (config->channel_mask == AUDIO_CHANNEL_NONE) {
+ proposed_channel_count = profile_get_default_channel_count(in->profile);
+ }
+ if (proposed_channel_count != 0) {
+ config->channel_mask = audio_channel_in_mask_from_count(proposed_channel_count);
+ if (config->channel_mask == AUDIO_CHANNEL_INVALID)
+ config->channel_mask =
+ audio_channel_mask_for_index_assignment_from_count(proposed_channel_count);
+ in->hal_channel_count = proposed_channel_count;
+ } else {
+ in->hal_channel_count = audio_channel_count_from_in_mask(config->channel_mask);
+ }
+ /* we can expose any channel mask, and emulate internally based on channel count. */
+ in->hal_channel_mask = config->channel_mask;
+
+ proxy_config.channels = profile_get_default_channel_count(in->profile);
+ proxy_prepare(&in->proxy, in->profile, &proxy_config);
+
+ in->standby = true;
+
+ in->conversion_buffer = NULL;
+ in->conversion_buffer_size = 0;
+
+ *stream_in = &in->stream;
+
+ return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ /* Close the pcm device */
+ in_standby(&stream->common);
+
+ free(in->conversion_buffer);
+
+ free(stream);
+}
+
+/*
+ * ADEV Functions
+ */
+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 -ENOSYS;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+#if TARGET_AUDIO_PRIMARY
+ struct mixer *mixer;
+ struct mixer_ctl *ctl;
+ struct audio_device * adev = (struct audio_device *)dev;
+
+ if ((0 > volume) || (1 < volume) || (NULL == adev))
+ return -EINVAL;
+
+ pthread_mutex_lock(&adev->lock);
+ adev->master_volume = (int)(volume*100);
+
+ if (!(mixer = mixer_open(0))) {
+ pthread_mutex_unlock(&adev->lock);
+ return -ENOSYS;
+ }
+
+ ctl = mixer_get_ctl_by_name(mixer, "HP Playback Volume");
+ mixer_ctl_set_value(ctl,0,adjust_volume(adev->master_volume));
+ mixer_ctl_set_value(ctl,1,adjust_volume(adev->master_volume));
+
+ mixer_close(mixer);
+ pthread_mutex_unlock(&adev->lock);
+ return 0;
+#else
+ return -ENOSYS;
+#endif
+}
+
+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 audio_device * adev = (struct audio_device *)dev;
+ pthread_mutex_lock(&adev->lock);
+ adev->mic_muted = state;
+ pthread_mutex_unlock(&adev->lock);
+ return -ENOSYS;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+ return -ENOSYS;
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+ return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+ struct audio_device *adev = (struct audio_device *)device;
+ free(device);
+
+ return 0;
+}
+
+static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device)
+{
+#if TARGET_AUDIO_PRIMARY
+ struct mixer *mixer;
+ struct mixer_ctl *ctl;
+#endif
+ if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+ return -EINVAL;
+
+ struct audio_device *adev = calloc(1, sizeof(struct audio_device));
+ if (!adev)
+ return -ENOMEM;
+
+ profile_init(&adev->out_profile, PCM_OUT);
+ profile_init(&adev->in_profile, PCM_IN);
+
+ 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.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;
+
+ *device = &adev->hw_device.common;
+#if TARGET_AUDIO_PRIMARY
+ mixer = mixer_open(0);
+
+ if (mixer) {
+ /* setting master volume to value 50 */
+ adev->master_volume = 50;
+
+ int ret = 0;
+ ctl = mixer_get_ctl_by_name(mixer, "HP Playback Switch");
+ ret = mixer_ctl_set_value(ctl,0,1);
+ ret = mixer_ctl_set_value(ctl,1,1);
+ ctl = mixer_get_ctl_by_name(mixer, "HP Playback Volume");
+ mixer_ctl_set_value(ctl,0,adjust_volume(adev->master_volume));
+ mixer_ctl_set_value(ctl,1,adjust_volume(adev->master_volume));
+ ctl = mixer_get_ctl_by_name(mixer, "HPO MIX DAC1 Switch");
+ mixer_ctl_set_value(ctl,0,1);
+ ctl = mixer_get_ctl_by_name(mixer, "HPO MIX DAC1 Switch");
+ mixer_ctl_set_value(ctl,0,1);
+ ctl = mixer_get_ctl_by_name(mixer, "OUT MIXR DAC R1 Switch");
+ mixer_ctl_set_value(ctl,0,1);
+ ctl = mixer_get_ctl_by_name(mixer, "OUT MIXL DAC L1 Switch");
+ mixer_ctl_set_value(ctl,0,1);
+ ctl = mixer_get_ctl_by_name(mixer, "Stereo DAC MIXR DAC R1 Switch");
+ mixer_ctl_set_value(ctl,0,1);
+ ctl = mixer_get_ctl_by_name(mixer, "Stereo DAC MIXL DAC L1 Switch");
+ mixer_ctl_set_value(ctl,0,1);
+
+ mixer_close(mixer);
+ }
+#endif
+ return 0;
+}
+
+static struct hw_module_methods_t hal_module_methods = {
+ .open = adev_open,
+};
+
+struct audio_module HAL_MODULE_INFO_SYM = {
+ .common = {
+ .tag = HARDWARE_MODULE_TAG,
+ .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
+ .hal_api_version = HARDWARE_HAL_API_VERSION,
+ .id = AUDIO_HARDWARE_MODULE_ID,
+ .name = "audio HW HAL",
+ .author = "The Android Open Source Project",
+ .methods = &hal_module_methods,
+ },
+};
diff --git a/peripheral/audio/generic/audio_policy.conf b/peripheral/audio/generic/audio_policy.conf
new file mode 100644
index 0000000..ec3624f
--- /dev/null
+++ b/peripheral/audio/generic/audio_policy.conf
@@ -0,0 +1,69 @@
+#
+# Copyright (C) 2015 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.
+#
+
+# Global configuration section: lists input and output devices always present on the device
+# as well as the output device selected by default.
+# Devices are designated by a string that corresponds to the enum in audio.h
+
+global_configuration {
+ attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
+ default_output_device AUDIO_DEVICE_OUT_SPEAKER
+ attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_REMOTE_SUBMIX
+}
+audio_hw_modules {
+ primary {
+ outputs {
+ primary {
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000|88200|96000|176400|192000|
+ channel_masks AUDIO_CHANNEL_OUT_STEREO
+ formats AUDIO_FORMAT_PCM_16_BIT
+ devices AUDIO_DEVICE_OUT_EARPIECE|AUDIO_DEVICE_OUT_SPEAKER|AUDIO_DEVICE_OUT_WIRED_HEADSET|AUDIO_DEVICE_OUT_WIRED_HEADPHONE|AUDIO_DEVICE_OUT_ALL_SCO|AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET|AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET|AUDIO_DEVICE_OUT_AUX_DIGITAL|AUDIO_DEVICE_OUT_USB_DEVICE
+ flags AUDIO_OUTPUT_FLAG_PRIMARY
+ }
+ }
+ inputs {
+ primary {
+ sampling_rates 8000|11025|16000|22050|32000|44100|48000
+ channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO
+ formats AUDIO_FORMAT_PCM_16_BIT
+ devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET|AUDIO_DEVICE_IN_WIRED_HEADSET|AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET|AUDIO_DEVICE_IN_USB_DEVICE
+ }
+ }
+ }
+ usb {
+ global_configuration {
+ attached_output_devices AUDIO_DEVICE_OUT_USB_DEVICE
+ attached_input_devices AUDIO_DEVICE_IN_USB_DEVICE
+ }
+ outputs {
+ usb_device {
+ sampling_rates dynamic
+ channel_masks dynamic
+ formats dynamic
+ devices AUDIO_DEVICE_OUT_USB_DEVICE
+ flags AUDIO_OUTPUT_FLAG_PRIMARY
+ }
+ }
+ inputs {
+ usb_device {
+ sampling_rates 8000|11025|12000|16000|22050|24000|32000|44100|48000
+ channel_masks AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_STEREO
+ formats AUDIO_FORMAT_PCM_16_BIT
+ devices AUDIO_DEVICE_IN_USB_DEVICE
+ }
+ }
+ }
+}
diff --git a/peripheral/audio/generic/media_codecs.xml b/peripheral/audio/generic/media_codecs.xml
new file mode 100644
index 0000000..934619c
--- /dev/null
+++ b/peripheral/audio/generic/media_codecs.xml
@@ -0,0 +1,82 @@
+<?xml version="1.0" encoding="utf-8" ?>
+<!-- Copyright (C) 2015 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.
+-->
+
+<!--
+<!DOCTYPE MediaCodecs [
+<!ELEMENT Include EMPTY>
+<!ATTLIST Include href CDATA #REQUIRED>
+<!ELEMENT MediaCodecs (Decoders|Encoders|Include)*>
+<!ELEMENT Decoders (MediaCodec|Include)*>
+<!ELEMENT Encoders (MediaCodec|Include)*>
+<!ELEMENT MediaCodec (Type|Quirk|Include)*>
+<!ATTLIST MediaCodec name CDATA #REQUIRED>
+<!ATTLIST MediaCodec type CDATA>
+<!ELEMENT Type EMPTY>
+<!ATTLIST Type name CDATA #REQUIRED>
+<!ELEMENT Quirk EMPTY>
+<!ATTLIST Quirk name CDATA #REQUIRED>
+]>
+
+There's a simple and a complex syntax to declare the availability of a
+media codec:
+
+A codec that properly follows the OpenMax spec and therefore doesn't have any
+quirks and that only supports a single content type can be declared like so:
+
+ <MediaCodec name="OMX.foo.bar" type="something/interesting" />
+
+If a codec has quirks OR supports multiple content types, the following syntax
+can be used:
+
+ <MediaCodec name="OMX.foo.bar" >
+ <Type name="something/interesting" />
+ <Type name="something/else" />
+ ...
+ <Quirk name="requires-allocate-on-input-ports" />
+ <Quirk name="requires-allocate-on-output-ports" />
+ <Quirk name="output-buffers-are-unreadable" />
+ </MediaCodec>
+
+Only the three quirks included above are recognized at this point:
+
+"requires-allocate-on-input-ports"
+ must be advertised if the component does not properly support specification
+ of input buffers using the OMX_UseBuffer(...) API but instead requires
+ OMX_AllocateBuffer to be used.
+
+"requires-allocate-on-output-ports"
+ must be advertised if the component does not properly support specification
+ of output buffers using the OMX_UseBuffer(...) API but instead requires
+ OMX_AllocateBuffer to be used.
+
+"output-buffers-are-unreadable"
+ must be advertised if the emitted output buffers of a decoder component
+ are not readable, i.e. use a custom format even though abusing one of
+ the official OMX colorspace constants.
+ Clients of such decoders will not be able to access the decoded data,
+ naturally making the component much less useful. The only use for
+ a component with this quirk is to render the output to the screen.
+ Audio decoders MUST NOT advertise this quirk.
+ Video decoders that advertise this quirk must be accompanied by a
+ corresponding color space converter for thumbnail extraction,
+ matching surfaceflinger support that can render the custom format to
+ a texture and possibly other code, so just DON'T USE THIS QUIRK.
+
+-->
+
+<MediaCodecs>
+ <Include href="media_codecs_google_audio.xml" />
+</MediaCodecs>
diff --git a/peripheral/audio/generic/peripheral.mk b/peripheral/audio/generic/peripheral.mk
new file mode 100644
index 0000000..1d17efb
--- /dev/null
+++ b/peripheral/audio/generic/peripheral.mk
@@ -0,0 +1,28 @@
+#
+# Copyright 2015 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.
+#
+
+# Audio configuration files
+PRODUCT_COPY_FILES += \
+ frameworks/av/media/libstagefright/data/media_codecs_google_audio.xml:system/etc/media_codecs_google_audio.xml
+
+PRODUCT_COPY_FILES += \
+ hardware/bsp/rockchip/peripheral/audio/generic/media_codecs.xml:system/etc/media_codecs.xml \
+ hardware/bsp/rockchip/peripheral/audio/generic/audio_policy.conf:system/etc/audio_policy.conf
+
+# Primary audio HAL
+DEVICE_PACKAGES += \
+ audio.primary.$(TARGET_BOARD_PLATFORM) \
+ audio.usb.$(TARGET_BOARD_PLATFORM)
