/*
* Copyright (C) 2015 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 "APM::AudioPort"
//#define LOG_NDEBUG 0
#include "TypeConverter.h"
#include "AudioPort.h"
#include "HwModule.h"
#include "AudioGain.h"
#include <policy.h>
#ifndef ARRAY_SIZE
#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
#endif
namespace android {
int32_t volatile AudioPort::mNextUniqueId = 1;
// --- AudioPort class implementation
void AudioPort::attach(const sp<HwModule>& module)
{
mModule = module;
}
// Note that is a different namespace than AudioFlinger unique IDs
audio_port_handle_t AudioPort::getNextUniqueId()
{
return static_cast<audio_port_handle_t>(android_atomic_inc(&mNextUniqueId));
}
audio_module_handle_t AudioPort::getModuleHandle() const
{
if (mModule == 0) {
return AUDIO_MODULE_HANDLE_NONE;
}
return mModule->mHandle;
}
uint32_t AudioPort::getModuleVersion() const
{
if (mModule == 0) {
return 0;
}
return mModule->getHalVersion();
}
const char *AudioPort::getModuleName() const
{
if (mModule == 0) {
return "invalid module";
}
return mModule->getName();
}
void AudioPort::toAudioPort(struct audio_port *port) const
{
// TODO: update this function once audio_port structure reflects the new profile definition.
// For compatibility reason: flatening the AudioProfile into audio_port structure.
SortedVector<audio_format_t> flatenedFormats;
SampleRateVector flatenedRates;
ChannelsVector flatenedChannels;
for (size_t profileIndex = 0; profileIndex < mProfiles.size(); profileIndex++) {
if (mProfiles[profileIndex]->isValid()) {
audio_format_t formatToExport = mProfiles[profileIndex]->getFormat();
const SampleRateVector &ratesToExport = mProfiles[profileIndex]->getSampleRates();
const ChannelsVector &channelsToExport = mProfiles[profileIndex]->getChannels();
if (flatenedFormats.indexOf(formatToExport) < 0) {
flatenedFormats.add(formatToExport);
}
for (size_t rateIndex = 0; rateIndex < ratesToExport.size(); rateIndex++) {
uint32_t rate = ratesToExport[rateIndex];
if (flatenedRates.indexOf(rate) < 0) {
flatenedRates.add(rate);
}
}
for (size_t chanIndex = 0; chanIndex < channelsToExport.size(); chanIndex++) {
audio_channel_mask_t channels = channelsToExport[chanIndex];
if (flatenedChannels.indexOf(channels) < 0) {
flatenedChannels.add(channels);
}
}
if (flatenedRates.size() > AUDIO_PORT_MAX_SAMPLING_RATES ||
flatenedChannels.size() > AUDIO_PORT_MAX_CHANNEL_MASKS ||
flatenedFormats.size() > AUDIO_PORT_MAX_FORMATS) {
ALOGE("%s: bailing out: cannot export profiles to port config", __FUNCTION__);
return;
}
}
}
port->role = mRole;
port->type = mType;
strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
port->num_sample_rates = flatenedRates.size();
port->num_channel_masks = flatenedChannels.size();
port->num_formats = flatenedFormats.size();
for (size_t i = 0; i < flatenedRates.size(); i++) {
port->sample_rates[i] = flatenedRates[i];
}
for (size_t i = 0; i < flatenedChannels.size(); i++) {
port->channel_masks[i] = flatenedChannels[i];
}
for (size_t i = 0; i < flatenedFormats.size(); i++) {
port->formats[i] = flatenedFormats[i];
}
ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
uint32_t i;
for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
port->gains[i] = mGains[i]->getGain();
}
port->num_gains = i;
}
void AudioPort::importAudioPort(const sp<AudioPort> port)
{
size_t indexToImport;
for (indexToImport = 0; indexToImport < port->mProfiles.size(); indexToImport++) {
const sp<AudioProfile> &profileToImport = port->mProfiles[indexToImport];
if (profileToImport->isValid()) {
// Import only valid port, i.e. valid format, non empty rates and channels masks
bool hasSameProfile = false;
for (size_t profileIndex = 0; profileIndex < mProfiles.size(); profileIndex++) {
if (*mProfiles[profileIndex] == *profileToImport) {
// never import a profile twice
hasSameProfile = true;
break;
}
}
if (hasSameProfile) { // never import a same profile twice
continue;
}
addAudioProfile(profileToImport);
}
}
}
void AudioPort::pickSamplingRate(uint32_t &pickedRate,const SampleRateVector &samplingRates) const
{
pickedRate = 0;
// For direct outputs, pick minimum sampling rate: this helps ensuring that the
// channel count / sampling rate combination chosen will be supported by the connected
// sink
if (isDirectOutput()) {
uint32_t samplingRate = UINT_MAX;
for (size_t i = 0; i < samplingRates.size(); i ++) {
if ((samplingRates[i] < samplingRate) && (samplingRates[i] > 0)) {
samplingRate = samplingRates[i];
}
}
pickedRate = (samplingRate == UINT_MAX) ? 0 : samplingRate;
} else {
uint32_t maxRate = SAMPLE_RATE_HZ_MAX;
// For mixed output and inputs, use max mixer sampling rates. Do not
// limit sampling rate otherwise
// For inputs, also see checkCompatibleSamplingRate().
if (mType != AUDIO_PORT_TYPE_MIX) {
maxRate = UINT_MAX;
}
// TODO: should mSamplingRates[] be ordered in terms of our preference
// and we return the first (and hence most preferred) match? This is of concern if
// we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
for (size_t i = 0; i < samplingRates.size(); i ++) {
if ((samplingRates[i] > pickedRate) && (samplingRates[i] <= maxRate)) {
pickedRate = samplingRates[i];
}
}
}
}
void AudioPort::pickChannelMask(audio_channel_mask_t &pickedChannelMask,
const ChannelsVector &channelMasks) const
{
pickedChannelMask = AUDIO_CHANNEL_NONE;
// For direct outputs, pick minimum channel count: this helps ensuring that the
// channel count / sampling rate combination chosen will be supported by the connected
// sink
if (isDirectOutput()) {
uint32_t channelCount = UINT_MAX;
for (size_t i = 0; i < channelMasks.size(); i ++) {
uint32_t cnlCount;
if (useInputChannelMask()) {
cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
} else {
cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
}
if ((cnlCount < channelCount) && (cnlCount > 0)) {
pickedChannelMask = channelMasks[i];
channelCount = cnlCount;
}
}
} else {
uint32_t channelCount = 0;
uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
// For mixed output and inputs, use max mixer channel count. Do not
// limit channel count otherwise
if (mType != AUDIO_PORT_TYPE_MIX) {
maxCount = UINT_MAX;
}
for (size_t i = 0; i < channelMasks.size(); i ++) {
uint32_t cnlCount;
if (useInputChannelMask()) {
cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
} else {
cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
}
if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
pickedChannelMask = channelMasks[i];
channelCount = cnlCount;
}
}
}
}
/* format in order of increasing preference */
const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
AUDIO_FORMAT_DEFAULT,
AUDIO_FORMAT_PCM_16_BIT,
AUDIO_FORMAT_PCM_8_24_BIT,
AUDIO_FORMAT_PCM_24_BIT_PACKED,
AUDIO_FORMAT_PCM_32_BIT,
AUDIO_FORMAT_PCM_FLOAT,
};
int AudioPort::compareFormats(audio_format_t format1, audio_format_t format2)
{
// NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
// compressed format and better than any PCM format. This is by design of pickFormat()
if (!audio_is_linear_pcm(format1)) {
if (!audio_is_linear_pcm(format2)) {
return 0;
}
return 1;
}
if (!audio_is_linear_pcm(format2)) {
return -1;
}
int index1 = -1, index2 = -1;
for (size_t i = 0;
(i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
i ++) {
if (sPcmFormatCompareTable[i] == format1) {
index1 = i;
}
if (sPcmFormatCompareTable[i] == format2) {
index2 = i;
}
}
// format1 not found => index1 < 0 => format2 > format1
// format2 not found => index2 < 0 => format2 < format1
return index1 - index2;
}
bool AudioPort::isBetterFormatMatch(audio_format_t newFormat,
audio_format_t currentFormat,
audio_format_t targetFormat)
{
if (newFormat == currentFormat) {
return false;
}
if (currentFormat == AUDIO_FORMAT_INVALID) {
return true;
}
if (newFormat == targetFormat) {
return true;
}
int currentDiffBytes = (int)audio_bytes_per_sample(targetFormat) -
audio_bytes_per_sample(currentFormat);
int newDiffBytes = (int)audio_bytes_per_sample(targetFormat) -
audio_bytes_per_sample(newFormat);
if (abs(newDiffBytes) < abs(currentDiffBytes)) {
return true;
} else if (abs(newDiffBytes) == abs(currentDiffBytes)) {
return (newDiffBytes >= 0);
}
return false;
}
void AudioPort::pickAudioProfile(uint32_t &samplingRate,
audio_channel_mask_t &channelMask,
audio_format_t &format) const
{
format = AUDIO_FORMAT_DEFAULT;
samplingRate = 0;
channelMask = AUDIO_CHANNEL_NONE;
// special case for uninitialized dynamic profile
if (!mProfiles.hasValidProfile()) {
return;
}
audio_format_t bestFormat = sPcmFormatCompareTable[ARRAY_SIZE(sPcmFormatCompareTable) - 1];
// For mixed output and inputs, use best mixer output format.
// Do not limit format otherwise
if ((mType != AUDIO_PORT_TYPE_MIX) || isDirectOutput()) {
bestFormat = AUDIO_FORMAT_INVALID;
}
for (size_t i = 0; i < mProfiles.size(); i ++) {
if (!mProfiles[i]->isValid()) {
continue;
}
audio_format_t formatToCompare = mProfiles[i]->getFormat();
if ((compareFormats(formatToCompare, format) > 0) &&
(compareFormats(formatToCompare, bestFormat) <= 0)) {
uint32_t pickedSamplingRate = 0;
audio_channel_mask_t pickedChannelMask = AUDIO_CHANNEL_NONE;
pickChannelMask(pickedChannelMask, mProfiles[i]->getChannels());
pickSamplingRate(pickedSamplingRate, mProfiles[i]->getSampleRates());
if (formatToCompare != AUDIO_FORMAT_DEFAULT && pickedChannelMask != AUDIO_CHANNEL_NONE
&& pickedSamplingRate != 0) {
format = formatToCompare;
channelMask = pickedChannelMask;
samplingRate = pickedSamplingRate;
// TODO: shall we return on the first one or still trying to pick a better Profile?
}
}
}
ALOGV("%s Port[nm:%s] profile rate=%d, format=%d, channels=%d", __FUNCTION__, mName.string(),
samplingRate, channelMask, format);
}
status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig, int index) const
{
if (index < 0 || (size_t)index >= mGains.size()) {
return BAD_VALUE;
}
return mGains[index]->checkConfig(gainConfig);
}
void AudioPort::dump(int fd, int spaces, bool verbose) const
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
if (!mName.isEmpty()) {
snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
result.append(buffer);
write(fd, result.string(), result.size());
}
if (verbose) {
mProfiles.dump(fd, spaces);
if (mGains.size() != 0) {
snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
result = buffer;
write(fd, result.string(), result.size());
for (size_t i = 0; i < mGains.size(); i++) {
mGains[i]->dump(fd, spaces + 2, i);
}
}
}
}
void AudioPort::log(const char* indent) const
{
ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.string(), mType, mRole);
}
// --- AudioPortConfig class implementation
AudioPortConfig::AudioPortConfig()
{
mSamplingRate = 0;
mChannelMask = AUDIO_CHANNEL_NONE;
mFormat = AUDIO_FORMAT_INVALID;
mGain.index = -1;
}
status_t AudioPortConfig::applyAudioPortConfig(const struct audio_port_config *config,
struct audio_port_config *backupConfig)
{
struct audio_port_config localBackupConfig;
status_t status = NO_ERROR;
localBackupConfig.config_mask = config->config_mask;
toAudioPortConfig(&localBackupConfig);
sp<AudioPort> audioport = getAudioPort();
if (audioport == 0) {
status = NO_INIT;
goto exit;
}
status = audioport->checkExactAudioProfile(config->sample_rate,
config->channel_mask,
config->format);
if (status != NO_ERROR) {
goto exit;
}
if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
mSamplingRate = config->sample_rate;
}
if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
mChannelMask = config->channel_mask;
}
if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
mFormat = config->format;
}
if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
status = audioport->checkGain(&config->gain, config->gain.index);
if (status != NO_ERROR) {
goto exit;
}
mGain = config->gain;
}
exit:
if (status != NO_ERROR) {
applyAudioPortConfig(&localBackupConfig);
}
if (backupConfig != NULL) {
*backupConfig = localBackupConfig;
}
return status;
}
void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
dstConfig->sample_rate = mSamplingRate;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
dstConfig->sample_rate = srcConfig->sample_rate;
}
} else {
dstConfig->sample_rate = 0;
}
if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
dstConfig->channel_mask = mChannelMask;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
dstConfig->channel_mask = srcConfig->channel_mask;
}
} else {
dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
}
if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
dstConfig->format = mFormat;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
dstConfig->format = srcConfig->format;
}
} else {
dstConfig->format = AUDIO_FORMAT_INVALID;
}
sp<AudioPort> audioport = getAudioPort();
if ((dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) && audioport != NULL) {
dstConfig->gain = mGain;
if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)
&& audioport->checkGain(&srcConfig->gain, srcConfig->gain.index) == OK) {
dstConfig->gain = srcConfig->gain;
}
} else {
dstConfig->gain.index = -1;
}
if (dstConfig->gain.index != -1) {
dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
} else {
dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
}
}
}; // namespace android