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platform_hardware_libhardware/modules/usbaudio/audio_hw.c
Eric Laurent c654dc6c1f am 7e9abab1: am b338957d: Merge "usbaudio: Return 0 bytes on failure in capture" 
* commit '7e9abab1940091b4c4d7589160855870fdda437e':
  usbaudio: Return 0 bytes on failure in capture
2015-01-09 18:04:14 +00:00

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/*
* Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#define LOG_TAG "usb_audio_hw"
/*#define LOG_NDEBUG 0*/
#include <errno.h>
#include <inttypes.h>
#include <pthread.h>
#include <stdint.h>
#include <stdlib.h>
#include <sys/time.h>
#include <log/log.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <hardware/audio.h>
#include <hardware/audio_alsaops.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <tinyalsa/asoundlib.h>
#include <audio_utils/channels.h>
/* FOR TESTING:
* Set k_force_channels to force the number of channels to present to AudioFlinger.
* 0 disables (this is default: present the device channels to AudioFlinger).
* 2 forces to legacy stereo mode.
*
* Others values can be tried (up to 8).
* TODO: AudioFlinger cannot support more than 8 active output channels
* at this time, so limiting logic needs to be put here or communicated from above.
*/
static const unsigned k_force_channels = 0;
#include "alsa_device_profile.h"
#include "alsa_device_proxy.h"
#include "audio_logging.h"
#define DEFAULT_INPUT_BUFFER_SIZE_MS 20
struct audio_device {
struct audio_hw_device hw_device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
/* output */
alsa_device_profile out_profile;
/* input */
alsa_device_profile in_profile;
bool mic_muted;
bool standby;
};
struct stream_out {
struct audio_stream_out stream;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
bool standby;
struct audio_device *dev; /* hardware information - only using this for the lock */
alsa_device_profile * profile;
alsa_device_proxy proxy; /* state of the stream */
unsigned hal_channel_count; /* channel count exposed to AudioFlinger.
* This may differ from the device channel count when
* the device is not compatible with AudioFlinger
* capabilities, e.g. exposes too many channels or
* too few channels. */
void * conversion_buffer; /* any conversions are put into here
* they could come from here too if
* there was a previous conversion */
size_t conversion_buffer_size; /* in bytes */
};
struct stream_in {
struct audio_stream_in stream;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
bool standby;
struct audio_device *dev; /* hardware information - only using this for the lock */
alsa_device_profile * profile;
alsa_device_proxy proxy; /* state of the stream */
unsigned hal_channel_count; /* channel count exposed to AudioFlinger.
* This may differ from the device channel count when
* the device is not compatible with AudioFlinger
* capabilities, e.g. exposes too many channels or
* too few channels. */
/* We may need to read more data from the device in order to data reduce to 16bit, 4chan */
void * conversion_buffer; /* any conversions are put into here
* they could come from here too if
* there was a previous conversion */
size_t conversion_buffer_size; /* in bytes */
};
/*
* Data Conversions
*/
/*
* Convert a buffer of packed (3-byte) PCM24LE samples to PCM16LE samples.
* in_buff points to the buffer of PCM24LE samples
* num_in_samples size of input buffer in SAMPLES
* out_buff points to the buffer to receive converted PCM16LE LE samples.
* returns
* the number of BYTES of output data.
* We are doing this since we *always* present to The Framework as A PCM16LE device, but need to
* support PCM24_3LE (24-bit, packed).
* NOTE:
* This conversion is safe to do in-place (in_buff == out_buff).
* TODO Move this to a utilities module.
*/
static size_t convert_24_3_to_16(const unsigned char * in_buff, size_t num_in_samples,
short * out_buff)
{
/*
* Move from front to back so that the conversion can be done in-place
* i.e. in_buff == out_buff
*/
/* we need 2 bytes in the output for every 3 bytes in the input */
unsigned char* dst_ptr = (unsigned char*)out_buff;
const unsigned char* src_ptr = in_buff;
size_t src_smpl_index;
for (src_smpl_index = 0; src_smpl_index < num_in_samples; src_smpl_index++) {
src_ptr++; /* lowest-(skip)-byte */
*dst_ptr++ = *src_ptr++; /* low-byte */
*dst_ptr++ = *src_ptr++; /* high-byte */
}
/* return number of *bytes* generated: */
return num_in_samples * 2;
}
/*
* Convert a buffer of packed (3-byte) PCM32 samples to PCM16LE samples.
* in_buff points to the buffer of PCM32 samples
* num_in_samples size of input buffer in SAMPLES
* out_buff points to the buffer to receive converted PCM16LE LE samples.
* returns
* the number of BYTES of output data.
* We are doing this since we *always* present to The Framework as A PCM16LE device, but need to
* support PCM_FORMAT_S32_LE (32-bit).
* NOTE:
* This conversion is safe to do in-place (in_buff == out_buff).
* TODO Move this to a utilities module.
*/
static size_t convert_32_to_16(const int32_t * in_buff, size_t num_in_samples, short * out_buff)
{
/*
* Move from front to back so that the conversion can be done in-place
* i.e. in_buff == out_buff
*/
short * dst_ptr = out_buff;
const int32_t* src_ptr = in_buff;
size_t src_smpl_index;
for (src_smpl_index = 0; src_smpl_index < num_in_samples; src_smpl_index++) {
*dst_ptr++ = *src_ptr++ >> 16;
}
/* return number of *bytes* generated: */
return num_in_samples * 2;
}
static char * device_get_parameters(alsa_device_profile * profile, const char * keys)
{
ALOGV("usb:audio_hw::device_get_parameters() keys:%s", keys);
if (profile->card < 0 || profile->device < 0) {
return strdup("");
}
struct str_parms *query = str_parms_create_str(keys);
struct str_parms *result = str_parms_create();
/* These keys are from hardware/libhardware/include/audio.h */
/* supported sample rates */
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
char* rates_list = profile_get_sample_rate_strs(profile);
str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
rates_list);
free(rates_list);
}
/* supported channel counts */
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
char* channels_list = profile_get_channel_count_strs(profile);
str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_CHANNELS,
channels_list);
free(channels_list);
}
/* supported sample formats */
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
char * format_params = profile_get_format_strs(profile);
str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS,
format_params);
free(format_params);
}
str_parms_destroy(query);
char* result_str = str_parms_to_str(result);
str_parms_destroy(result);
ALOGV("usb:audio_hw::device_get_parameters = %s", result_str);
return result_str;
}
/*
* HAl Functions
*/
/**
* NOTE: when multiple mutexes have to be acquired, always respect the
* following order: hw device > out stream
*/
/*
* OUT functions
*/
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
uint32_t rate = proxy_get_sample_rate(&((struct stream_out*)stream)->proxy);
ALOGV("out_get_sample_rate() = %d", rate);
return rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return 0;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
const struct stream_out* out = (const struct stream_out*)stream;
size_t buffer_size =
proxy_get_period_size(&out->proxy) * audio_stream_out_frame_size(&(out->stream));
return buffer_size;
}
static uint32_t out_get_channels(const struct audio_stream *stream)
{
const struct stream_out *out = (const struct stream_out*)stream;
return audio_channel_out_mask_from_count(out->hal_channel_count);
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
/* Note: The HAL doesn't do any FORMAT conversion at this time. It
* Relies on the framework to provide data in the specified format.
* This could change in the future.
*/
alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
return format;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
return 0;
}
static int out_standby(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
if (!out->standby) {
proxy_close(&out->proxy);
out->standby = true;
}
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
return 0;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
ALOGV("usb:audio_hw::out out_set_parameters() keys:%s", kvpairs);
struct stream_out *out = (struct stream_out *)stream;
char value[32];
int param_val;
int routing = 0;
int ret_value = 0;
int card = -1;
int device = -1;
struct str_parms * parms = str_parms_create_str(kvpairs);
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
param_val = str_parms_get_str(parms, "card", value, sizeof(value));
if (param_val >= 0)
card = atoi(value);
param_val = str_parms_get_str(parms, "device", value, sizeof(value));
if (param_val >= 0)
device = atoi(value);
if (card >= 0 && device >= 0 && !profile_is_cached_for(out->profile, card, device)) {
/* cannot read pcm device info if playback is active */
if (!out->standby)
ret_value = -ENOSYS;
else {
int saved_card = out->profile->card;
int saved_device = out->profile->device;
out->profile->card = card;
out->profile->device = device;
ret_value = profile_read_device_info(out->profile) ? 0 : -EINVAL;
if (ret_value != 0) {
out->profile->card = saved_card;
out->profile->device = saved_device;
}
}
}
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
str_parms_destroy(parms);
return ret_value;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
struct stream_out *out = (struct stream_out *)stream;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
char * params_str = device_get_parameters(out->profile, keys);
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
return params_str;
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
return proxy_get_latency(proxy);
}
static int out_set_volume(struct audio_stream_out *stream, float left, float right)
{
return -ENOSYS;
}
/* must be called with hw device and output stream mutexes locked */
static int start_output_stream(struct stream_out *out)
{
ALOGV("usb:audio_hw::out start_output_stream(card:%d device:%d)",
out->profile->card, out->profile->device);
return proxy_open(&out->proxy);
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
{
int ret;
struct stream_out *out = (struct stream_out *)stream;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
ret = start_output_stream(out);
if (ret != 0) {
pthread_mutex_unlock(&out->dev->lock);
goto err;
}
out->standby = false;
}
pthread_mutex_unlock(&out->dev->lock);
alsa_device_proxy* proxy = &out->proxy;
const void * write_buff = buffer;
int num_write_buff_bytes = bytes;
const int num_device_channels = proxy_get_channel_count(proxy); /* what we told alsa */
const int num_req_channels = out->hal_channel_count; /* what we told AudioFlinger */
if (num_device_channels != num_req_channels) {
/* allocate buffer */
const size_t required_conversion_buffer_size =
bytes * num_device_channels / num_req_channels;
if (required_conversion_buffer_size > out->conversion_buffer_size) {
out->conversion_buffer_size = required_conversion_buffer_size;
out->conversion_buffer = realloc(out->conversion_buffer,
out->conversion_buffer_size);
}
/* convert data */
const audio_format_t audio_format = out_get_format(&(out->stream.common));
const unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
num_write_buff_bytes =
adjust_channels(write_buff, num_req_channels,
out->conversion_buffer, num_device_channels,
sample_size_in_bytes, num_write_buff_bytes);
write_buff = out->conversion_buffer;
}
if (write_buff != NULL && num_write_buff_bytes != 0) {
proxy_write(&out->proxy, write_buff, num_write_buff_bytes);
}
pthread_mutex_unlock(&out->lock);
return bytes;
err:
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common));
}
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream, uint32_t *dsp_frames)
{
return -EINVAL;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
/* FIXME - This needs to be implemented */
return -EINVAL;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream, int64_t *timestamp)
{
return -EINVAL;
}
static int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address __unused)
{
ALOGV("usb:audio_hw::out adev_open_output_stream() handle:0x%X, device:0x%X, flags:0x%X",
handle, devices, flags);
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
if (!out)
return -ENOMEM;
/* setup function pointers */
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
out->stream.common.get_channels = out_get_channels;
out->stream.common.get_format = out_get_format;
out->stream.common.set_format = out_set_format;
out->stream.common.standby = out_standby;
out->stream.common.dump = out_dump;
out->stream.common.set_parameters = out_set_parameters;
out->stream.common.get_parameters = out_get_parameters;
out->stream.common.add_audio_effect = out_add_audio_effect;
out->stream.common.remove_audio_effect = out_remove_audio_effect;
out->stream.get_latency = out_get_latency;
out->stream.set_volume = out_set_volume;
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
out->stream.get_presentation_position = out_get_presentation_position;
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->dev = adev;
out->profile = &adev->out_profile;
// build this to hand to the alsa_device_proxy
struct pcm_config proxy_config;
memset(&proxy_config, 0, sizeof(proxy_config));
int ret = 0;
/* Rate */
if (config->sample_rate == 0) {
proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
} else if (profile_is_sample_rate_valid(out->profile, config->sample_rate)) {
proxy_config.rate = config->sample_rate;
} else {
proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
ret = -EINVAL;
}
/* Format */
if (config->format == AUDIO_FORMAT_DEFAULT) {
proxy_config.format = profile_get_default_format(out->profile);
config->format = audio_format_from_pcm_format(proxy_config.format);
} else {
enum pcm_format fmt = pcm_format_from_audio_format(config->format);
if (profile_is_format_valid(out->profile, fmt)) {
proxy_config.format = fmt;
} else {
proxy_config.format = profile_get_default_format(out->profile);
config->format = audio_format_from_pcm_format(proxy_config.format);
ret = -EINVAL;
}
}
/* Channels */
unsigned proposed_channel_count = profile_get_default_channel_count(out->profile);
if (k_force_channels) {
proposed_channel_count = k_force_channels;
} else if (config->channel_mask != AUDIO_CHANNEL_NONE) {
proposed_channel_count = audio_channel_count_from_out_mask(config->channel_mask);
}
/* we can expose any channel count mask, and emulate internally. */
config->channel_mask = audio_channel_out_mask_from_count(proposed_channel_count);
out->hal_channel_count = proposed_channel_count;
/* no validity checks are needed as proxy_prepare() forces channel_count to be valid.
* and we emulate any channel count discrepancies in out_write(). */
proxy_config.channels = proposed_channel_count;
proxy_prepare(&out->proxy, out->profile, &proxy_config);
/* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
ret = 0;
out->conversion_buffer = NULL;
out->conversion_buffer_size = 0;
out->standby = true;
*stream_out = &out->stream;
return ret;
err_open:
free(out);
*stream_out = NULL;
return -ENOSYS;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
ALOGV("usb:audio_hw::out adev_close_output_stream()");
struct stream_out *out = (struct stream_out *)stream;
/* Close the pcm device */
out_standby(&stream->common);
free(out->conversion_buffer);
out->conversion_buffer = NULL;
out->conversion_buffer_size = 0;
free(stream);
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
/* TODO This needs to be calculated based on format/channels/rate */
return 320;
}
/*
* IN functions
*/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
uint32_t rate = proxy_get_sample_rate(&((const struct stream_in *)stream)->proxy);
ALOGV("in_get_sample_rate() = %d", rate);
return rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("in_set_sample_rate(%d) - NOPE", rate);
return -ENOSYS;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
const struct stream_in * in = ((const struct stream_in*)stream);
return proxy_get_period_size(&in->proxy) * audio_stream_in_frame_size(&(in->stream));
}
static uint32_t in_get_channels(const struct audio_stream *stream)
{
const struct stream_in *in = (const struct stream_in*)stream;
return audio_channel_in_mask_from_count(in->hal_channel_count);
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
/* TODO Here is the code we need when we support arbitrary input formats
* alsa_device_proxy * proxy = ((struct stream_in*)stream)->proxy;
* audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
* ALOGV("in_get_format() = %d", format);
* return format;
*/
/* Input only supports PCM16 */
/* TODO When AudioPolicyManager & AudioFlinger supports arbitrary input formats
rewrite this to return the ACTUAL channel format (above) */
return AUDIO_FORMAT_PCM_16_BIT;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
ALOGV("in_set_format(%d) - NOPE", format);
return -ENOSYS;
}
static int in_standby(struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
if (!in->standby) {
proxy_close(&in->proxy);
in->standby = true;
}
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&in->dev->lock);
return 0;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
ALOGV("usb: audio_hw::in in_set_parameters() keys:%s", kvpairs);
struct stream_in *in = (struct stream_in *)stream;
char value[32];
int param_val;
int routing = 0;
int ret_value = 0;
int card = -1;
int device = -1;
struct str_parms * parms = str_parms_create_str(kvpairs);
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
/* Device Connection Message ("card=1,device=0") */
param_val = str_parms_get_str(parms, "card", value, sizeof(value));
if (param_val >= 0)
card = atoi(value);
param_val = str_parms_get_str(parms, "device", value, sizeof(value));
if (param_val >= 0)
device = atoi(value);
if (card >= 0 && device >= 0 && !profile_is_cached_for(in->profile, card, device)) {
/* cannot read pcm device info if playback is active */
if (!in->standby)
ret_value = -ENOSYS;
else {
int saved_card = in->profile->card;
int saved_device = in->profile->device;
in->profile->card = card;
in->profile->device = device;
ret_value = profile_read_device_info(in->profile) ? 0 : -EINVAL;
if (ret_value != 0) {
in->profile->card = saved_card;
in->profile->device = saved_device;
}
}
}
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&in->dev->lock);
str_parms_destroy(parms);
return ret_value;
}
static char * in_get_parameters(const struct audio_stream *stream, const char *keys)
{
struct stream_in *in = (struct stream_in *)stream;
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
char * params_str = device_get_parameters(in->profile, keys);
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&in->dev->lock);
return params_str;
}
static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
return 0;
}
/* must be called with hw device and output stream mutexes locked */
static int start_input_stream(struct stream_in *in)
{
ALOGV("usb:audio_hw::start_input_stream(card:%d device:%d)",
in->profile->card, in->profile->device);
return proxy_open(&in->proxy);
}
/* TODO mutex stuff here (see out_write) */
static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
{
size_t num_read_buff_bytes = 0;
void * read_buff = buffer;
void * out_buff = buffer;
struct stream_in * in = (struct stream_in *)stream;
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
if (in->standby) {
if (start_input_stream(in) != 0) {
pthread_mutex_unlock(&in->dev->lock);
goto err;
}
in->standby = false;
}
pthread_mutex_unlock(&in->dev->lock);
alsa_device_profile * profile = in->profile;
/*
* OK, we need to figure out how much data to read to be able to output the requested
* number of bytes in the HAL format (16-bit, stereo).
*/
num_read_buff_bytes = bytes;
int num_device_channels = proxy_get_channel_count(&in->proxy);
int num_req_channels = in->hal_channel_count;
if (num_device_channels != num_req_channels) {
num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels;
}
enum pcm_format format = proxy_get_format(&in->proxy);
if (format == PCM_FORMAT_S24_3LE) {
/* 24-bit USB device */
num_read_buff_bytes = (3 * num_read_buff_bytes) / 2;
} else if (format == PCM_FORMAT_S32_LE) {
/* 32-bit USB device */
num_read_buff_bytes = num_read_buff_bytes * 2;
}
/* Setup/Realloc the conversion buffer (if necessary). */
if (num_read_buff_bytes != bytes) {
if (num_read_buff_bytes > in->conversion_buffer_size) {
/*TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
(and do these conversions themselves) */
in->conversion_buffer_size = num_read_buff_bytes;
in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
}
read_buff = in->conversion_buffer;
}
if (proxy_read(&in->proxy, read_buff, num_read_buff_bytes) == 0) {
/*
* Do any conversions necessary to send the data in the format specified to/by the HAL
* (but different from the ALSA format), such as 24bit ->16bit, or 4chan -> 2chan.
*/
if (format != PCM_FORMAT_S16_LE) {
/* we need to convert */
if (num_device_channels != num_req_channels) {
out_buff = read_buff;
}
if (format == PCM_FORMAT_S24_3LE) {
num_read_buff_bytes =
convert_24_3_to_16(read_buff, num_read_buff_bytes / 3, out_buff);
} else if (format == PCM_FORMAT_S32_LE) {
num_read_buff_bytes =
convert_32_to_16(read_buff, num_read_buff_bytes / 4, out_buff);
} else {
LOG_ALWAYS_FATAL("Unsupported format");
num_read_buff_bytes = 0;
goto err;
}
}
if (num_device_channels != num_req_channels) {
// ALOGV("chans dev:%d req:%d", num_device_channels, num_req_channels);
out_buff = buffer;
/* Num Channels conversion */
if (num_device_channels != num_req_channels) {
audio_format_t audio_format = in_get_format(&(in->stream.common));
unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
num_read_buff_bytes =
adjust_channels(read_buff, num_device_channels,
out_buff, num_req_channels,
sample_size_in_bytes, num_read_buff_bytes);
}
}
/* no need to acquire in->dev->lock to read mic_muted here as we don't change its state */
if (num_read_buff_bytes > 0 && in->dev->mic_muted)
memset(buffer, 0, num_read_buff_bytes);
} else {
num_read_buff_bytes = 0;
}
err:
pthread_mutex_unlock(&in->lock);
return num_read_buff_bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
return 0;
}
static int adev_open_input_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in,
audio_input_flags_t flags __unused,
const char *address __unused,
audio_source_t source __unused)
{
ALOGV("usb: in adev_open_input_stream() rate:%" PRIu32 ", chanMask:0x%" PRIX32 ", fmt:%" PRIu8,
config->sample_rate, config->channel_mask, config->format);
struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
int ret = 0;
if (in == NULL)
return -ENOMEM;
/* setup function pointers */
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
in->stream.common.get_buffer_size = in_get_buffer_size;
in->stream.common.get_channels = in_get_channels;
in->stream.common.get_format = in_get_format;
in->stream.common.set_format = in_set_format;
in->stream.common.standby = in_standby;
in->stream.common.dump = in_dump;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
in->stream.common.add_audio_effect = in_add_audio_effect;
in->stream.common.remove_audio_effect = in_remove_audio_effect;
in->stream.set_gain = in_set_gain;
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
in->dev = (struct audio_device *)dev;
in->profile = &in->dev->in_profile;
struct pcm_config proxy_config;
memset(&proxy_config, 0, sizeof(proxy_config));
/* Rate */
if (config->sample_rate == 0) {
proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
} else if (profile_is_sample_rate_valid(in->profile, config->sample_rate)) {
proxy_config.rate = config->sample_rate;
} else {
proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
ret = -EINVAL;
}
/* Format */
/* until the framework supports format conversion, just take what it asks for
* i.e. AUDIO_FORMAT_PCM_16_BIT */
if (config->format == AUDIO_FORMAT_DEFAULT) {
/* just return AUDIO_FORMAT_PCM_16_BIT until the framework supports other input
* formats */
config->format = AUDIO_FORMAT_PCM_16_BIT;
proxy_config.format = PCM_FORMAT_S16_LE;
} else if (config->format == AUDIO_FORMAT_PCM_16_BIT) {
/* Always accept AUDIO_FORMAT_PCM_16_BIT until the framework supports other input
* formats */
proxy_config.format = PCM_FORMAT_S16_LE;
} else {
/* When the framework support other formats, validate here */
config->format = AUDIO_FORMAT_PCM_16_BIT;
proxy_config.format = PCM_FORMAT_S16_LE;
ret = -EINVAL;
}
/* Channels */
unsigned proposed_channel_count = profile_get_default_channel_count(in->profile);
if (k_force_channels) {
proposed_channel_count = k_force_channels;
} else if (config->channel_mask != AUDIO_CHANNEL_NONE) {
proposed_channel_count = audio_channel_count_from_in_mask(config->channel_mask);
}
/* we can expose any channel count mask, and emulate internally. */
config->channel_mask = audio_channel_in_mask_from_count(proposed_channel_count);
in->hal_channel_count = proposed_channel_count;
proxy_config.channels = profile_get_default_channel_count(in->profile);
proxy_prepare(&in->proxy, in->profile, &proxy_config);
in->standby = true;
in->conversion_buffer = NULL;
in->conversion_buffer_size = 0;
*stream_in = &in->stream;
return ret;
}
static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream)
{
struct stream_in *in = (struct stream_in *)stream;
/* Close the pcm device */
in_standby(&stream->common);
free(in->conversion_buffer);
free(stream);
}
/*
* ADEV Functions
*/
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
ALOGV("audio_hw:usb adev_set_parameters(%s)", kvpairs);
struct audio_device * adev = (struct audio_device *)dev;
char value[32];
int param_val;
struct str_parms * parms = str_parms_create_str(kvpairs);
/* Check for the "disconnect" message */
param_val = str_parms_get_str(parms, "disconnect", value, sizeof(value));
if (param_val >= 0) {
audio_devices_t device = (audio_devices_t)atoi(value);
param_val = str_parms_get_str(parms, "card", value, sizeof(value));
int alsa_card = param_val >= 0 ? atoi(value) : -1;
param_val = str_parms_get_str(parms, "device", value, sizeof(value));
int alsa_device = param_val >= 0 ? atoi(value) : -1;
if (alsa_card >= 0 && alsa_device >= 0) {
/* "decache" the profile */
pthread_mutex_lock(&adev->lock);
if (device == AUDIO_DEVICE_OUT_USB_DEVICE &&
profile_is_cached_for(&adev->out_profile, alsa_card, alsa_device)) {
profile_decache(&adev->out_profile);
}
if (device == AUDIO_DEVICE_IN_USB_DEVICE &&
profile_is_cached_for(&adev->in_profile, alsa_card, alsa_device)) {
profile_decache(&adev->in_profile);
}
pthread_mutex_unlock(&adev->lock);
}
}
str_parms_destroy(parms);
return 0;
}
static char * adev_get_parameters(const struct audio_hw_device *dev, const char *keys)
{
return strdup("");
}
static int adev_init_check(const struct audio_hw_device *dev)
{
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
return -ENOSYS;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
struct audio_device * adev = (struct audio_device *)dev;
pthread_mutex_lock(&adev->lock);
adev->mic_muted = state;
pthread_mutex_unlock(&adev->lock);
return -ENOSYS;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
return -ENOSYS;
}
static int adev_dump(const audio_hw_device_t *device, int fd)
{
return 0;
}
static int adev_close(hw_device_t *device)
{
struct audio_device *adev = (struct audio_device *)device;
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device)
{
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
struct audio_device *adev = calloc(1, sizeof(struct audio_device));
if (!adev)
return -ENOMEM;
profile_init(&adev->out_profile, PCM_OUT);
profile_init(&adev->in_profile, PCM_IN);
adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->hw_device.common.module = (struct hw_module_t *)module;
adev->hw_device.common.close = adev_close;
adev->hw_device.init_check = adev_init_check;
adev->hw_device.set_voice_volume = adev_set_voice_volume;
adev->hw_device.set_master_volume = adev_set_master_volume;
adev->hw_device.set_mode = adev_set_mode;
adev->hw_device.set_mic_mute = adev_set_mic_mute;
adev->hw_device.get_mic_mute = adev_get_mic_mute;
adev->hw_device.set_parameters = adev_set_parameters;
adev->hw_device.get_parameters = adev_get_parameters;
adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
adev->hw_device.open_output_stream = adev_open_output_stream;
adev->hw_device.close_output_stream = adev_close_output_stream;
adev->hw_device.open_input_stream = adev_open_input_stream;
adev->hw_device.close_input_stream = adev_close_input_stream;
adev->hw_device.dump = adev_dump;
*device = &adev->hw_device.common;
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 = "USB audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};
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