initial Raspberry Vanilla AOSP 12 commit

* Audio - Based on AOSP hikey audio HAL: https://android.googlesource.com/device/linaro/hikey/+/refs/heads/master/audio/ - ALSA based on Android-x86 - ALSA loop is used to support HDMI audio on Pi 4 because VC4 HDMI audio devices use SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE that is not supported on Android https://github.com/raspberrypi/linux/issues/4651 https://github.com/raspberrypi/linux/issues/4654 * Bluetooth - AOSP Broadcom vendor stack with some additional fixes from android-rpi * Camera - libcamera for official Raspberry Pi CSI camera modules based on GloDroid - AOSP external camera HAL for UVC USB webcams: https://source.android.com/docs/core/camera/external-usb-cameras * Graphics - OpenGL & Vulkan: upstream Mesa with GloDroid patches - Upstream drm_hwcomposer with force resolution patch from Android-x86 and my additional fixes - minigbm based on GloDroid * Health - Based on AOSP cuttlefish health HAL * Kernel - Merge of Raspberry Pi and AOSP common kernel with my additional fixes and configurations https://github.com/raspberrypi/linux https://android.googlesource.com/kernel/common/ * Lights - Based on LineageOS Xiaomi msm8996-common lights HAL * suspend_blocker - Based on AOSP cuttlefish * v4l2_codec2 - Based on AOSP with my Raspberry Pi 4 specific fixes * Wifi - AOSP Broadcom vendor stack with my additional fixes
2022-09-14 15:06:12 +03:00
commit bbdbc45eea
85 changed files with 3127 additions and 0 deletions
--- a/audio/Android.bp
+++ b/audio/Android.bp
@@ -0,0 +1,24 @@
+// Copyright (C) 2015 The Android Open Source Project
+// Copyright (C) 2021-2022 KonstaKANG
+//
+// SPDX-License-Identifier: Apache-2.0
+
+cc_library_shared {
+    name: "audio.primary.rpi",
+    relative_install_path: "hw",
+    proprietary: true,
+    srcs: ["audio_hw.c"],
+    include_dirs: [
+        "external/expat/lib",
+        "external/tinyalsa/include",
+        "system/media/audio_effects/include",
+        "system/media/audio_utils/include",
+    ],
+    header_libs: ["libhardware_headers"],
+    shared_libs: [
+        "libcutils",
+        "liblog",
+        "libtinyalsa",
+    ],
+    cflags: ["-Wno-unused-parameter"],
+}
--- a/audio/audio_hw.c
+++ b/audio/audio_hw.c
@@ -0,0 +1,708 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ * Copyright (C) 2021-2022 KonstaKANG
+ *
+ * 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_rpi"
+//#define LOG_NDEBUG 0
+
+#include <errno.h>
+#include <malloc.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/time.h>
+#include <stdlib.h>
+#include <unistd.h>
+
+#include <log/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>
+
+
+/* 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;
+};
+
+static int get_pcm_card()
+{
+    char card[PROPERTY_VALUE_MAX];
+    property_get("persist.audio.pcm.card", card, "0");
+    return atoi(card);
+}
+
+static int get_pcm_device()
+{
+    char device[PROPERTY_VALUE_MAX];
+    property_get("persist.audio.pcm.device", device, "0");
+    return atoi(device);
+}
+
+/* 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(get_pcm_card(), get_pcm_device(), 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);
+
+    /* 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 value[32];
+    int val = 0;
+    int ret = -EINVAL;
+
+    if (kvpairs == NULL || kvpairs[0] == 0) {
+        return 0;
+    }
+
+    parms = str_parms_create_str(kvpairs);
+
+    if (str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)) >= 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);
+        ret = 0;
+    }
+
+    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;
+
+    /* 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(get_pcm_card(), get_pcm_device(), 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 __unused *dev,
+        audio_io_handle_t handle,
+        audio_devices_t devices,
+        struct audio_config *config,
+        struct audio_stream_in **stream_in,
+        audio_input_flags_t flags __unused,
+        const char *address __unused,
+        audio_source_t source __unused)
+{
+    struct stub_stream_in *in;
+
+    ALOGV("adev_open_input_stream...");
+
+    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)
+{
+    struct alsa_audio_device *adev;
+
+    ALOGV("adev_open: %s", name);
+
+    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 = "Raspberry Pi audio HW HAL",
+        .author = "The Android Open Source Project",
+        .methods = &hal_module_methods,
+    },
+};
--- a/audio/audio_policy_configuration.xml
+++ b/audio/audio_policy_configuration.xml
@@ -0,0 +1,106 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- 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.
+-->
+<audioPolicyConfiguration version="7.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+    <modules>
+        <module name="primary" halVersion="2.0">
+            <attachedDevices>
+                <item>Speaker</item>
+                <item>Built-In Mic</item>
+            </attachedDevices>
+            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <mixPorts>
+                <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="primary input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="Wired Headset" type="AUDIO_DEVICE_OUT_WIRED_HEADSET" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="Wired Headphones" type="AUDIO_DEVICE_OUT_WIRED_HEADPHONE" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 16000"
+                             channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </devicePort>
+                <devicePort tagName="BT SCO Headset" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 16000"
+                             channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </devicePort>
+                <devicePort tagName="BT SCO Car Kit" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 16000"
+                             channelMasks="AUDIO_CHANNEL_OUT_MONO"/>
+                </devicePort>
+                <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+                </devicePort>
+                <devicePort tagName="Wired Headset Mic" type="AUDIO_DEVICE_IN_WIRED_HEADSET" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 11025 12000 16000 22050 24000 32000 44100 48000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO AUDIO_CHANNEL_IN_STEREO"/>
+                </devicePort>
+                <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000 16000"
+                             channelMasks="AUDIO_CHANNEL_IN_MONO"/>
+                </devicePort>
+            </devicePorts>
+            <routes>
+                <route type="mix" sink="Speaker"
+                       sources="primary output"/>
+                <route type="mix" sink="Wired Headset"
+                       sources="primary output"/>
+                <route type="mix" sink="Wired Headphones"
+                       sources="primary output"/>
+                <route type="mix" sink="BT SCO"
+                       sources="primary output"/>
+                <route type="mix" sink="BT SCO Headset"
+                       sources="primary output"/>
+                <route type="mix" sink="BT SCO Car Kit"
+                       sources="primary output"/>
+                <route type="mix" sink="primary input"
+                       sources="Built-In Mic,Wired Headset Mic,BT SCO Headset Mic"/>
+            </routes>
+        </module>
+
+        <xi:include href="a2dp_in_audio_policy_configuration_7_0.xml"/>
+        <xi:include href="usb_audio_policy_configuration.xml"/>
+        <xi:include href="r_submix_audio_policy_configuration.xml"/>
+        <xi:include href="bluetooth_audio_policy_configuration_7_0.xml"/>
+    </modules>
+
+    <xi:include href="audio_policy_volumes.xml"/>
+    <xi:include href="default_volume_tables.xml"/>
+</audioPolicyConfiguration>
--- a/audio/vc4-hdmi.conf
+++ b/audio/vc4-hdmi.conf
@@ -0,0 +1,68 @@
+# Configuration for the VC4-HDMI sound card using software IEC958
+# subframe conversion
+
+<confdir:pcm/hdmi.conf>
+vc4-hdmi.pcm.hdmi.0 {
+	@args [ CARD AES0 AES1 AES2 AES3 ]
+	@args.CARD {
+		type string
+	}
+	@args.AES0 {
+		type integer
+	}
+	@args.AES1 {
+		type integer
+	}
+	@args.AES2 {
+		type integer
+	}
+	@args.AES3 {
+		type integer
+	}
+	type iec958
+	slave {
+		format IEC958_SUBFRAME_LE
+		pcm {
+			type hooks
+			slave.pcm {
+				type hw
+				card $CARD
+				device 0
+			}
+			hooks.0 {
+				type ctl_elems
+				hook_args [
+				{
+					name "IEC958 Playback Default"
+					interface PCM
+					optional true
+					lock true
+					preserve true
+					value [ $AES0 $AES1 $AES2 $AES3 ]
+				}
+				]
+			}
+		}
+	}
+	status [ $AES0 $AES1 $AES2 $AES3 ]
+}
+
+# default with plug
+vc4-hdmi.pcm.default {
+	@args [ CARD ]
+	@args.CARD {
+		type string
+	}
+	type plug
+	slave.pcm {
+		type softvol
+		slave.pcm {
+			@func concat
+			strings [ "hdmi:" $CARD ]
+		}
+		control {
+			name "PCM Playback Volume"
+			card $CARD
+		}
+	}
+}