Android 中打开音频流所用的配置
我们在 AudioPolicyManager::onNewAudioModulesAvailableInt(DeviceVector *newDevices) 函数中看到它创建了 SwAudioOutputDescriptor 对象，后者的构造函数的定义 (位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp) 如下：
AudioOutputDescriptor::AudioOutputDescriptor(const sp& policyAudioPort,
AudioPolicyClientInterface *clientInterface)
: mPolicyAudioPort(policyAudioPort), mClientInterface(clientInterface)
{
if (mPolicyAudioPort.get() != nullptr) {
mPolicyAudioPort->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
if (mPolicyAudioPort->asAudioPort()->getGains().size() > 0) {
mPolicyAudioPort->asAudioPort()->getGains()[0]->getDefaultConfig(&mGain);
}
}
}
. . . . . .
SwAudioOutputDescriptor::SwAudioOutputDescriptor(const sp& profile,
AudioPolicyClientInterface *clientInterface)
: AudioOutputDescriptor(profile, clientInterface),
mProfile(profile), mIoHandle(AUDIO_IO_HANDLE_NONE), mLatency(0),
mFlags((audio_output_flags_t)0),
mOutput1(0), mOutput2(0), mDirectOpenCount(0),
mDirectClientSession(AUDIO_SESSION_NONE)
{
if (profile != NULL) {
mFlags = (audio_output_flags_t)profile->getFlags();
}
}
Android 中打开音频输出流，所需的主要参数采样率、通道掩码（通道数）、采样格式、增益和标记来自于传入的 IOProfile。如 Android 音频设备信息加载 一文中的说明，IOProfile 的信息主要来自于解析 audio_policy_configuration.xml 音频策略配置文件，更具地说，来自于音频策略配置文件中的 mixPort 元素，如下面 (位于 device/generic/car/emulator/audio/audio_policy_configuration.xml) 这样：
bus0_media_out
. . . . . .
IOProfile 通过 AudioRoute 与 DeviceDescriptor 建立连接。在音频策略配置文件中，AudioRoute 和 DeviceDescriptor 对应的 XML 元素定义如下面这样：
. . . . . .
SwAudioOutputDescriptor 的标记和增益直接来自于音频策略配置文件的 mixPort 元素。音频策略配置文件支持为 mixPort 定义增益，也支持为 devicePort 定义增益，上面的这个配置文件没有为 mixPort 定义增益。采样率、通道掩码（通道数）和采样格式则会从多个 AudioProfile 中选择最佳的一个，具体的策略如 PolicyAudioPort::pickAudioProfile() 函数的定义(位于 frameworks/av/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp)：
void PolicyAudioPort::pickSamplingRate(uint32_t &pickedRate,
const SampleRateSet &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 (const auto rate : samplingRates) {
if ((rate < samplingRate) && (rate > 0)) {
samplingRate = rate;
}
}
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 (asAudioPort()->getType() == 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 (const auto rate : samplingRates) {
if ((rate > pickedRate) && (rate <= maxRate)) {
pickedRate = rate;
}
}
}
}
void PolicyAudioPort::pickChannelMask(audio_channel_mask_t &pickedChannelMask,
const ChannelMaskSet &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 (const auto channelMask : channelMasks) {
uint32_t cnlCount;
if (asAudioPort()->useInputChannelMask()) {
cnlCount = audio_channel_count_from_in_mask(channelMask);
} else {
cnlCount = audio_channel_count_from_out_mask(channelMask);
}
if ((cnlCount < channelCount) && (cnlCount > 0)) {
pickedChannelMask = channelMask;
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 (asAudioPort()->getType() != AUDIO_PORT_TYPE_MIX) {
maxCount = UINT_MAX;
}
for (const auto channelMask : channelMasks) {
uint32_t cnlCount;
if (asAudioPort()->useInputChannelMask()) {
cnlCount = audio_channel_count_from_in_mask(channelMask);
} else {
cnlCount = audio_channel_count_from_out_mask(channelMask);
}
if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
pickedChannelMask = channelMask;
channelCount = cnlCount;
}
}
}
}
/* format in order of increasing preference */
const audio_format_t PolicyAudioPort::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 PolicyAudioPort::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;
}
uint32_t PolicyAudioPort::formatDistance(audio_format_t format1, audio_format_t format2)
{
if (format1 == format2) {
return 0;
}
if (format1 == AUDIO_FORMAT_INVALID || format2 == AUDIO_FORMAT_INVALID) {
return kFormatDistanceMax;
}
int diffBytes = (int)audio_bytes_per_sample(format1) -
audio_bytes_per_sample(format2);
return abs(diffBytes);
}
bool PolicyAudioPort::isBetterFormatMatch(audio_format_t newFormat,
audio_format_t currentFormat,
audio_format_t targetFormat)
{
return formatDistance(newFormat, targetFormat) < formatDistance(currentFormat, targetFormat);
}
void PolicyAudioPort::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 (!asAudioPort()->hasValidAudioProfile()) {
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 ((asAudioPort()->getType() != AUDIO_PORT_TYPE_MIX) || isDirectOutput()) {
bestFormat = AUDIO_FORMAT_INVALID;
}
const AudioProfileVector& audioProfiles = asAudioPort()->getAudioProfiles();
for (size_t i = 0; i < audioProfiles.size(); i ++) {
if (!audioProfiles[i]->isValid()) {
continue;
}
audio_format_t formatToCompare = audioProfiles[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, audioProfiles[i]->getChannels());
pickSamplingRate(pickedSamplingRate, audioProfiles[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__,
asAudioPort()->getName().c_str(), samplingRate, channelMask, format);
}
对于采样格式，精度越高优先级越高；对于通道掩码（通道数），如果是直接输出 (Direct Output)，则通道数越小优先级越高，否则，通道数越大优先级越高；对于采样率，如果是直接输出 (Direct Output)，则采样率越小优先级越高，否则，采样率越大优先级越高。