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platform_hardware_libhardware/modules/audio_remote_submix/audio_hw.cpp
Eric Laurent 854a10af10 Add debug folder for audioserver 
Use /data/misc/audioserver instead of data/misc/media for
audioserver debug.

Bug: 27064332
Change-Id: Ic213ee0354d9ab1ed1980e8c3d07cd239597ad2e
2016-02-24 00:27:46 +00:00

1782 lines
72 KiB
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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 "r_submix"
//#define LOG_NDEBUG 0
#include <errno.h>
#include <pthread.h>
#include <stdint.h>
#include <stdlib.h>
#include <sys/param.h>
#include <sys/time.h>
#include <sys/limits.h>
#include <cutils/compiler.h>
#include <cutils/log.h>
#include <cutils/properties.h>
#include <cutils/str_parms.h>
#include <hardware/audio.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <media/AudioParameter.h>
#include <media/AudioBufferProvider.h>
#include <media/nbaio/MonoPipe.h>
#include <media/nbaio/MonoPipeReader.h>
#include <utils/String8.h>
#define LOG_STREAMS_TO_FILES 0
#if LOG_STREAMS_TO_FILES
#include <fcntl.h>
#include <stdio.h>
#include <sys/stat.h>
#endif // LOG_STREAMS_TO_FILES
extern "C" {
namespace android {
// Set to 1 to enable extremely verbose logging in this module.
#define SUBMIX_VERBOSE_LOGGING 0
#if SUBMIX_VERBOSE_LOGGING
#define SUBMIX_ALOGV(...) ALOGV(__VA_ARGS__)
#define SUBMIX_ALOGE(...) ALOGE(__VA_ARGS__)
#else
#define SUBMIX_ALOGV(...)
#define SUBMIX_ALOGE(...)
#endif // SUBMIX_VERBOSE_LOGGING
// NOTE: This value will be rounded up to the nearest power of 2 by MonoPipe().
#define DEFAULT_PIPE_SIZE_IN_FRAMES (1024*4)
// Value used to divide the MonoPipe() buffer into segments that are written to the source and
// read from the sink. The maximum latency of the device is the size of the MonoPipe's buffer
// the minimum latency is the MonoPipe buffer size divided by this value.
#define DEFAULT_PIPE_PERIOD_COUNT 4
// The duration of MAX_READ_ATTEMPTS * READ_ATTEMPT_SLEEP_MS must be stricly inferior to
// the duration of a record buffer at the current record sample rate (of the device, not of
// the recording itself). Here we have:
// 3 * 5ms = 15ms < 1024 frames * 1000 / 48000 = 21.333ms
#define MAX_READ_ATTEMPTS 3
#define READ_ATTEMPT_SLEEP_MS 5 // 5ms between two read attempts when pipe is empty
#define DEFAULT_SAMPLE_RATE_HZ 48000 // default sample rate
// See NBAIO_Format frameworks/av/include/media/nbaio/NBAIO.h.
#define DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT
// A legacy user of this device does not close the input stream when it shuts down, which
// results in the application opening a new input stream before closing the old input stream
// handle it was previously using. Setting this value to 1 allows multiple clients to open
// multiple input streams from this device. If this option is enabled, each input stream returned
// is *the same stream* which means that readers will race to read data from these streams.
#define ENABLE_LEGACY_INPUT_OPEN 1
// Whether channel conversion (16-bit signed PCM mono->stereo, stereo->mono) is enabled.
#define ENABLE_CHANNEL_CONVERSION 1
// Whether resampling is enabled.
#define ENABLE_RESAMPLING 1
#if LOG_STREAMS_TO_FILES
// Folder to save stream log files to.
#define LOG_STREAM_FOLDER "/data/misc/audioserver"
// Log filenames for input and output streams.
#define LOG_STREAM_OUT_FILENAME LOG_STREAM_FOLDER "/r_submix_out.raw"
#define LOG_STREAM_IN_FILENAME LOG_STREAM_FOLDER "/r_submix_in.raw"
// File permissions for stream log files.
#define LOG_STREAM_FILE_PERMISSIONS (S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH)
#endif // LOG_STREAMS_TO_FILES
// limit for number of read error log entries to avoid spamming the logs
#define MAX_READ_ERROR_LOGS 5
// Common limits macros.
#ifndef min
#define min(a, b) ((a) < (b) ? (a) : (b))
#endif // min
#ifndef max
#define max(a, b) ((a) > (b) ? (a) : (b))
#endif // max
// Set *result_variable_ptr to true if value_to_find is present in the array array_to_search,
// otherwise set *result_variable_ptr to false.
#define SUBMIX_VALUE_IN_SET(value_to_find, array_to_search, result_variable_ptr) \
{ \
size_t i; \
*(result_variable_ptr) = false; \
for (i = 0; i < sizeof(array_to_search) / sizeof((array_to_search)[0]); i++) { \
if ((value_to_find) == (array_to_search)[i]) { \
*(result_variable_ptr) = true; \
break; \
} \
} \
}
// Configuration of the submix pipe.
struct submix_config {
// Channel mask field in this data structure is set to either input_channel_mask or
// output_channel_mask depending upon the last stream to be opened on this device.
struct audio_config common;
// Input stream and output stream channel masks. This is required since input and output
// channel bitfields are not equivalent.
audio_channel_mask_t input_channel_mask;
audio_channel_mask_t output_channel_mask;
#if ENABLE_RESAMPLING
// Input stream and output stream sample rates.
uint32_t input_sample_rate;
uint32_t output_sample_rate;
#endif // ENABLE_RESAMPLING
size_t pipe_frame_size; // Number of bytes in each audio frame in the pipe.
size_t buffer_size_frames; // Size of the audio pipe in frames.
// Maximum number of frames buffered by the input and output streams.
size_t buffer_period_size_frames;
};
#define MAX_ROUTES 10
typedef struct route_config {
struct submix_config config;
char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
// Pipe variables: they handle the ring buffer that "pipes" audio:
// - from the submix virtual audio output == what needs to be played
// remotely, seen as an output for AudioFlinger
// - to the virtual audio source == what is captured by the component
// which "records" the submix / virtual audio source, and handles it as needed.
// A usecase example is one where the component capturing the audio is then sending it over
// Wifi for presentation on a remote Wifi Display device (e.g. a dongle attached to a TV, or a
// TV with Wifi Display capabilities), or to a wireless audio player.
sp<MonoPipe> rsxSink;
sp<MonoPipeReader> rsxSource;
// Pointers to the current input and output stream instances. rsxSink and rsxSource are
// destroyed if both and input and output streams are destroyed.
struct submix_stream_out *output;
struct submix_stream_in *input;
#if ENABLE_RESAMPLING
// Buffer used as temporary storage for resampled data prior to returning data to the output
// stream.
int16_t resampler_buffer[DEFAULT_PIPE_SIZE_IN_FRAMES];
#endif // ENABLE_RESAMPLING
} route_config_t;
struct submix_audio_device {
struct audio_hw_device device;
route_config_t routes[MAX_ROUTES];
// Device lock, also used to protect access to submix_audio_device from the input and output
// streams.
pthread_mutex_t lock;
};
struct submix_stream_out {
struct audio_stream_out stream;
struct submix_audio_device *dev;
int route_handle;
bool output_standby;
uint64_t frames_written;
uint64_t frames_written_since_standby;
#if LOG_STREAMS_TO_FILES
int log_fd;
#endif // LOG_STREAMS_TO_FILES
};
struct submix_stream_in {
struct audio_stream_in stream;
struct submix_audio_device *dev;
int route_handle;
bool input_standby;
bool output_standby_rec_thr; // output standby state as seen from record thread
// wall clock when recording starts
struct timespec record_start_time;
// how many frames have been requested to be read
uint64_t read_counter_frames;
#if ENABLE_LEGACY_INPUT_OPEN
// Number of references to this input stream.
volatile int32_t ref_count;
#endif // ENABLE_LEGACY_INPUT_OPEN
#if LOG_STREAMS_TO_FILES
int log_fd;
#endif // LOG_STREAMS_TO_FILES
volatile int16_t read_error_count;
};
// Determine whether the specified sample rate is supported by the submix module.
static bool sample_rate_supported(const uint32_t sample_rate)
{
// Set of sample rates supported by Format_from_SR_C() frameworks/av/media/libnbaio/NAIO.cpp.
static const unsigned int supported_sample_rates[] = {
8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
};
bool return_value;
SUBMIX_VALUE_IN_SET(sample_rate, supported_sample_rates, &return_value);
return return_value;
}
// Determine whether the specified sample rate is supported, if it is return the specified sample
// rate, otherwise return the default sample rate for the submix module.
static uint32_t get_supported_sample_rate(uint32_t sample_rate)
{
return sample_rate_supported(sample_rate) ? sample_rate : DEFAULT_SAMPLE_RATE_HZ;
}
// Determine whether the specified channel in mask is supported by the submix module.
static bool channel_in_mask_supported(const audio_channel_mask_t channel_in_mask)
{
// Set of channel in masks supported by Format_from_SR_C()
// frameworks/av/media/libnbaio/NAIO.cpp.
static const audio_channel_mask_t supported_channel_in_masks[] = {
AUDIO_CHANNEL_IN_MONO, AUDIO_CHANNEL_IN_STEREO,
};
bool return_value;
SUBMIX_VALUE_IN_SET(channel_in_mask, supported_channel_in_masks, &return_value);
return return_value;
}
// Determine whether the specified channel in mask is supported, if it is return the specified
// channel in mask, otherwise return the default channel in mask for the submix module.
static audio_channel_mask_t get_supported_channel_in_mask(
const audio_channel_mask_t channel_in_mask)
{
return channel_in_mask_supported(channel_in_mask) ? channel_in_mask :
static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_IN_STEREO);
}
// Determine whether the specified channel out mask is supported by the submix module.
static bool channel_out_mask_supported(const audio_channel_mask_t channel_out_mask)
{
// Set of channel out masks supported by Format_from_SR_C()
// frameworks/av/media/libnbaio/NAIO.cpp.
static const audio_channel_mask_t supported_channel_out_masks[] = {
AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO,
};
bool return_value;
SUBMIX_VALUE_IN_SET(channel_out_mask, supported_channel_out_masks, &return_value);
return return_value;
}
// Determine whether the specified channel out mask is supported, if it is return the specified
// channel out mask, otherwise return the default channel out mask for the submix module.
static audio_channel_mask_t get_supported_channel_out_mask(
const audio_channel_mask_t channel_out_mask)
{
return channel_out_mask_supported(channel_out_mask) ? channel_out_mask :
static_cast<audio_channel_mask_t>(AUDIO_CHANNEL_OUT_STEREO);
}
// Get a pointer to submix_stream_out given an audio_stream_out that is embedded within the
// structure.
static struct submix_stream_out * audio_stream_out_get_submix_stream_out(
struct audio_stream_out * const stream)
{
ALOG_ASSERT(stream);
return reinterpret_cast<struct submix_stream_out *>(reinterpret_cast<uint8_t *>(stream) -
offsetof(struct submix_stream_out, stream));
}
// Get a pointer to submix_stream_out given an audio_stream that is embedded within the structure.
static struct submix_stream_out * audio_stream_get_submix_stream_out(
struct audio_stream * const stream)
{
ALOG_ASSERT(stream);
return audio_stream_out_get_submix_stream_out(
reinterpret_cast<struct audio_stream_out *>(stream));
}
// Get a pointer to submix_stream_in given an audio_stream_in that is embedded within the
// structure.
static struct submix_stream_in * audio_stream_in_get_submix_stream_in(
struct audio_stream_in * const stream)
{
ALOG_ASSERT(stream);
return reinterpret_cast<struct submix_stream_in *>(reinterpret_cast<uint8_t *>(stream) -
offsetof(struct submix_stream_in, stream));
}
// Get a pointer to submix_stream_in given an audio_stream that is embedded within the structure.
static struct submix_stream_in * audio_stream_get_submix_stream_in(
struct audio_stream * const stream)
{
ALOG_ASSERT(stream);
return audio_stream_in_get_submix_stream_in(
reinterpret_cast<struct audio_stream_in *>(stream));
}
// Get a pointer to submix_audio_device given a pointer to an audio_device that is embedded within
// the structure.
static struct submix_audio_device * audio_hw_device_get_submix_audio_device(
struct audio_hw_device *device)
{
ALOG_ASSERT(device);
return reinterpret_cast<struct submix_audio_device *>(reinterpret_cast<uint8_t *>(device) -
offsetof(struct submix_audio_device, device));
}
// Compare an audio_config with input channel mask and an audio_config with output channel mask
// returning false if they do *not* match, true otherwise.
static bool audio_config_compare(const audio_config * const input_config,
const audio_config * const output_config)
{
#if !ENABLE_CHANNEL_CONVERSION
const uint32_t input_channels = audio_channel_count_from_in_mask(input_config->channel_mask);
const uint32_t output_channels = audio_channel_count_from_out_mask(output_config->channel_mask);
if (input_channels != output_channels) {
ALOGE("audio_config_compare() channel count mismatch input=%d vs. output=%d",
input_channels, output_channels);
return false;
}
#endif // !ENABLE_CHANNEL_CONVERSION
#if ENABLE_RESAMPLING
if (input_config->sample_rate != output_config->sample_rate &&
audio_channel_count_from_in_mask(input_config->channel_mask) != 1) {
#else
if (input_config->sample_rate != output_config->sample_rate) {
#endif // ENABLE_RESAMPLING
ALOGE("audio_config_compare() sample rate mismatch %ul vs. %ul",
input_config->sample_rate, output_config->sample_rate);
return false;
}
if (input_config->format != output_config->format) {
ALOGE("audio_config_compare() format mismatch %x vs. %x",
input_config->format, output_config->format);
return false;
}
// This purposely ignores offload_info as it's not required for the submix device.
return true;
}
// If one doesn't exist, create a pipe for the submix audio device rsxadev of size
// buffer_size_frames and optionally associate "in" or "out" with the submix audio device.
// Must be called with lock held on the submix_audio_device
static void submix_audio_device_create_pipe_l(struct submix_audio_device * const rsxadev,
const struct audio_config * const config,
const size_t buffer_size_frames,
const uint32_t buffer_period_count,
struct submix_stream_in * const in,
struct submix_stream_out * const out,
const char *address,
int route_idx)
{
ALOG_ASSERT(in || out);
ALOG_ASSERT(route_idx > -1);
ALOG_ASSERT(route_idx < MAX_ROUTES);
ALOGD("submix_audio_device_create_pipe_l(addr=%s, idx=%d)", address, route_idx);
// Save a reference to the specified input or output stream and the associated channel
// mask.
if (in) {
in->route_handle = route_idx;
rsxadev->routes[route_idx].input = in;
rsxadev->routes[route_idx].config.input_channel_mask = config->channel_mask;
#if ENABLE_RESAMPLING
rsxadev->routes[route_idx].config.input_sample_rate = config->sample_rate;
// If the output isn't configured yet, set the output sample rate to the maximum supported
// sample rate such that the smallest possible input buffer is created, and put a default
// value for channel count
if (!rsxadev->routes[route_idx].output) {
rsxadev->routes[route_idx].config.output_sample_rate = 48000;
rsxadev->routes[route_idx].config.output_channel_mask = AUDIO_CHANNEL_OUT_STEREO;
}
#endif // ENABLE_RESAMPLING
}
if (out) {
out->route_handle = route_idx;
rsxadev->routes[route_idx].output = out;
rsxadev->routes[route_idx].config.output_channel_mask = config->channel_mask;
#if ENABLE_RESAMPLING
rsxadev->routes[route_idx].config.output_sample_rate = config->sample_rate;
#endif // ENABLE_RESAMPLING
}
// Save the address
strncpy(rsxadev->routes[route_idx].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN);
ALOGD(" now using address %s for route %d", rsxadev->routes[route_idx].address, route_idx);
// If a pipe isn't associated with the device, create one.
if (rsxadev->routes[route_idx].rsxSink == NULL || rsxadev->routes[route_idx].rsxSource == NULL)
{
struct submix_config * const device_config = &rsxadev->routes[route_idx].config;
uint32_t channel_count;
if (out)
channel_count = audio_channel_count_from_out_mask(config->channel_mask);
else
channel_count = audio_channel_count_from_in_mask(config->channel_mask);
#if ENABLE_CHANNEL_CONVERSION
// If channel conversion is enabled, allocate enough space for the maximum number of
// possible channels stored in the pipe for the situation when the number of channels in
// the output stream don't match the number in the input stream.
const uint32_t pipe_channel_count = max(channel_count, 2);
#else
const uint32_t pipe_channel_count = channel_count;
#endif // ENABLE_CHANNEL_CONVERSION
const NBAIO_Format format = Format_from_SR_C(config->sample_rate, pipe_channel_count,
config->format);
const NBAIO_Format offers[1] = {format};
size_t numCounterOffers = 0;
// Create a MonoPipe with optional blocking set to true.
MonoPipe* sink = new MonoPipe(buffer_size_frames, format, true /*writeCanBlock*/);
// Negotiation between the source and sink cannot fail as the device open operation
// creates both ends of the pipe using the same audio format.
ssize_t index = sink->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
MonoPipeReader* source = new MonoPipeReader(sink);
numCounterOffers = 0;
index = source->negotiate(offers, 1, NULL, numCounterOffers);
ALOG_ASSERT(index == 0);
ALOGV("submix_audio_device_create_pipe_l(): created pipe");
// Save references to the source and sink.
ALOG_ASSERT(rsxadev->routes[route_idx].rsxSink == NULL);
ALOG_ASSERT(rsxadev->routes[route_idx].rsxSource == NULL);
rsxadev->routes[route_idx].rsxSink = sink;
rsxadev->routes[route_idx].rsxSource = source;
// Store the sanitized audio format in the device so that it's possible to determine
// the format of the pipe source when opening the input device.
memcpy(&device_config->common, config, sizeof(device_config->common));
device_config->buffer_size_frames = sink->maxFrames();
device_config->buffer_period_size_frames = device_config->buffer_size_frames /
buffer_period_count;
if (in) device_config->pipe_frame_size = audio_stream_in_frame_size(&in->stream);
if (out) device_config->pipe_frame_size = audio_stream_out_frame_size(&out->stream);
#if ENABLE_CHANNEL_CONVERSION
// Calculate the pipe frame size based upon the number of channels.
device_config->pipe_frame_size = (device_config->pipe_frame_size * pipe_channel_count) /
channel_count;
#endif // ENABLE_CHANNEL_CONVERSION
SUBMIX_ALOGV("submix_audio_device_create_pipe_l(): pipe frame size %zd, pipe size %zd, "
"period size %zd", device_config->pipe_frame_size,
device_config->buffer_size_frames, device_config->buffer_period_size_frames);
}
}
// Release references to the sink and source. Input and output threads may maintain references
// to these objects via StrongPointer (sp<MonoPipe> and sp<MonoPipeReader>) which they can use
// before they shutdown.
// Must be called with lock held on the submix_audio_device
static void submix_audio_device_release_pipe_l(struct submix_audio_device * const rsxadev,
int route_idx)
{
ALOG_ASSERT(route_idx > -1);
ALOG_ASSERT(route_idx < MAX_ROUTES);
ALOGD("submix_audio_device_release_pipe_l(idx=%d) addr=%s", route_idx,
rsxadev->routes[route_idx].address);
if (rsxadev->routes[route_idx].rsxSink != 0) {
rsxadev->routes[route_idx].rsxSink.clear();
rsxadev->routes[route_idx].rsxSink = 0;
}
if (rsxadev->routes[route_idx].rsxSource != 0) {
rsxadev->routes[route_idx].rsxSource.clear();
rsxadev->routes[route_idx].rsxSource = 0;
}
memset(rsxadev->routes[route_idx].address, 0, AUDIO_DEVICE_MAX_ADDRESS_LEN);
#ifdef ENABLE_RESAMPLING
memset(rsxadev->routes[route_idx].resampler_buffer, 0,
sizeof(int16_t) * DEFAULT_PIPE_SIZE_IN_FRAMES);
#endif
}
// Remove references to the specified input and output streams. When the device no longer
// references input and output streams destroy the associated pipe.
// Must be called with lock held on the submix_audio_device
static void submix_audio_device_destroy_pipe_l(struct submix_audio_device * const rsxadev,
const struct submix_stream_in * const in,
const struct submix_stream_out * const out)
{
MonoPipe* sink;
ALOGV("submix_audio_device_destroy_pipe_l()");
int route_idx = -1;
if (in != NULL) {
#if ENABLE_LEGACY_INPUT_OPEN
const_cast<struct submix_stream_in*>(in)->ref_count--;
route_idx = in->route_handle;
ALOG_ASSERT(rsxadev->routes[route_idx].input == in);
if (in->ref_count == 0) {
rsxadev->routes[route_idx].input = NULL;
}
ALOGV("submix_audio_device_destroy_pipe_l(): input ref_count %d", in->ref_count);
#else
rsxadev->input = NULL;
#endif // ENABLE_LEGACY_INPUT_OPEN
}
if (out != NULL) {
route_idx = out->route_handle;
ALOG_ASSERT(rsxadev->routes[route_idx].output == out);
rsxadev->routes[route_idx].output = NULL;
}
if (route_idx != -1 &&
rsxadev->routes[route_idx].input == NULL && rsxadev->routes[route_idx].output == NULL) {
submix_audio_device_release_pipe_l(rsxadev, route_idx);
ALOGD("submix_audio_device_destroy_pipe_l(): pipe destroyed");
}
}
// Sanitize the user specified audio config for a submix input / output stream.
static void submix_sanitize_config(struct audio_config * const config, const bool is_input_format)
{
config->channel_mask = is_input_format ? get_supported_channel_in_mask(config->channel_mask) :
get_supported_channel_out_mask(config->channel_mask);
config->sample_rate = get_supported_sample_rate(config->sample_rate);
config->format = DEFAULT_FORMAT;
}
// Verify a submix input or output stream can be opened.
// Must be called with lock held on the submix_audio_device
static bool submix_open_validate_l(const struct submix_audio_device * const rsxadev,
int route_idx,
const struct audio_config * const config,
const bool opening_input)
{
bool input_open;
bool output_open;
audio_config pipe_config;
// Query the device for the current audio config and whether input and output streams are open.
output_open = rsxadev->routes[route_idx].output != NULL;
input_open = rsxadev->routes[route_idx].input != NULL;
memcpy(&pipe_config, &rsxadev->routes[route_idx].config.common, sizeof(pipe_config));
// If the stream is already open, don't open it again.
if (opening_input ? !ENABLE_LEGACY_INPUT_OPEN && input_open : output_open) {
ALOGE("submix_open_validate_l(): %s stream already open.", opening_input ? "Input" :
"Output");
return false;
}
SUBMIX_ALOGV("submix_open_validate_l(): sample rate=%d format=%x "
"%s_channel_mask=%x", config->sample_rate, config->format,
opening_input ? "in" : "out", config->channel_mask);
// If either stream is open, verify the existing audio config the pipe matches the user
// specified config.
if (input_open || output_open) {
const audio_config * const input_config = opening_input ? config : &pipe_config;
const audio_config * const output_config = opening_input ? &pipe_config : config;
// Get the channel mask of the open device.
pipe_config.channel_mask =
opening_input ? rsxadev->routes[route_idx].config.output_channel_mask :
rsxadev->routes[route_idx].config.input_channel_mask;
if (!audio_config_compare(input_config, output_config)) {
ALOGE("submix_open_validate_l(): Unsupported format.");
return false;
}
}
return true;
}
// Must be called with lock held on the submix_audio_device
static status_t submix_get_route_idx_for_address_l(const struct submix_audio_device * const rsxadev,
const char* address, /*in*/
int *idx /*out*/)
{
// Do we already have a route for this address
int route_idx = -1;
int route_empty_idx = -1; // index of an empty route slot that can be used if needed
for (int i=0 ; i < MAX_ROUTES ; i++) {
if (strcmp(rsxadev->routes[i].address, "") == 0) {
route_empty_idx = i;
}
if (strncmp(rsxadev->routes[i].address, address, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
route_idx = i;
break;
}
}
if ((route_idx == -1) && (route_empty_idx == -1)) {
ALOGE("Cannot create new route for address %s, max number of routes reached", address);
return -ENOMEM;
}
if (route_idx == -1) {
route_idx = route_empty_idx;
}
*idx = route_idx;
return OK;
}
// Calculate the maximum size of the pipe buffer in frames for the specified stream.
static size_t calculate_stream_pipe_size_in_frames(const struct audio_stream *stream,
const struct submix_config *config,
const size_t pipe_frames,
const size_t stream_frame_size)
{
const size_t pipe_frame_size = config->pipe_frame_size;
const size_t max_frame_size = max(stream_frame_size, pipe_frame_size);
return (pipe_frames * config->pipe_frame_size) / max_frame_size;
}
/* audio HAL functions */
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
const_cast<struct audio_stream *>(stream));
#if ENABLE_RESAMPLING
const uint32_t out_rate = out->dev->routes[out->route_handle].config.output_sample_rate;
#else
const uint32_t out_rate = out->dev->routes[out->route_handle].config.common.sample_rate;
#endif // ENABLE_RESAMPLING
SUBMIX_ALOGV("out_get_sample_rate() returns %u for addr %s",
out_rate, out->dev->routes[out->route_handle].address);
return out_rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
#if ENABLE_RESAMPLING
// The sample rate of the stream can't be changed once it's set since this would change the
// output buffer size and hence break playback to the shared pipe.
if (rate != out->dev->routes[out->route_handle].config.output_sample_rate) {
ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from "
"%u to %u for addr %s",
out->dev->routes[out->route_handle].config.output_sample_rate, rate,
out->dev->routes[out->route_handle].address);
return -ENOSYS;
}
#endif // ENABLE_RESAMPLING
if (!sample_rate_supported(rate)) {
ALOGE("out_set_sample_rate(rate=%u) rate unsupported", rate);
return -ENOSYS;
}
SUBMIX_ALOGV("out_set_sample_rate(rate=%u)", rate);
out->dev->routes[out->route_handle].config.common.sample_rate = rate;
return 0;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
const_cast<struct audio_stream *>(stream));
const struct submix_config * const config = &out->dev->routes[out->route_handle].config;
const size_t stream_frame_size =
audio_stream_out_frame_size((const struct audio_stream_out *)stream);
const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
stream, config, config->buffer_period_size_frames, stream_frame_size);
const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
SUBMIX_ALOGV("out_get_buffer_size() returns %zu bytes, %zu frames",
buffer_size_bytes, buffer_size_frames);
return buffer_size_bytes;
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
const_cast<struct audio_stream *>(stream));
uint32_t channel_mask = out->dev->routes[out->route_handle].config.output_channel_mask;
SUBMIX_ALOGV("out_get_channels() returns %08x", channel_mask);
return channel_mask;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(
const_cast<struct audio_stream *>(stream));
const audio_format_t format = out->dev->routes[out->route_handle].config.common.format;
SUBMIX_ALOGV("out_get_format() returns %x", format);
return format;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
const struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
if (format != out->dev->routes[out->route_handle].config.common.format) {
ALOGE("out_set_format(format=%x) format unsupported", format);
return -ENOSYS;
}
SUBMIX_ALOGV("out_set_format(format=%x)", format);
return 0;
}
static int out_standby(struct audio_stream *stream)
{
ALOGI("out_standby()");
struct submix_stream_out * const out = audio_stream_get_submix_stream_out(stream);
struct submix_audio_device * const rsxadev = out->dev;
pthread_mutex_lock(&rsxadev->lock);
out->output_standby = true;
out->frames_written_since_standby = 0;
pthread_mutex_unlock(&rsxadev->lock);
return 0;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
(void)stream;
(void)fd;
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
int exiting = -1;
AudioParameter parms = AudioParameter(String8(kvpairs));
SUBMIX_ALOGV("out_set_parameters() kvpairs='%s'", kvpairs);
// FIXME this is using hard-coded strings but in the future, this functionality will be
// converted to use audio HAL extensions required to support tunneling
if ((parms.getInt(String8("exiting"), exiting) == NO_ERROR) && (exiting > 0)) {
struct submix_audio_device * const rsxadev =
audio_stream_get_submix_stream_out(stream)->dev;
pthread_mutex_lock(&rsxadev->lock);
{ // using the sink
sp<MonoPipe> sink =
rsxadev->routes[audio_stream_get_submix_stream_out(stream)->route_handle]
.rsxSink;
if (sink == NULL) {
pthread_mutex_unlock(&rsxadev->lock);
return 0;
}
ALOGD("out_set_parameters(): shutting down MonoPipe sink");
sink->shutdown(true);
} // done using the sink
pthread_mutex_unlock(&rsxadev->lock);
}
return 0;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
(void)stream;
(void)keys;
return strdup("");
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
const struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(
const_cast<struct audio_stream_out *>(stream));
const struct submix_config * const config = &out->dev->routes[out->route_handle].config;
const size_t stream_frame_size =
audio_stream_out_frame_size(stream);
const size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
&stream->common, config, config->buffer_size_frames, stream_frame_size);
const uint32_t sample_rate = out_get_sample_rate(&stream->common);
const uint32_t latency_ms = (buffer_size_frames * 1000) / sample_rate;
SUBMIX_ALOGV("out_get_latency() returns %u ms, size in frames %zu, sample rate %u",
latency_ms, buffer_size_frames, sample_rate);
return latency_ms;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
(void)stream;
(void)left;
(void)right;
return -ENOSYS;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
SUBMIX_ALOGV("out_write(bytes=%zd)", bytes);
ssize_t written_frames = 0;
const size_t frame_size = audio_stream_out_frame_size(stream);
struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream);
struct submix_audio_device * const rsxadev = out->dev;
const size_t frames = bytes / frame_size;
pthread_mutex_lock(&rsxadev->lock);
out->output_standby = false;
sp<MonoPipe> sink = rsxadev->routes[out->route_handle].rsxSink;
if (sink != NULL) {
if (sink->isShutdown()) {
sink.clear();
pthread_mutex_unlock(&rsxadev->lock);
SUBMIX_ALOGV("out_write(): pipe shutdown, ignoring the write.");
// the pipe has already been shutdown, this buffer will be lost but we must
// simulate timing so we don't drain the output faster than realtime
usleep(frames * 1000000 / out_get_sample_rate(&stream->common));
return bytes;
}
} else {
pthread_mutex_unlock(&rsxadev->lock);
ALOGE("out_write without a pipe!");
ALOG_ASSERT("out_write without a pipe!");
return 0;
}
// If the write to the sink would block when no input stream is present, flush enough frames
// from the pipe to make space to write the most recent data.
{
const size_t availableToWrite = sink->availableToWrite();
sp<MonoPipeReader> source = rsxadev->routes[out->route_handle].rsxSource;
if (rsxadev->routes[out->route_handle].input == NULL && availableToWrite < frames) {
static uint8_t flush_buffer[64];
const size_t flushBufferSizeFrames = sizeof(flush_buffer) / frame_size;
size_t frames_to_flush_from_source = frames - availableToWrite;
SUBMIX_ALOGV("out_write(): flushing %d frames from the pipe to avoid blocking",
frames_to_flush_from_source);
while (frames_to_flush_from_source) {
const size_t flush_size = min(frames_to_flush_from_source, flushBufferSizeFrames);
frames_to_flush_from_source -= flush_size;
// read does not block
source->read(flush_buffer, flush_size);
}
}
}
pthread_mutex_unlock(&rsxadev->lock);
written_frames = sink->write(buffer, frames);
#if LOG_STREAMS_TO_FILES
if (out->log_fd >= 0) write(out->log_fd, buffer, written_frames * frame_size);
#endif // LOG_STREAMS_TO_FILES
if (written_frames < 0) {
if (written_frames == (ssize_t)NEGOTIATE) {
ALOGE("out_write() write to pipe returned NEGOTIATE");
pthread_mutex_lock(&rsxadev->lock);
sink.clear();
pthread_mutex_unlock(&rsxadev->lock);
written_frames = 0;
return 0;
} else {
// write() returned UNDERRUN or WOULD_BLOCK, retry
ALOGE("out_write() write to pipe returned unexpected %zd", written_frames);
written_frames = sink->write(buffer, frames);
}
}
pthread_mutex_lock(&rsxadev->lock);
sink.clear();
if (written_frames > 0) {
out->frames_written_since_standby += written_frames;
out->frames_written += written_frames;
}
pthread_mutex_unlock(&rsxadev->lock);
if (written_frames < 0) {
ALOGE("out_write() failed writing to pipe with %zd", written_frames);
return 0;
}
const ssize_t written_bytes = written_frames * frame_size;
SUBMIX_ALOGV("out_write() wrote %zd bytes %zd frames", written_bytes, written_frames);
return written_bytes;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
if (stream == NULL || frames == NULL || timestamp == NULL) {
return -EINVAL;
}
const submix_stream_out *out = audio_stream_out_get_submix_stream_out(
const_cast<struct audio_stream_out *>(stream));
struct submix_audio_device * const rsxadev = out->dev;
int ret = -EWOULDBLOCK;
pthread_mutex_lock(&rsxadev->lock);
const ssize_t frames_in_pipe =
rsxadev->routes[out->route_handle].rsxSource->availableToRead();
if (CC_UNLIKELY(frames_in_pipe < 0)) {
*frames = out->frames_written;
ret = 0;
} else if (out->frames_written >= (uint64_t)frames_in_pipe) {
*frames = out->frames_written - frames_in_pipe;
ret = 0;
}
pthread_mutex_unlock(&rsxadev->lock);
if (ret == 0) {
clock_gettime(CLOCK_MONOTONIC, timestamp);
}
SUBMIX_ALOGV("out_get_presentation_position() got frames=%llu timestamp sec=%llu",
frames ? *frames : -1, timestamp ? timestamp->tv_sec : -1);
return ret;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
if (stream == NULL || dsp_frames == NULL) {
return -EINVAL;
}
const submix_stream_out *out = audio_stream_out_get_submix_stream_out(
const_cast<struct audio_stream_out *>(stream));
struct submix_audio_device * const rsxadev = out->dev;
pthread_mutex_lock(&rsxadev->lock);
const ssize_t frames_in_pipe =
rsxadev->routes[out->route_handle].rsxSource->availableToRead();
if (CC_UNLIKELY(frames_in_pipe < 0)) {
*dsp_frames = (uint32_t)out->frames_written_since_standby;
} else {
*dsp_frames = out->frames_written_since_standby > (uint64_t) frames_in_pipe ?
(uint32_t)(out->frames_written_since_standby - frames_in_pipe) : 0;
}
pthread_mutex_unlock(&rsxadev->lock);
return 0;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
(void)stream;
(void)effect;
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
(void)stream;
(void)effect;
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
int64_t *timestamp)
{
(void)stream;
(void)timestamp;
return -EINVAL;
}
/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
const_cast<struct audio_stream*>(stream));
#if ENABLE_RESAMPLING
const uint32_t rate = in->dev->routes[in->route_handle].config.input_sample_rate;
#else
const uint32_t rate = in->dev->routes[in->route_handle].config.common.sample_rate;
#endif // ENABLE_RESAMPLING
SUBMIX_ALOGV("in_get_sample_rate() returns %u", rate);
return rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
#if ENABLE_RESAMPLING
// The sample rate of the stream can't be changed once it's set since this would change the
// input buffer size and hence break recording from the shared pipe.
if (rate != in->dev->routes[in->route_handle].config.input_sample_rate) {
ALOGE("in_set_sample_rate() resampling enabled can't change sample rate from "
"%u to %u", in->dev->routes[in->route_handle].config.input_sample_rate, rate);
return -ENOSYS;
}
#endif // ENABLE_RESAMPLING
if (!sample_rate_supported(rate)) {
ALOGE("in_set_sample_rate(rate=%u) rate unsupported", rate);
return -ENOSYS;
}
in->dev->routes[in->route_handle].config.common.sample_rate = rate;
SUBMIX_ALOGV("in_set_sample_rate() set %u", rate);
return 0;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
const_cast<struct audio_stream*>(stream));
const struct submix_config * const config = &in->dev->routes[in->route_handle].config;
const size_t stream_frame_size =
audio_stream_in_frame_size((const struct audio_stream_in *)stream);
size_t buffer_size_frames = calculate_stream_pipe_size_in_frames(
stream, config, config->buffer_period_size_frames, stream_frame_size);
#if ENABLE_RESAMPLING
// Scale the size of the buffer based upon the maximum number of frames that could be returned
// given the ratio of output to input sample rate.
buffer_size_frames = (size_t)(((float)buffer_size_frames *
(float)config->input_sample_rate) /
(float)config->output_sample_rate);
#endif // ENABLE_RESAMPLING
const size_t buffer_size_bytes = buffer_size_frames * stream_frame_size;
SUBMIX_ALOGV("in_get_buffer_size() returns %zu bytes, %zu frames", buffer_size_bytes,
buffer_size_frames);
return buffer_size_bytes;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
const_cast<struct audio_stream*>(stream));
const audio_channel_mask_t channel_mask =
in->dev->routes[in->route_handle].config.input_channel_mask;
SUBMIX_ALOGV("in_get_channels() returns %x", channel_mask);
return channel_mask;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(
const_cast<struct audio_stream*>(stream));
const audio_format_t format = in->dev->routes[in->route_handle].config.common.format;
SUBMIX_ALOGV("in_get_format() returns %x", format);
return format;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
const struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
if (format != in->dev->routes[in->route_handle].config.common.format) {
ALOGE("in_set_format(format=%x) format unsupported", format);
return -ENOSYS;
}
SUBMIX_ALOGV("in_set_format(format=%x)", format);
return 0;
}
static int in_standby(struct audio_stream *stream)
{
ALOGI("in_standby()");
struct submix_stream_in * const in = audio_stream_get_submix_stream_in(stream);
struct submix_audio_device * const rsxadev = in->dev;
pthread_mutex_lock(&rsxadev->lock);
in->input_standby = true;
pthread_mutex_unlock(&rsxadev->lock);
return 0;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
(void)stream;
(void)fd;
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
(void)stream;
(void)kvpairs;
return 0;
}
static char * in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
(void)stream;
(void)keys;
return strdup("");
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
(void)stream;
(void)gain;
return 0;
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream);
struct submix_audio_device * const rsxadev = in->dev;
struct audio_config *format;
const size_t frame_size = audio_stream_in_frame_size(stream);
const size_t frames_to_read = bytes / frame_size;
SUBMIX_ALOGV("in_read bytes=%zu", bytes);
pthread_mutex_lock(&rsxadev->lock);
const bool output_standby = rsxadev->routes[in->route_handle].output == NULL
? true : rsxadev->routes[in->route_handle].output->output_standby;
const bool output_standby_transition = (in->output_standby_rec_thr != output_standby);
in->output_standby_rec_thr = output_standby;
if (in->input_standby || output_standby_transition) {
in->input_standby = false;
// keep track of when we exit input standby (== first read == start "real recording")
// or when we start recording silence, and reset projected time
int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time);
if (rc == 0) {
in->read_counter_frames = 0;
}
}
in->read_counter_frames += frames_to_read;
size_t remaining_frames = frames_to_read;
{
// about to read from audio source
sp<MonoPipeReader> source = rsxadev->routes[in->route_handle].rsxSource;
if (source == NULL) {
in->read_error_count++;// ok if it rolls over
ALOGE_IF(in->read_error_count < MAX_READ_ERROR_LOGS,
"no audio pipe yet we're trying to read! (not all errors will be logged)");
pthread_mutex_unlock(&rsxadev->lock);
usleep(frames_to_read * 1000000 / in_get_sample_rate(&stream->common));
memset(buffer, 0, bytes);
return bytes;
}
pthread_mutex_unlock(&rsxadev->lock);
// read the data from the pipe (it's non blocking)
int attempts = 0;
char* buff = (char*)buffer;
#if ENABLE_CHANNEL_CONVERSION
// Determine whether channel conversion is required.
const uint32_t input_channels = audio_channel_count_from_in_mask(
rsxadev->routes[in->route_handle].config.input_channel_mask);
const uint32_t output_channels = audio_channel_count_from_out_mask(
rsxadev->routes[in->route_handle].config.output_channel_mask);
if (input_channels != output_channels) {
SUBMIX_ALOGV("in_read(): %d output channels will be converted to %d "
"input channels", output_channels, input_channels);
// Only support 16-bit PCM channel conversion from mono to stereo or stereo to mono.
ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
AUDIO_FORMAT_PCM_16_BIT);
ALOG_ASSERT((input_channels == 1 && output_channels == 2) ||
(input_channels == 2 && output_channels == 1));
}
#endif // ENABLE_CHANNEL_CONVERSION
#if ENABLE_RESAMPLING
const uint32_t input_sample_rate = in_get_sample_rate(&stream->common);
const uint32_t output_sample_rate =
rsxadev->routes[in->route_handle].config.output_sample_rate;
const size_t resampler_buffer_size_frames =
sizeof(rsxadev->routes[in->route_handle].resampler_buffer) /
sizeof(rsxadev->routes[in->route_handle].resampler_buffer[0]);
float resampler_ratio = 1.0f;
// Determine whether resampling is required.
if (input_sample_rate != output_sample_rate) {
resampler_ratio = (float)output_sample_rate / (float)input_sample_rate;
// Only support 16-bit PCM mono resampling.
// NOTE: Resampling is performed after the channel conversion step.
ALOG_ASSERT(rsxadev->routes[in->route_handle].config.common.format ==
AUDIO_FORMAT_PCM_16_BIT);
ALOG_ASSERT(audio_channel_count_from_in_mask(
rsxadev->routes[in->route_handle].config.input_channel_mask) == 1);
}
#endif // ENABLE_RESAMPLING
while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
ssize_t frames_read = -1977;
size_t read_frames = remaining_frames;
#if ENABLE_RESAMPLING
char* const saved_buff = buff;
if (resampler_ratio != 1.0f) {
// Calculate the number of frames from the pipe that need to be read to generate
// the data for the input stream read.
const size_t frames_required_for_resampler = (size_t)(
(float)read_frames * (float)resampler_ratio);
read_frames = min(frames_required_for_resampler, resampler_buffer_size_frames);
// Read into the resampler buffer.
buff = (char*)rsxadev->routes[in->route_handle].resampler_buffer;
}
#endif // ENABLE_RESAMPLING
#if ENABLE_CHANNEL_CONVERSION
if (output_channels == 1 && input_channels == 2) {
// Need to read half the requested frames since the converted output
// data will take twice the space (mono->stereo).
read_frames /= 2;
}
#endif // ENABLE_CHANNEL_CONVERSION
SUBMIX_ALOGV("in_read(): frames available to read %zd", source->availableToRead());
frames_read = source->read(buff, read_frames);
SUBMIX_ALOGV("in_read(): frames read %zd", frames_read);
#if ENABLE_CHANNEL_CONVERSION
// Perform in-place channel conversion.
// NOTE: In the following "input stream" refers to the data returned by this function
// and "output stream" refers to the data read from the pipe.
if (input_channels != output_channels && frames_read > 0) {
int16_t *data = (int16_t*)buff;
if (output_channels == 2 && input_channels == 1) {
// Offset into the output stream data in samples.
ssize_t output_stream_offset = 0;
for (ssize_t input_stream_frame = 0; input_stream_frame < frames_read;
input_stream_frame++, output_stream_offset += 2) {
// Average the content from both channels.
data[input_stream_frame] = ((int32_t)data[output_stream_offset] +
(int32_t)data[output_stream_offset + 1]) / 2;
}
} else if (output_channels == 1 && input_channels == 2) {
// Offset into the input stream data in samples.
ssize_t input_stream_offset = (frames_read - 1) * 2;
for (ssize_t output_stream_frame = frames_read - 1; output_stream_frame >= 0;
output_stream_frame--, input_stream_offset -= 2) {
const short sample = data[output_stream_frame];
data[input_stream_offset] = sample;
data[input_stream_offset + 1] = sample;
}
}
}
#endif // ENABLE_CHANNEL_CONVERSION
#if ENABLE_RESAMPLING
if (resampler_ratio != 1.0f) {
SUBMIX_ALOGV("in_read(): resampling %zd frames", frames_read);
const int16_t * const data = (int16_t*)buff;
int16_t * const resampled_buffer = (int16_t*)saved_buff;
// Resample with *no* filtering - if the data from the ouptut stream was really
// sampled at a different rate this will result in very nasty aliasing.
const float output_stream_frames = (float)frames_read;
size_t input_stream_frame = 0;
for (float output_stream_frame = 0.0f;
output_stream_frame < output_stream_frames &&
input_stream_frame < remaining_frames;
output_stream_frame += resampler_ratio, input_stream_frame++) {
resampled_buffer[input_stream_frame] = data[(size_t)output_stream_frame];
}
ALOG_ASSERT(input_stream_frame <= (ssize_t)resampler_buffer_size_frames);
SUBMIX_ALOGV("in_read(): resampler produced %zd frames", input_stream_frame);
frames_read = input_stream_frame;
buff = saved_buff;
}
#endif // ENABLE_RESAMPLING
if (frames_read > 0) {
#if LOG_STREAMS_TO_FILES
if (in->log_fd >= 0) write(in->log_fd, buff, frames_read * frame_size);
#endif // LOG_STREAMS_TO_FILES
remaining_frames -= frames_read;
buff += frames_read * frame_size;
SUBMIX_ALOGV(" in_read (att=%d) got %zd frames, remaining=%zu",
attempts, frames_read, remaining_frames);
} else {
attempts++;
SUBMIX_ALOGE(" in_read read returned %zd", frames_read);
usleep(READ_ATTEMPT_SLEEP_MS * 1000);
}
}
// done using the source
pthread_mutex_lock(&rsxadev->lock);
source.clear();
pthread_mutex_unlock(&rsxadev->lock);
}
if (remaining_frames > 0) {
const size_t remaining_bytes = remaining_frames * frame_size;
SUBMIX_ALOGV(" clearing remaining_frames = %zu", remaining_frames);
memset(((char*)buffer)+ bytes - remaining_bytes, 0, remaining_bytes);
}
// compute how much we need to sleep after reading the data by comparing the wall clock with
// the projected time at which we should return.
struct timespec time_after_read;// wall clock after reading from the pipe
struct timespec record_duration;// observed record duration
int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read);
const uint32_t sample_rate = in_get_sample_rate(&stream->common);
if (rc == 0) {
// for how long have we been recording?
record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec;
record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec;
if (record_duration.tv_nsec < 0) {
record_duration.tv_sec--;
record_duration.tv_nsec += 1000000000;
}
// read_counter_frames contains the number of frames that have been read since the
// beginning of recording (including this call): it's converted to usec and compared to
// how long we've been recording for, which gives us how long we must wait to sync the
// projected recording time, and the observed recording time.
long projected_vs_observed_offset_us =
((int64_t)(in->read_counter_frames
- (record_duration.tv_sec*sample_rate)))
* 1000000 / sample_rate
- (record_duration.tv_nsec / 1000);
SUBMIX_ALOGV(" record duration %5lds %3ldms, will wait: %7ldus",
record_duration.tv_sec, record_duration.tv_nsec/1000000,
projected_vs_observed_offset_us);
if (projected_vs_observed_offset_us > 0) {
usleep(projected_vs_observed_offset_us);
}
}
SUBMIX_ALOGV("in_read returns %zu", bytes);
return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
(void)stream;
return 0;
}
static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
(void)stream;
(void)effect;
return 0;
}
static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
(void)stream;
(void)effect;
return 0;
}
static int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address)
{
struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev);
ALOGD("adev_open_output_stream(address=%s)", address);
struct submix_stream_out *out;
bool force_pipe_creation = false;
(void)handle;
(void)devices;
(void)flags;
*stream_out = NULL;
// Make sure it's possible to open the device given the current audio config.
submix_sanitize_config(config, false);
int route_idx = -1;
pthread_mutex_lock(&rsxadev->lock);
status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx);
if (res != OK) {
ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address);
pthread_mutex_unlock(&rsxadev->lock);
return res;
}
if (!submix_open_validate_l(rsxadev, route_idx, config, false)) {
ALOGE("adev_open_output_stream(): Unable to open output stream for address %s", address);
pthread_mutex_unlock(&rsxadev->lock);
return -EINVAL;
}
out = (struct submix_stream_out *)calloc(1, sizeof(struct submix_stream_out));
if (!out) {
pthread_mutex_unlock(&rsxadev->lock);
return -ENOMEM;
}
// Initialize the function pointer tables (v-tables).
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
out->stream.common.get_channels = out_get_channels;
out->stream.common.get_format = out_get_format;
out->stream.common.set_format = out_set_format;
out->stream.common.standby = out_standby;
out->stream.common.dump = out_dump;
out->stream.common.set_parameters = out_set_parameters;
out->stream.common.get_parameters = out_get_parameters;
out->stream.common.add_audio_effect = out_add_audio_effect;
out->stream.common.remove_audio_effect = out_remove_audio_effect;
out->stream.get_latency = out_get_latency;
out->stream.set_volume = out_set_volume;
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
#if ENABLE_RESAMPLING
// Recreate the pipe with the correct sample rate so that MonoPipe.write() rate limits
// writes correctly.
force_pipe_creation = rsxadev->routes[route_idx].config.common.sample_rate
!= config->sample_rate;
#endif // ENABLE_RESAMPLING
// If the sink has been shutdown or pipe recreation is forced (see above), delete the pipe so
// that it's recreated.
if ((rsxadev->routes[route_idx].rsxSink != NULL
&& rsxadev->routes[route_idx].rsxSink->isShutdown()) || force_pipe_creation) {
submix_audio_device_release_pipe_l(rsxadev, route_idx);
}
// Store a pointer to the device from the output stream.
out->dev = rsxadev;
// Initialize the pipe.
ALOGV("adev_open_output_stream(): about to create pipe at index %d", route_idx);
submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES,
DEFAULT_PIPE_PERIOD_COUNT, NULL, out, address, route_idx);
#if LOG_STREAMS_TO_FILES
out->log_fd = open(LOG_STREAM_OUT_FILENAME, O_CREAT | O_TRUNC | O_WRONLY,
LOG_STREAM_FILE_PERMISSIONS);
ALOGE_IF(out->log_fd < 0, "adev_open_output_stream(): log file open failed %s",
strerror(errno));
ALOGV("adev_open_output_stream(): log_fd = %d", out->log_fd);
#endif // LOG_STREAMS_TO_FILES
// Return the output stream.
*stream_out = &out->stream;
pthread_mutex_unlock(&rsxadev->lock);
return 0;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device(
const_cast<struct audio_hw_device*>(dev));
struct submix_stream_out * const out = audio_stream_out_get_submix_stream_out(stream);
pthread_mutex_lock(&rsxadev->lock);
ALOGD("adev_close_output_stream() addr = %s", rsxadev->routes[out->route_handle].address);
submix_audio_device_destroy_pipe_l(audio_hw_device_get_submix_audio_device(dev), NULL, out);
#if LOG_STREAMS_TO_FILES
if (out->log_fd >= 0) close(out->log_fd);
#endif // LOG_STREAMS_TO_FILES
pthread_mutex_unlock(&rsxadev->lock);
free(out);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
(void)dev;
(void)kvpairs;
return -ENOSYS;
}
static char * adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
(void)dev;
(void)keys;
return strdup("");;
}
static int adev_init_check(const struct audio_hw_device *dev)
{
ALOGI("adev_init_check()");
(void)dev;
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
(void)dev;
(void)volume;
return -ENOSYS;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
(void)dev;
(void)volume;
return -ENOSYS;
}
static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
{
(void)dev;
(void)volume;
return -ENOSYS;
}
static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
{
(void)dev;
(void)muted;
return -ENOSYS;
}
static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
{
(void)dev;
(void)muted;
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
(void)dev;
(void)mode;
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
(void)dev;
(void)state;
return -ENOSYS;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
(void)dev;
(void)state;
return -ENOSYS;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
if (audio_is_linear_pcm(config->format)) {
size_t max_buffer_period_size_frames = 0;
struct submix_audio_device * rsxadev = audio_hw_device_get_submix_audio_device(
const_cast<struct audio_hw_device*>(dev));
// look for the largest buffer period size
for (int i = 0 ; i < MAX_ROUTES ; i++) {
if (rsxadev->routes[i].config.buffer_period_size_frames > max_buffer_period_size_frames)
{
max_buffer_period_size_frames = rsxadev->routes[i].config.buffer_period_size_frames;
}
}
const size_t frame_size_in_bytes = audio_channel_count_from_in_mask(config->channel_mask) *
audio_bytes_per_sample(config->format);
const size_t buffer_size = max_buffer_period_size_frames * frame_size_in_bytes;
SUBMIX_ALOGV("adev_get_input_buffer_size() returns %zu bytes, %zu frames",
buffer_size, buffer_period_size_frames);
return buffer_size;
}
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,
audio_source_t source __unused)
{
struct submix_audio_device *rsxadev = audio_hw_device_get_submix_audio_device(dev);
struct submix_stream_in *in;
ALOGD("adev_open_input_stream(addr=%s)", address);
(void)handle;
(void)devices;
*stream_in = NULL;
// Do we already have a route for this address
int route_idx = -1;
pthread_mutex_lock(&rsxadev->lock);
status_t res = submix_get_route_idx_for_address_l(rsxadev, address, &route_idx);
if (res != OK) {
ALOGE("Error %d looking for address=%s in adev_open_output_stream", res, address);
pthread_mutex_unlock(&rsxadev->lock);
return res;
}
// Make sure it's possible to open the device given the current audio config.
submix_sanitize_config(config, true);
if (!submix_open_validate_l(rsxadev, route_idx, config, true)) {
ALOGE("adev_open_input_stream(): Unable to open input stream.");
pthread_mutex_unlock(&rsxadev->lock);
return -EINVAL;
}
#if ENABLE_LEGACY_INPUT_OPEN
in = rsxadev->routes[route_idx].input;
if (in) {
in->ref_count++;
sp<MonoPipe> sink = rsxadev->routes[route_idx].rsxSink;
ALOG_ASSERT(sink != NULL);
// If the sink has been shutdown, delete the pipe.
if (sink != NULL) {
if (sink->isShutdown()) {
ALOGD(" Non-NULL shut down sink when opening input stream, releasing, refcount=%d",
in->ref_count);
submix_audio_device_release_pipe_l(rsxadev, in->route_handle);
} else {
ALOGD(" Non-NULL sink when opening input stream, refcount=%d", in->ref_count);
}
} else {
ALOGE("NULL sink when opening input stream, refcount=%d", in->ref_count);
}
}
#else
in = NULL;
#endif // ENABLE_LEGACY_INPUT_OPEN
if (!in) {
in = (struct submix_stream_in *)calloc(1, sizeof(struct submix_stream_in));
if (!in) return -ENOMEM;
in->ref_count = 1;
// Initialize the function pointer tables (v-tables).
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 = rsxadev;
#if LOG_STREAMS_TO_FILES
in->log_fd = -1;
#endif
}
// Initialize the input stream.
in->read_counter_frames = 0;
in->input_standby = true;
if (rsxadev->routes[route_idx].output != NULL) {
in->output_standby_rec_thr = rsxadev->routes[route_idx].output->output_standby;
} else {
in->output_standby_rec_thr = true;
}
in->read_error_count = 0;
// Initialize the pipe.
ALOGV("adev_open_input_stream(): about to create pipe");
submix_audio_device_create_pipe_l(rsxadev, config, DEFAULT_PIPE_SIZE_IN_FRAMES,
DEFAULT_PIPE_PERIOD_COUNT, in, NULL, address, route_idx);
#if LOG_STREAMS_TO_FILES
if (in->log_fd >= 0) close(in->log_fd);
in->log_fd = open(LOG_STREAM_IN_FILENAME, O_CREAT | O_TRUNC | O_WRONLY,
LOG_STREAM_FILE_PERMISSIONS);
ALOGE_IF(in->log_fd < 0, "adev_open_input_stream(): log file open failed %s",
strerror(errno));
ALOGV("adev_open_input_stream(): log_fd = %d", in->log_fd);
#endif // LOG_STREAMS_TO_FILES
// Return the input stream.
*stream_in = &in->stream;
pthread_mutex_unlock(&rsxadev->lock);
return 0;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream)
{
struct submix_audio_device * const rsxadev = audio_hw_device_get_submix_audio_device(dev);
struct submix_stream_in * const in = audio_stream_in_get_submix_stream_in(stream);
ALOGD("adev_close_input_stream()");
pthread_mutex_lock(&rsxadev->lock);
submix_audio_device_destroy_pipe_l(rsxadev, in, NULL);
#if LOG_STREAMS_TO_FILES
if (in->log_fd >= 0) close(in->log_fd);
#endif // LOG_STREAMS_TO_FILES
#if ENABLE_LEGACY_INPUT_OPEN
if (in->ref_count == 0) free(in);
#else
free(in);
#endif // ENABLE_LEGACY_INPUT_OPEN
pthread_mutex_unlock(&rsxadev->lock);
}
static int adev_dump(const audio_hw_device_t *device, int fd)
{
const struct submix_audio_device * rsxadev = //audio_hw_device_get_submix_audio_device(device);
reinterpret_cast<const struct submix_audio_device *>(
reinterpret_cast<const uint8_t *>(device) -
offsetof(struct submix_audio_device, device));
char msg[100];
int n = sprintf(msg, "\nReroute submix audio module:\n");
write(fd, &msg, n);
for (int i=0 ; i < MAX_ROUTES ; i++) {
n = sprintf(msg, " route[%d] rate in=%d out=%d, addr=[%s]\n", i,
rsxadev->routes[i].config.input_sample_rate,
rsxadev->routes[i].config.output_sample_rate,
rsxadev->routes[i].address);
write(fd, &msg, n);
}
return 0;
}
static int adev_close(hw_device_t *device)
{
ALOGI("adev_close()");
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
ALOGI("adev_open(name=%s)", name);
struct submix_audio_device *rsxadev;
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
rsxadev = (submix_audio_device*) calloc(1, sizeof(struct submix_audio_device));
if (!rsxadev)
return -ENOMEM;
rsxadev->device.common.tag = HARDWARE_DEVICE_TAG;
rsxadev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
rsxadev->device.common.module = (struct hw_module_t *) module;
rsxadev->device.common.close = adev_close;
rsxadev->device.init_check = adev_init_check;
rsxadev->device.set_voice_volume = adev_set_voice_volume;
rsxadev->device.set_master_volume = adev_set_master_volume;
rsxadev->device.get_master_volume = adev_get_master_volume;
rsxadev->device.set_master_mute = adev_set_master_mute;
rsxadev->device.get_master_mute = adev_get_master_mute;
rsxadev->device.set_mode = adev_set_mode;
rsxadev->device.set_mic_mute = adev_set_mic_mute;
rsxadev->device.get_mic_mute = adev_get_mic_mute;
rsxadev->device.set_parameters = adev_set_parameters;
rsxadev->device.get_parameters = adev_get_parameters;
rsxadev->device.get_input_buffer_size = adev_get_input_buffer_size;
rsxadev->device.open_output_stream = adev_open_output_stream;
rsxadev->device.close_output_stream = adev_close_output_stream;
rsxadev->device.open_input_stream = adev_open_input_stream;
rsxadev->device.close_input_stream = adev_close_input_stream;
rsxadev->device.dump = adev_dump;
for (int i=0 ; i < MAX_ROUTES ; i++) {
memset(&rsxadev->routes[i], 0, sizeof(route_config));
strcpy(rsxadev->routes[i].address, "");
}
*device = &rsxadev->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 */ "Wifi Display audio HAL",
/* author */ "The Android Open Source Project",
/* methods */ &hal_module_methods,
/* dso */ NULL,
/* reserved */ { 0 },
},
};
} //namespace android
} //extern "C"
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