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
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,
+ },
+};
