modules/audio/audio_hw.c - LeafOS-Project/android_hardware_libhardware - Gitiles

 /*
  * 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_default"
 //#define LOG_NDEBUG 0

 #include <errno.h>
 #include <malloc.h>
 #include <pthread.h>
 #include <stdint.h>
 #include <stdlib.h>
 #include <string.h>
 #include <time.h>
 #include <unistd.h>

 #include <log/log.h>

 #include <hardware/audio.h>
 #include <hardware/hardware.h>
 #include <system/audio.h>

 #define STUB_DEFAULT_SAMPLE_RATE   48000
 #define STUB_DEFAULT_AUDIO_FORMAT  AUDIO_FORMAT_PCM_16_BIT

 #define STUB_INPUT_BUFFER_MILLISECONDS  20
 #define STUB_INPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_IN_STEREO

 #define STUB_OUTPUT_BUFFER_MILLISECONDS  10
 #define STUB_OUTPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_OUT_STEREO

 struct stub_audio_device {
     struct audio_hw_device device;
 };

 struct stub_stream_out {
     struct audio_stream_out stream;
     int64_t last_write_time_us;
     uint32_t sample_rate;
     audio_channel_mask_t channel_mask;
     audio_format_t format;
     size_t frame_count;
 };

 struct stub_stream_in {
     struct audio_stream_in stream;
     int64_t last_read_time_us;
     uint32_t sample_rate;
     audio_channel_mask_t channel_mask;
     audio_format_t format;
     size_t frame_count;
 };

 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
 {
     const struct stub_stream_out *out = (const struct stub_stream_out *)stream;

     ALOGV("out_get_sample_rate: %u", out->sample_rate);
     return out->sample_rate;
 }

 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
 {
     struct stub_stream_out *out = (struct stub_stream_out *)stream;

     ALOGV("out_set_sample_rate: %d", rate);
     out->sample_rate = rate;
     return 0;
 }

 static size_t out_get_buffer_size(const struct audio_stream *stream)
 {
     const struct stub_stream_out *out = (const struct stub_stream_out *)stream;
     size_t buffer_size = out->frame_count *
                          audio_stream_out_frame_size(&out->stream);

     ALOGV("out_get_buffer_size: %zu", buffer_size);
     return buffer_size;
 }

 static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
 {
     const struct stub_stream_out *out = (const struct stub_stream_out *)stream;

     ALOGV("out_get_channels: %x", out->channel_mask);
     return out->channel_mask;
 }

 static audio_format_t out_get_format(const struct audio_stream *stream)
 {
     const struct stub_stream_out *out = (const struct stub_stream_out *)stream;

     ALOGV("out_get_format: %d", out->format);
     return out->format;
 }

 static int out_set_format(struct audio_stream *stream, audio_format_t format)
 {
     struct stub_stream_out *out = (struct stub_stream_out *)stream;

     ALOGV("out_set_format: %d", format);
     out->format = format;
     return 0;
 }

 static int out_standby(struct audio_stream *stream)
 {
     ALOGV("out_standby");
     // out->last_write_time_us = 0; unnecessary as a stale write time has same effect
     return 0;
 }

 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");
     return 0;
 }

 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");
     return STUB_OUTPUT_BUFFER_MILLISECONDS;
 }

 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)
 {
     ALOGV("out_write: bytes: %zu", bytes);

     /* XXX: fake timing for audio output */
     struct stub_stream_out *out = (struct stub_stream_out *)stream;
     struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
     clock_gettime(CLOCK_MONOTONIC, &t);
     const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
     const int64_t elapsed_time_since_last_write = now - out->last_write_time_us;
     int64_t sleep_time = bytes * 1000000LL / audio_stream_out_frame_size(stream) /
                out_get_sample_rate(&stream->common) - elapsed_time_since_last_write;
     if (sleep_time > 0) {
         usleep(sleep_time);
     } else {
         // we don't sleep when we exit standby (this is typical for a real alsa buffer).
         sleep_time = 0;
     }
     out->last_write_time_us = now + sleep_time;
     // last_write_time_us is an approximation of when the (simulated) alsa
     // buffer is believed completely full. The usleep above waits for more space
     // in the buffer, but by the end of the sleep the buffer is considered
     // topped-off.
     //
     // On the subsequent out_write(), we measure the elapsed time spent in
     // the mixer. This is subtracted from the sleep estimate based on frames,
     // thereby accounting for drain in the alsa buffer during mixing.
     // This is a crude approximation; we don't handle underruns precisely.
     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_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)
 {
     const struct stub_stream_in *in = (const struct stub_stream_in *)stream;

     ALOGV("in_get_sample_rate: %u", in->sample_rate);
     return in->sample_rate;
 }

 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
 {
     struct stub_stream_in *in = (struct stub_stream_in *)stream;

     ALOGV("in_set_sample_rate: %u", rate);
     in->sample_rate = rate;
     return 0;
 }

 static size_t in_get_buffer_size(const struct audio_stream *stream)
 {
     const struct stub_stream_in *in = (const struct stub_stream_in *)stream;
     size_t buffer_size = in->frame_count *
                          audio_stream_in_frame_size(&in->stream);

     ALOGV("in_get_buffer_size: %zu", buffer_size);
     return buffer_size;
 }

 static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
 {
     const struct stub_stream_in *in = (const struct stub_stream_in *)stream;

     ALOGV("in_get_channels: %x", in->channel_mask);
     return in->channel_mask;
 }

 static audio_format_t in_get_format(const struct audio_stream *stream)
 {
     const struct stub_stream_in *in = (const struct stub_stream_in *)stream;

     ALOGV("in_get_format: %d", in->format);
     return in->format;
 }

 static int in_set_format(struct audio_stream *stream, audio_format_t format)
 {
     struct stub_stream_in *in = (struct stub_stream_in *)stream;

     ALOGV("in_set_format: %d", format);
     in->format = format;
     return 0;
 }

 static int in_standby(struct audio_stream *stream)
 {
     struct stub_stream_in *in = (struct stub_stream_in *)stream;
     in->last_read_time_us = 0;
     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 */
     struct stub_stream_in *in = (struct stub_stream_in *)stream;
     struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
     clock_gettime(CLOCK_MONOTONIC, &t);
     const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;

     // we do a full sleep when exiting standby.
     const bool standby = in->last_read_time_us == 0;
     const int64_t elapsed_time_since_last_read = standby ?
             0 : now - in->last_read_time_us;
     int64_t sleep_time = bytes * 1000000LL / audio_stream_in_frame_size(stream) /
             in_get_sample_rate(&stream->common) - elapsed_time_since_last_read;
     if (sleep_time > 0) {
         usleep(sleep_time);
     } else {
         sleep_time = 0;
     }
     in->last_read_time_us = now + sleep_time;
     // last_read_time_us is an approximation of when the (simulated) alsa
     // buffer is drained by the read, and is empty.
     //
     // On the subsequent in_read(), we measure the elapsed time spent in
     // the recording thread. This is subtracted from the sleep estimate based on frames,
     // thereby accounting for fill in the alsa buffer during the interim.
     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 size_t samples_per_milliseconds(size_t milliseconds,
                                        uint32_t sample_rate,
                                        size_t channel_count)
 {
     return milliseconds * sample_rate * channel_count / 1000;
 }

 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...");

     *stream_out = NULL;
     struct stub_stream_out *out =
             (struct stub_stream_out *)calloc(1, sizeof(struct stub_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->sample_rate = config->sample_rate;
     if (out->sample_rate == 0)
         out->sample_rate = STUB_DEFAULT_SAMPLE_RATE;
     out->channel_mask = config->channel_mask;
     if (out->channel_mask == AUDIO_CHANNEL_NONE)
         out->channel_mask = STUB_OUTPUT_DEFAULT_CHANNEL_MASK;
     out->format = config->format;
     if (out->format == AUDIO_FORMAT_DEFAULT)
         out->format = STUB_DEFAULT_AUDIO_FORMAT;
     out->frame_count = samples_per_milliseconds(
                            STUB_OUTPUT_BUFFER_MILLISECONDS,
                            out->sample_rate, 1);

     ALOGV("adev_open_output_stream: sample_rate: %u, channels: %x, format: %d,"
           " frames: %zu", out->sample_rate, out->channel_mask, out->format,
           out->frame_count);
     *stream_out = &out->stream;
     return 0;
 }

 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)
 {
     size_t buffer_size = samples_per_milliseconds(
                              STUB_INPUT_BUFFER_MILLISECONDS,
                              config->sample_rate,
                              audio_channel_count_from_in_mask(
                                  config->channel_mask));

     if (!audio_has_proportional_frames(config->format)) {
         // Since the audio data is not proportional choose an arbitrary size for
         // the buffer.
         buffer_size *= 4;
     } else {
         buffer_size *= audio_bytes_per_sample(config->format);
     }
     ALOGV("adev_get_input_buffer_size: %zu", buffer_size);
     return buffer_size;
 }

 static int adev_open_input_stream(struct audio_hw_device *dev,
                                   audio_io_handle_t handle,
                                   audio_devices_t devices,
                                   struct audio_config *config,
                                   struct audio_stream_in **stream_in,
                                   audio_input_flags_t flags __unused,
                                   const char *address __unused,
                                   audio_source_t source __unused)
 {
     ALOGV("adev_open_input_stream...");

     *stream_in = NULL;
     struct stub_stream_in *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;
     in->sample_rate = config->sample_rate;
     if (in->sample_rate == 0)
         in->sample_rate = STUB_DEFAULT_SAMPLE_RATE;
     in->channel_mask = config->channel_mask;
     if (in->channel_mask == AUDIO_CHANNEL_NONE)
         in->channel_mask = STUB_INPUT_DEFAULT_CHANNEL_MASK;
     in->format = config->format;
     if (in->format == AUDIO_FORMAT_DEFAULT)
         in->format = STUB_DEFAULT_AUDIO_FORMAT;
     in->frame_count = samples_per_milliseconds(
                           STUB_INPUT_BUFFER_MILLISECONDS, in->sample_rate, 1);

     ALOGV("adev_open_input_stream: sample_rate: %u, channels: %x, format: %d,"
           "frames: %zu", in->sample_rate, in->channel_mask, in->format,
           in->frame_count);
     *stream_in = &in->stream;
     return 0;
 }

 static void adev_close_input_stream(struct audio_hw_device *dev,
                                    struct audio_stream_in *in)
 {
     ALOGV("adev_close_input_stream...");
     return;
 }

 static int adev_dump(const audio_hw_device_t *device, int fd)
 {
     ALOGV("adev_dump");
     return 0;
 }

 static int adev_close(hw_device_t *device)
 {
     ALOGV("adev_close");
     free(device);
     return 0;
 }

 static int adev_open(const hw_module_t* module, const char* name,
                      hw_device_t** device)
 {
     ALOGV("adev_open: %s", name);

     struct stub_audio_device *adev;

     if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
         return -EINVAL;

     adev = calloc(1, sizeof(struct stub_audio_device));
     if (!adev)
         return -ENOMEM;

     adev->device.common.tag = HARDWARE_DEVICE_TAG;
     adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
     adev->device.common.module = (struct hw_module_t *) module;
     adev->device.common.close = adev_close;

     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.get_master_volume = adev_get_master_volume;
     adev->device.set_master_mute = adev_set_master_mute;
     adev->device.get_master_mute = adev_get_master_mute;
     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;

     *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,
         .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
         .hal_api_version = HARDWARE_HAL_API_VERSION,
         .id = AUDIO_HARDWARE_MODULE_ID,
         .name = "Default audio HW HAL",
         .author = "The Android Open Source Project",
         .methods = &hal_module_methods,
     },
 };
	/*
	* 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_default"
	//#define LOG_NDEBUG 0

	#include <errno.h>
	#include <malloc.h>
	#include <pthread.h>
	#include <stdint.h>
	#include <stdlib.h>
	#include <string.h>
	#include <time.h>
	#include <unistd.h>

	#include <log/log.h>

	#include <hardware/audio.h>
	#include <hardware/hardware.h>
	#include <system/audio.h>

	#define STUB_DEFAULT_SAMPLE_RATE 48000
	#define STUB_DEFAULT_AUDIO_FORMAT AUDIO_FORMAT_PCM_16_BIT

	#define STUB_INPUT_BUFFER_MILLISECONDS 20
	#define STUB_INPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_IN_STEREO

	#define STUB_OUTPUT_BUFFER_MILLISECONDS 10
	#define STUB_OUTPUT_DEFAULT_CHANNEL_MASK AUDIO_CHANNEL_OUT_STEREO

	struct stub_audio_device {
	struct audio_hw_device device;
	};

	struct stub_stream_out {
	struct audio_stream_out stream;
	int64_t last_write_time_us;
	uint32_t sample_rate;
	audio_channel_mask_t channel_mask;
	audio_format_t format;
	size_t frame_count;
	};

	struct stub_stream_in {
	struct audio_stream_in stream;
	int64_t last_read_time_us;
	uint32_t sample_rate;
	audio_channel_mask_t channel_mask;
	audio_format_t format;
	size_t frame_count;
	};

	static uint32_t out_get_sample_rate(const struct audio_stream *stream)
	{
	const struct stub_stream_out out = (const struct stub_stream_out )stream;

	ALOGV("out_get_sample_rate: %u", out->sample_rate);
	return out->sample_rate;
	}

	static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
	{
	struct stub_stream_out out = (struct stub_stream_out )stream;

	ALOGV("out_set_sample_rate: %d", rate);
	out->sample_rate = rate;
	return 0;
	}

	static size_t out_get_buffer_size(const struct audio_stream *stream)
	{
	const struct stub_stream_out out = (const struct stub_stream_out )stream;
	size_t buffer_size = out->frame_count *
	audio_stream_out_frame_size(&out->stream);

	ALOGV("out_get_buffer_size: %zu", buffer_size);
	return buffer_size;
	}

	static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
	{
	const struct stub_stream_out out = (const struct stub_stream_out )stream;

	ALOGV("out_get_channels: %x", out->channel_mask);
	return out->channel_mask;
	}

	static audio_format_t out_get_format(const struct audio_stream *stream)
	{
	const struct stub_stream_out out = (const struct stub_stream_out )stream;

	ALOGV("out_get_format: %d", out->format);
	return out->format;
	}

	static int out_set_format(struct audio_stream *stream, audio_format_t format)
	{
	struct stub_stream_out out = (struct stub_stream_out )stream;

	ALOGV("out_set_format: %d", format);
	out->format = format;
	return 0;
	}

	static int out_standby(struct audio_stream *stream)
	{
	ALOGV("out_standby");
	// out->last_write_time_us = 0; unnecessary as a stale write time has same effect
	return 0;
	}

	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");
	return 0;
	}

	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");
	return STUB_OUTPUT_BUFFER_MILLISECONDS;
	}

	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)
	{
	ALOGV("out_write: bytes: %zu", bytes);

	/* XXX: fake timing for audio output */
	struct stub_stream_out out = (struct stub_stream_out )stream;
	struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
	clock_gettime(CLOCK_MONOTONIC, &t);
	const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;
	const int64_t elapsed_time_since_last_write = now - out->last_write_time_us;
	int64_t sleep_time = bytes * 1000000LL / audio_stream_out_frame_size(stream) /
	out_get_sample_rate(&stream->common) - elapsed_time_since_last_write;
	if (sleep_time > 0) {
	usleep(sleep_time);
	} else {
	// we don't sleep when we exit standby (this is typical for a real alsa buffer).
	sleep_time = 0;
	}
	out->last_write_time_us = now + sleep_time;
	// last_write_time_us is an approximation of when the (simulated) alsa
	// buffer is believed completely full. The usleep above waits for more space
	// in the buffer, but by the end of the sleep the buffer is considered
	// topped-off.
	//
	// On the subsequent out_write(), we measure the elapsed time spent in
	// the mixer. This is subtracted from the sleep estimate based on frames,
	// thereby accounting for drain in the alsa buffer during mixing.
	// This is a crude approximation; we don't handle underruns precisely.
	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_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)
	{
	const struct stub_stream_in in = (const struct stub_stream_in )stream;

	ALOGV("in_get_sample_rate: %u", in->sample_rate);
	return in->sample_rate;
	}

	static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
	{
	struct stub_stream_in in = (struct stub_stream_in )stream;

	ALOGV("in_set_sample_rate: %u", rate);
	in->sample_rate = rate;
	return 0;
	}

	static size_t in_get_buffer_size(const struct audio_stream *stream)
	{
	const struct stub_stream_in in = (const struct stub_stream_in )stream;
	size_t buffer_size = in->frame_count *
	audio_stream_in_frame_size(&in->stream);

	ALOGV("in_get_buffer_size: %zu", buffer_size);
	return buffer_size;
	}

	static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
	{
	const struct stub_stream_in in = (const struct stub_stream_in )stream;

	ALOGV("in_get_channels: %x", in->channel_mask);
	return in->channel_mask;
	}

	static audio_format_t in_get_format(const struct audio_stream *stream)
	{
	const struct stub_stream_in in = (const struct stub_stream_in )stream;

	ALOGV("in_get_format: %d", in->format);
	return in->format;
	}

	static int in_set_format(struct audio_stream *stream, audio_format_t format)
	{
	struct stub_stream_in in = (struct stub_stream_in )stream;

	ALOGV("in_set_format: %d", format);
	in->format = format;
	return 0;
	}

	static int in_standby(struct audio_stream *stream)
	{
	struct stub_stream_in in = (struct stub_stream_in )stream;
	in->last_read_time_us = 0;
	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 */
	struct stub_stream_in in = (struct stub_stream_in )stream;
	struct timespec t = { .tv_sec = 0, .tv_nsec = 0 };
	clock_gettime(CLOCK_MONOTONIC, &t);
	const int64_t now = (t.tv_sec * 1000000000LL + t.tv_nsec) / 1000;

	// we do a full sleep when exiting standby.
	const bool standby = in->last_read_time_us == 0;
	const int64_t elapsed_time_since_last_read = standby ?
	0 : now - in->last_read_time_us;
	int64_t sleep_time = bytes * 1000000LL / audio_stream_in_frame_size(stream) /
	in_get_sample_rate(&stream->common) - elapsed_time_since_last_read;
	if (sleep_time > 0) {
	usleep(sleep_time);
	} else {
	sleep_time = 0;
	}
	in->last_read_time_us = now + sleep_time;
	// last_read_time_us is an approximation of when the (simulated) alsa
	// buffer is drained by the read, and is empty.
	//
	// On the subsequent in_read(), we measure the elapsed time spent in
	// the recording thread. This is subtracted from the sleep estimate based on frames,
	// thereby accounting for fill in the alsa buffer during the interim.
	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 size_t samples_per_milliseconds(size_t milliseconds,
	uint32_t sample_rate,
	size_t channel_count)
	{
	return milliseconds * sample_rate * channel_count / 1000;
	}

	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...");

	*stream_out = NULL;
	struct stub_stream_out *out =
	(struct stub_stream_out *)calloc(1, sizeof(struct stub_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->sample_rate = config->sample_rate;
	if (out->sample_rate == 0)
	out->sample_rate = STUB_DEFAULT_SAMPLE_RATE;
	out->channel_mask = config->channel_mask;
	if (out->channel_mask == AUDIO_CHANNEL_NONE)
	out->channel_mask = STUB_OUTPUT_DEFAULT_CHANNEL_MASK;
	out->format = config->format;
	if (out->format == AUDIO_FORMAT_DEFAULT)
	out->format = STUB_DEFAULT_AUDIO_FORMAT;
	out->frame_count = samples_per_milliseconds(
	STUB_OUTPUT_BUFFER_MILLISECONDS,
	out->sample_rate, 1);

	ALOGV("adev_open_output_stream: sample_rate: %u, channels: %x, format: %d,"
	" frames: %zu", out->sample_rate, out->channel_mask, out->format,
	out->frame_count);
	*stream_out = &out->stream;
	return 0;
	}

	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)
	{
	size_t buffer_size = samples_per_milliseconds(
	STUB_INPUT_BUFFER_MILLISECONDS,
	config->sample_rate,
	audio_channel_count_from_in_mask(
	config->channel_mask));

	if (!audio_has_proportional_frames(config->format)) {
	// Since the audio data is not proportional choose an arbitrary size for
	// the buffer.
	buffer_size *= 4;
	} else {
	buffer_size *= audio_bytes_per_sample(config->format);
	}
	ALOGV("adev_get_input_buffer_size: %zu", buffer_size);
	return buffer_size;
	}

	static int adev_open_input_stream(struct audio_hw_device *dev,
	audio_io_handle_t handle,
	audio_devices_t devices,
	struct audio_config *config,
	struct audio_stream_in **stream_in,
	audio_input_flags_t flags __unused,
	const char *address __unused,
	audio_source_t source __unused)
	{
	ALOGV("adev_open_input_stream...");

	*stream_in = NULL;
	struct stub_stream_in 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;
	in->sample_rate = config->sample_rate;
	if (in->sample_rate == 0)
	in->sample_rate = STUB_DEFAULT_SAMPLE_RATE;
	in->channel_mask = config->channel_mask;
	if (in->channel_mask == AUDIO_CHANNEL_NONE)
	in->channel_mask = STUB_INPUT_DEFAULT_CHANNEL_MASK;
	in->format = config->format;
	if (in->format == AUDIO_FORMAT_DEFAULT)
	in->format = STUB_DEFAULT_AUDIO_FORMAT;
	in->frame_count = samples_per_milliseconds(
	STUB_INPUT_BUFFER_MILLISECONDS, in->sample_rate, 1);

	ALOGV("adev_open_input_stream: sample_rate: %u, channels: %x, format: %d,"
	"frames: %zu", in->sample_rate, in->channel_mask, in->format,
	in->frame_count);
	*stream_in = &in->stream;
	return 0;
	}

	static void adev_close_input_stream(struct audio_hw_device *dev,
	struct audio_stream_in *in)
	{
	ALOGV("adev_close_input_stream...");
	return;
	}

	static int adev_dump(const audio_hw_device_t *device, int fd)
	{
	ALOGV("adev_dump");
	return 0;
	}

	static int adev_close(hw_device_t *device)
	{
	ALOGV("adev_close");
	free(device);
	return 0;
	}

	static int adev_open(const hw_module_t* module, const char* name,
	hw_device_t** device)
	{
	ALOGV("adev_open: %s", name);

	struct stub_audio_device *adev;

	if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
	return -EINVAL;

	adev = calloc(1, sizeof(struct stub_audio_device));
	if (!adev)
	return -ENOMEM;

	adev->device.common.tag = HARDWARE_DEVICE_TAG;
	adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
	adev->device.common.module = (struct hw_module_t *) module;
	adev->device.common.close = adev_close;

	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.get_master_volume = adev_get_master_volume;
	adev->device.set_master_mute = adev_set_master_mute;
	adev->device.get_master_mute = adev_get_master_mute;
	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;

	*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,
	.module_api_version = AUDIO_MODULE_API_VERSION_0_1,
	.hal_api_version = HARDWARE_HAL_API_VERSION,
	.id = AUDIO_HARDWARE_MODULE_ID,
	.name = "Default audio HW HAL",
	.author = "The Android Open Source Project",
	.methods = &hal_module_methods,
	},
	};