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platform_hardware_libhardware/modules/audio_remote_submix/audio_hw.cpp
Mikhail Naganov 80179932c9 r_submix: Add some tests 
Add simple tests to verify remote submix behavior regarding
blocking writes. Currently one of the tests fails (does not finish),
this needs to be fixed.

Also fixed some minor issues in remote submix module code.

Bug: 73175392
Test: r_submix_tests
Change-Id: Ic88d0c385c0102e35b4f751f9c5cd8a6488949c8
2018-02-15 17:09:47 -08:00

1779 lines
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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 <unistd.h>
#include <cutils/compiler.h>
#include <cutils/properties.h>
#include <cutils/str_parms.h>
#include <log/log.h>
#include <utils/String8.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>
#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 {
// Uncomment to enable extremely verbose logging in this module.
// #define SUBMIX_VERBOSE_LOGGING
#if defined(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 uint16_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();
}
if (rsxadev->routes[route_idx].rsxSource != 0) {
rsxadev->routes[route_idx].rsxSource.clear();
}
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)
{
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 %llu frames from the pipe to avoid blocking",
(unsigned long long)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 ? (unsigned long long)*frames : -1ULL,
timestamp ? (unsigned long long)timestamp->tv_sec : -1ULL);
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;
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, max_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_input_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 = snprintf(msg, sizeof(msg), "\nReroute submix audio module:\n");
write(fd, &msg, n);
for (int i=0 ; i < MAX_ROUTES ; i++) {
n = snprintf(msg, sizeof(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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