Enable audio support

- PCM output works via music player
- Sometimes ringtones fail to play
- Modem routing is untested but present
- PCM input needs to be implemented

Change-Id: Ib58bef9674e1c9bb896be521c3d95c4e07e0442b

2 files changed, 850 insertions, 0 deletions

diff --git a/audio/Android.mk b/audio/Android.mk
new file mode 100644
index 0000000..fe4d1bd
--- /dev/null
+++ b/audio/Android.mk
@@ -0,0 +1,29 @@
+# Copyright (C) 2011 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 := audio.primary.tuna
+LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw
+LOCAL_SRC_FILES := audio_hw.c
+LOCAL_C_INCLUDES += \
+	external/tinyalsa/include \
+	external/speex/include
+LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libspeexresampler
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
+
diff --git a/audio/audio_hw.c b/audio/audio_hw.c
new file mode 100644
index 0000000..86590db
--- /dev/null
+++ b/audio/audio_hw.c
@@ -0,0 +1,821 @@
+/*
+ * Copyright (C) 2011 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_primary"
+/*#define LOG_NDEBUG 0*/
+
+#include <errno.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/time.h>
+#include <stdlib.h>
+
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+
+#include <hardware/hardware.h>
+#include <system/audio.h>
+#include <hardware/audio.h>
+
+#include <tinyalsa/asoundlib.h>
+#include <speex/speex_resampler.h>
+
+/* Mixer control names */
+#define MIXER_DL1_MEDIA_PLAYBACK_VOLUME     "DL1 Media Playback Volume"
+#define MIXER_DL1_VOICE_PLAYBACK_VOLUME     "DL1 Voice Playback Volume"
+#define MIXER_DL2_MEDIA_PLAYBACK_VOLUME     "DL2 Media Playback Volume"
+#define MIXER_SDT_DL_VOLUME                 "SDT DL Volume"
+
+#define MIXER_HEADSET_PLAYBACK_VOLUME       "Headset Playback Volume"
+#define MIXER_HANDSFREE_PLAYBACK_VOLUME     "Handsfree Playback Volume"
+#define MIXER_EARPHONE_PLAYBACK_VOLUME      "Earphone Playback Volume"
+
+#define MIXER_DL1_MIXER_MULTIMEDIA          "DL1 Mixer Multimedia"
+#define MIXER_DL1_MIXER_VOICE               "DL1 Mixer Voice"
+#define MIXER_DL2_MIXER_MULTIMEDIA          "DL2 Mixer Multimedia"
+#define MIXER_SIDETONE_MIXER_PLAYBACK       "Sidetone Mixer Playback"
+#define MIXER_DL1_PDM_SWITCH                "DL1 PDM Switch"
+
+#define MIXER_HS_LEFT_PLAYBACK              "HS Left Playback"
+#define MIXER_HS_RIGHT_PLAYBACK             "HS Right Playback"
+#define MIXER_HF_LEFT_PLAYBACK              "HF Left Playback"
+#define MIXER_HF_RIGHT_PLAYBACK             "HF Right Playback"
+#define MIXER_EARPHONE_DRIVER_SWITCH        "Earphone Driver Switch"
+
+#define MIXER_ANALOG_LEFT_CAPTURE_ROUTE     "Analog Left Capture Route"
+#define MIXER_CAPTURE_PREAMPLIFIER_VOLUME   "Capture Preamplifier Volume"
+#define MIXER_CAPTURE_VOLUME                "Capture Volume"
+#define MIXER_AMIC_UL_VOLUME                "AMIC UL Volume"
+#define MIXER_AUDUL_VOICE_UL_VOLUME         "AUDUL Voice UL Volume"
+
+/* Mixer control gain and route values */
+#define MIXER_ABE_GAIN_0DB                  120
+#define MIXER_ABE_GAIN_MINUS1DB             118
+#define MIXER_CODEC_VOLUME_MAX              15
+#define MIXER_PLAYBACK_HS_DAC               "HS DAC"
+#define MIXER_PLAYBACK_HF_DAC               "HF DAC"
+#define MIXER_MAIN_MIC                      "Main Mic"
+
+/* ALSA ports for OMAP4 */
+#define PORT_MM 0
+#define PORT_MM2_UL 1
+#define PORT_VX 2
+#define PORT_TONES 3
+#define PORT_VIBRA 4
+#define PORT_MODEM 5
+#define PORT_MM_LP 5
+
+#define RESAMPLER_BUFFER_SIZE 8192
+
+struct pcm_config pcm_config_mm = {
+    .channels = 2,
+    .rate = 48000,
+    .period_size = 1024,
+    .period_count = 4,
+    .format = PCM_FORMAT_S16_LE,
+};
+
+struct pcm_config pcm_config_vx = {
+    .channels = 1,
+    .rate = 8000,
+    .period_size = 1024,
+    .period_count = 2,
+    .format = PCM_FORMAT_S16_LE,
+};
+
+#define MIN(x, y) ((x) > (y) ? (y) : (x))
+
+struct route_setting
+{
+    char *ctl_name;
+    int intval;
+    char *strval;
+};
+
+struct route_setting mm_speaker[] = {
+    {
+        .ctl_name = MIXER_DL2_MEDIA_PLAYBACK_VOLUME,
+        .intval = MIXER_ABE_GAIN_MINUS1DB,
+    },
+    {
+        .ctl_name = MIXER_HANDSFREE_PLAYBACK_VOLUME,
+        .intval = 26, /* max for no distortion */
+    },
+    {
+        .ctl_name = MIXER_DL2_MIXER_MULTIMEDIA,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_HF_LEFT_PLAYBACK,
+        .strval = MIXER_PLAYBACK_HF_DAC,
+    },
+    {
+        .ctl_name = MIXER_HF_RIGHT_PLAYBACK,
+        .strval = MIXER_PLAYBACK_HF_DAC,
+    },
+    {
+        .ctl_name = NULL,
+    },
+};
+
+struct route_setting mm_headset[] = {
+    {
+        .ctl_name = MIXER_DL1_MEDIA_PLAYBACK_VOLUME,
+        .intval = MIXER_ABE_GAIN_MINUS1DB,
+    },
+    {
+        .ctl_name = MIXER_SDT_DL_VOLUME,
+        .intval = MIXER_ABE_GAIN_0DB,
+    },
+    {
+        .ctl_name = MIXER_HEADSET_PLAYBACK_VOLUME,
+        .intval = 8, /* reasonable maximum */
+    },
+    {
+        .ctl_name = MIXER_DL1_MIXER_MULTIMEDIA,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_SIDETONE_MIXER_PLAYBACK,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_DL1_PDM_SWITCH,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_HS_LEFT_PLAYBACK,
+        .strval = MIXER_PLAYBACK_HS_DAC,
+    },
+    {
+        .ctl_name = MIXER_HS_RIGHT_PLAYBACK,
+        .strval = MIXER_PLAYBACK_HS_DAC,
+    },
+    {
+        .ctl_name = NULL,
+    },
+};
+
+struct route_setting modem[] = {
+    {
+        .ctl_name = MIXER_DL1_MEDIA_PLAYBACK_VOLUME,
+        .intval = MIXER_ABE_GAIN_MINUS1DB,
+    },
+    {
+        .ctl_name = MIXER_SDT_DL_VOLUME,
+        .intval = MIXER_ABE_GAIN_0DB,
+    },
+    {
+        .ctl_name = MIXER_HEADSET_PLAYBACK_VOLUME,
+        .intval = 8, /* reasonable maximum */
+    },
+    {
+        .ctl_name = MIXER_DL1_MIXER_MULTIMEDIA,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_SIDETONE_MIXER_PLAYBACK,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_DL1_PDM_SWITCH,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_HS_LEFT_PLAYBACK,
+        .strval = MIXER_PLAYBACK_HS_DAC,
+    },
+    {
+        .ctl_name = MIXER_HS_RIGHT_PLAYBACK,
+        .strval = MIXER_PLAYBACK_HS_DAC,
+    },
+    {
+        .ctl_name = MIXER_DL1_VOICE_PLAYBACK_VOLUME,
+        .intval = MIXER_ABE_GAIN_MINUS1DB,
+    },
+    {
+        .ctl_name = MIXER_DL1_MIXER_VOICE,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_ANALOG_LEFT_CAPTURE_ROUTE,
+        .strval = MIXER_MAIN_MIC,
+    },
+    {
+        .ctl_name = MIXER_CAPTURE_PREAMPLIFIER_VOLUME,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_CAPTURE_VOLUME,
+        .intval = 2,
+    },
+    {
+        .ctl_name = MIXER_AMIC_UL_VOLUME,
+        .intval = MIXER_ABE_GAIN_0DB,
+    },
+    {
+        .ctl_name = MIXER_AUDUL_VOICE_UL_VOLUME,
+        .intval = MIXER_ABE_GAIN_0DB,
+    },
+    {
+        .ctl_name = NULL,
+    },
+};
+
+struct route_setting earphone_switch[] = {
+    {
+        .ctl_name = MIXER_EARPHONE_DRIVER_SWITCH,
+        .intval = 1,
+    },
+    {
+        .ctl_name = MIXER_EARPHONE_PLAYBACK_VOLUME,
+        .intval = 10, /* reasonable maximum */
+    },
+    {
+        .ctl_name = NULL,
+    },
+};
+
+struct tuna_audio_device {
+    struct audio_hw_device device;
+
+    pthread_mutex_t lock;
+    struct mixer *mixer;
+    int mode;
+    int out_device;
+};
+
+struct tuna_stream_out {
+    struct audio_stream_out stream;
+
+    pthread_mutex_t lock;
+    struct pcm_config config;
+    struct pcm *pcm;
+    SpeexResamplerState *speex;
+    char *buffer;
+
+    struct tuna_audio_device *dev;
+};
+
+struct tuna_stream_in {
+    struct audio_stream_in stream;
+
+    struct pcm *pcm;
+};
+
+/* The enable flag when 0 makes the assumption that enums are disabled by
+ * "Off" and integers/booleans by 0 */
+static int set_route_by_array(struct mixer *mixer, struct route_setting *route,
+                              int enable)
+{
+    struct mixer_ctl *ctl;
+    unsigned int i, j;
+
+    /* Go through the route array and set each value */
+    i = 0;
+    while (route[i].ctl_name) {
+        ctl = mixer_get_ctl_by_name(mixer, route[i].ctl_name);
+        if (!ctl)
+            return -EINVAL;
+
+        if (route[i].strval) {
+            if (enable)
+                mixer_ctl_set_enum_by_string(ctl, route[i].strval);
+            else
+                mixer_ctl_set_enum_by_string(ctl, "Off");
+        } else {
+            /* This ensures multiple (i.e. stereo) values are set jointly */
+            for (j = 0; j < mixer_ctl_get_num_values(ctl); j++) {
+                if (enable)
+                    mixer_ctl_set_value(ctl, j, route[i].intval);
+                else
+                    mixer_ctl_set_value(ctl, j, 0);
+            }
+        }
+        i++;
+    }
+
+    return 0;
+}
+
+static int select_route(struct tuna_audio_device *adev)
+{
+    if (adev->mode == AUDIO_MODE_IN_CALL) {
+        /* todo: modem routing is untested */
+        set_route_by_array(adev->mixer, modem, 1);
+        set_route_by_array(adev->mixer, earphone_switch, 1);
+    } else if (adev->mode == AUDIO_MODE_NORMAL) {
+        set_route_by_array(adev->mixer, modem, 0);
+
+        switch (adev->out_device) {
+        case AUDIO_DEVICE_OUT_SPEAKER:
+            set_route_by_array(adev->mixer, mm_speaker, 1);
+            set_route_by_array(adev->mixer, mm_headset, 0);
+            set_route_by_array(adev->mixer, earphone_switch, 0);
+            break;
+        case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+            set_route_by_array(adev->mixer, mm_headset, 1);
+            set_route_by_array(adev->mixer, mm_speaker, 0);
+            set_route_by_array(adev->mixer, earphone_switch, 0);
+            break;
+        case AUDIO_DEVICE_OUT_EARPIECE:
+            set_route_by_array(adev->mixer, mm_headset, 1);
+            set_route_by_array(adev->mixer, mm_speaker, 0);
+            set_route_by_array(adev->mixer, earphone_switch, 1);
+            break;
+        default:
+            /* off */
+            break;
+        };
+    }
+
+    return 0;
+}
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+    return 44100;
+}
+
+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)
+{
+    struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
+
+    return pcm_get_buffer_size(out->pcm);
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+    return AUDIO_CHANNEL_OUT_STEREO;
+}
+
+static int out_get_format(const struct audio_stream *stream)
+{
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int out_set_format(struct audio_stream *stream, int format)
+{
+    return 0;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+    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)
+{
+    struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
+    struct tuna_audio_device *adev = out->dev;
+    struct str_parms *parms;
+    char *str;
+    char value[32];
+    int ret;
+
+    parms = str_parms_create_str(kvpairs);
+    pthread_mutex_lock(&adev->lock);
+
+    ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+    if (ret >= 0) {
+        if (adev->out_device != atoi(value)) {
+            adev->out_device = atoi(value);
+            select_route(adev);
+        }
+    }
+
+    pthread_mutex_unlock(&adev->lock);
+    str_parms_destroy(parms);
+    return ret;
+}
+
+static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+    return strdup("");
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+    int bytes_per_sample;
+
+    if (pcm_config_mm.format == PCM_FORMAT_S32_LE)
+        bytes_per_sample = 4;
+    else
+        bytes_per_sample = 2;
+
+    return (pcm_config_mm.period_size * pcm_config_mm.period_count * 1000) /
+           (44100 * pcm_config_mm.channels * bytes_per_sample);
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+                          float right)
+{
+    return -ENOSYS;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
+                         size_t bytes)
+{
+    int ret;
+    struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
+    struct tuna_audio_device *adev = out->dev;
+    spx_uint32_t in_frames = bytes / 4; /* todo */
+    spx_uint32_t out_frames = RESAMPLER_BUFFER_SIZE / 4;
+    unsigned int total_bytes;
+    unsigned int max_bytes;
+    unsigned int remaining_bytes;
+    unsigned int pos;
+
+    pthread_mutex_lock(&out->lock);
+    speex_resampler_process_interleaved_int(out->speex, buffer, &in_frames,
+                                            (spx_int16_t *)out->buffer,
+                                            &out_frames);
+
+    total_bytes = out_frames * 4;
+    max_bytes = pcm_get_buffer_size(out->pcm);
+    remaining_bytes = total_bytes;
+    for (pos = 0; pos < total_bytes; pos += max_bytes) {
+        int bytes_to_write = MIN(max_bytes, remaining_bytes);
+
+        ret = pcm_write(out->pcm, (void *)(out->buffer + pos), bytes_to_write);
+
+        if (ret != 0) {
+            usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
+                   out_get_sample_rate(&stream->common));
+            pthread_mutex_unlock(&out->lock);
+            return bytes;
+        }
+
+        remaining_bytes -= bytes_to_write;
+    }
+
+    pthread_mutex_unlock(&out->lock);
+    return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+                                   uint32_t *dsp_frames)
+{
+    return -EINVAL;
+}
+
+/** audio_stream_in implementation **/
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+    return 8000;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+    return 0;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+    return 320;
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+    return AUDIO_CHANNEL_IN_MONO;
+}
+
+static int in_get_format(const struct audio_stream *stream)
+{
+    return AUDIO_FORMAT_PCM_16_BIT;
+}
+
+static int in_set_format(struct audio_stream *stream, int format)
+{
+    return 0;
+}
+
+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)
+{
+    /* XXX: fake timing for audio input */
+    usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
+           in_get_sample_rate(&stream->common));
+    return bytes;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+    return 0;
+}
+
+
+static int adev_open_output_stream(struct audio_hw_device *dev,
+                                   uint32_t devices, int *format,
+                                   uint32_t *channels, uint32_t *sample_rate,
+                                   struct audio_stream_out **stream_out)
+{
+    struct tuna_audio_device *ladev = (struct tuna_audio_device *)dev;
+    struct tuna_stream_out *out;
+    int ret;
+
+    out = (struct tuna_stream_out *)calloc(1, sizeof(struct tuna_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.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->config = pcm_config_mm;
+
+    out->pcm = pcm_open(0, PORT_MM, PCM_OUT, &out->config);
+    if (!pcm_is_ready(out->pcm)) {
+        LOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm));
+        pcm_close(out->pcm);
+        ret = -ENOMEM;
+        goto err_open;
+    }
+
+    out->speex = speex_resampler_init(2, 44100, 48000,
+                                      SPEEX_RESAMPLER_QUALITY_DEFAULT, &ret);
+    speex_resampler_reset_mem(out->speex);
+    out->buffer = malloc(RESAMPLER_BUFFER_SIZE); /* todo: allow for reallocing */
+
+    out->dev = ladev;
+
+    *stream_out = &out->stream;
+    return 0;
+
+err_open:
+    free(out);
+    *stream_out = NULL;
+    return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev,
+                                     struct audio_stream_out *stream)
+{
+    struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
+
+    free(out->buffer);
+    speex_resampler_destroy(out->speex);
+    pcm_close(out->pcm);
+    free(stream);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+    return -ENOSYS;
+}
+
+static char * adev_get_parameters(const struct audio_hw_device *dev,
+                                  const char *keys)
+{
+    return NULL;
+}
+
+static int adev_init_check(const struct audio_hw_device *dev)
+{
+    return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
+{
+    return -ENOSYS;
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, int mode)
+{
+    struct tuna_audio_device *adev = (struct tuna_audio_device *)dev;
+
+    pthread_mutex_lock(&adev->lock);
+    if (adev->mode != mode) {
+        adev->mode = mode;
+        select_route(adev);
+    }
+    pthread_mutex_unlock(&adev->lock);
+
+    return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+    return -ENOSYS;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+    return -ENOSYS;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
+                                         uint32_t sample_rate, int format,
+                                         int channel_count)
+{
+    return 320;
+}
+
+static int adev_open_input_stream(struct audio_hw_device *dev, uint32_t devices,
+                                  int *format, uint32_t *channels,
+                                  uint32_t *sample_rate,
+                                  audio_in_acoustics_t acoustics,
+                                  struct audio_stream_in **stream_in)
+{
+    struct tuna_audio_device *ladev = (struct tuna_audio_device *)dev;
+    struct tuna_stream_in *in;
+    int ret;
+
+    in = (struct tuna_stream_in *)calloc(1, sizeof(struct tuna_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.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;
+
+err_open:
+    free(in);
+    *stream_in = NULL;
+    return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+                                   struct audio_stream_in *in)
+{
+    return;
+}
+
+static int adev_dump(const audio_hw_device_t *device, int fd)
+{
+    return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+    struct tuna_audio_device *adev = (struct tuna_audio_device *)device;
+
+    mixer_close(adev->mixer);
+    free(device);
+    return 0;
+}
+
+static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev)
+{
+    return (/* OUT */
+            AUDIO_DEVICE_OUT_EARPIECE |
+            AUDIO_DEVICE_OUT_SPEAKER |
+            AUDIO_DEVICE_OUT_WIRED_HEADSET |
+            AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
+            AUDIO_DEVICE_OUT_AUX_DIGITAL |
+            AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
+            AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET |
+            AUDIO_DEVICE_OUT_ALL_SCO |
+            AUDIO_DEVICE_OUT_DEFAULT |
+            /* IN */
+            AUDIO_DEVICE_IN_COMMUNICATION |
+            AUDIO_DEVICE_IN_AMBIENT |
+            AUDIO_DEVICE_IN_BUILTIN_MIC |
+            AUDIO_DEVICE_IN_WIRED_HEADSET |
+            AUDIO_DEVICE_IN_AUX_DIGITAL |
+            AUDIO_DEVICE_IN_BACK_MIC |
+            AUDIO_DEVICE_IN_ALL_SCO |
+            AUDIO_DEVICE_IN_DEFAULT);
+}
+
+static int adev_open(const hw_module_t* module, const char* name,
+                     hw_device_t** device)
+{
+    struct tuna_audio_device *adev;
+    int ret;
+
+    if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
+        return -EINVAL;
+
+    adev = calloc(1, sizeof(struct tuna_audio_device));
+    if (!adev)
+        return -ENOMEM;
+
+    adev->device.common.tag = HARDWARE_DEVICE_TAG;
+    adev->device.common.version = 0;
+    adev->device.common.module = (struct hw_module_t *) module;
+    adev->device.common.close = adev_close;
+
+    adev->device.get_supported_devices = adev_get_supported_devices;
+    adev->device.init_check = adev_init_check;
+    adev->device.set_voice_volume = adev_set_voice_volume;
+    adev->device.set_master_volume = adev_set_master_volume;
+    adev->device.set_mode = adev_set_mode;
+    adev->device.set_mic_mute = adev_set_mic_mute;
+    adev->device.get_mic_mute = adev_get_mic_mute;
+    adev->device.set_parameters = adev_set_parameters;
+    adev->device.get_parameters = adev_get_parameters;
+    adev->device.get_input_buffer_size = adev_get_input_buffer_size;
+    adev->device.open_output_stream = adev_open_output_stream;
+    adev->device.close_output_stream = adev_close_output_stream;
+    adev->device.open_input_stream = adev_open_input_stream;
+    adev->device.close_input_stream = adev_close_input_stream;
+    adev->device.dump = adev_dump;
+
+    adev->mixer = mixer_open(0);
+    if (!adev->mixer) {
+        free(adev);
+        return -ENOMEM;
+    }
+
+    adev->mode = AUDIO_MODE_INVALID;
+    adev->out_device = 0;
+
+    *device = &adev->device.common;
+
+    return 0;
+}
+
+static struct hw_module_methods_t hal_module_methods = {
+    .open = adev_open,
+};
+
+struct audio_module HAL_MODULE_INFO_SYM = {
+    .common = {
+        .tag = HARDWARE_MODULE_TAG,
+        .version_major = 1,
+        .version_minor = 0,
+        .id = AUDIO_HARDWARE_MODULE_ID,
+        .name = "Tuna audio HW HAL",
+        .author = "The Android Open Source Project",
+        .methods = &hal_module_methods,
+    },
+};