audio: Provide a way for Module to specify nominal latency
The latency figure depends on the module implementation.
Instead of using a hardcoded value, each module should be
able to specify its own value. This value is then used
for calculating the minimum buffer size.
Set the nominal latency of the primary (CF) module to a high
value since the virtual device implementation fails CTS tests
if it attempts to pretend that it provides low latency.
Bug: 302132812
Test: atest CtsMediaAudioTestCases --test-filter=".*AudioTrackTest.*"
Test: atest CtsMediaAudioTestCases --test-filter=".*AudioRecordTest.*"
Change-Id: I8ce9f230378eea787c9b3c7ce3660c1e4e7bc895
diff --git a/audio/aidl/default/Module.cpp b/audio/aidl/default/Module.cpp
index a77397d..f725670 100644
--- a/audio/aidl/default/Module.cpp
+++ b/audio/aidl/default/Module.cpp
@@ -202,15 +202,17 @@
LOG(ERROR) << __func__ << ": non-positive buffer size " << in_bufferSizeFrames;
return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_ARGUMENT);
}
- if (in_bufferSizeFrames < kMinimumStreamBufferSizeFrames) {
- LOG(ERROR) << __func__ << ": insufficient buffer size " << in_bufferSizeFrames
- << ", must be at least " << kMinimumStreamBufferSizeFrames;
- return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_ARGUMENT);
- }
auto& configs = getConfig().portConfigs;
auto portConfigIt = findById<AudioPortConfig>(configs, in_portConfigId);
// Since this is a private method, it is assumed that
// validity of the portConfigId has already been checked.
+ const int32_t minimumStreamBufferSizeFrames = calculateBufferSizeFrames(
+ getNominalLatencyMs(*portConfigIt), portConfigIt->sampleRate.value().value);
+ if (in_bufferSizeFrames < minimumStreamBufferSizeFrames) {
+ LOG(ERROR) << __func__ << ": insufficient buffer size " << in_bufferSizeFrames
+ << ", must be at least " << minimumStreamBufferSizeFrames;
+ return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_ARGUMENT);
+ }
const size_t frameSize =
getFrameSizeInBytes(portConfigIt->format.value(), portConfigIt->channelMask.value());
if (frameSize == 0) {
@@ -238,8 +240,8 @@
StreamContext temp(
std::make_unique<StreamContext::CommandMQ>(1, true /*configureEventFlagWord*/),
std::make_unique<StreamContext::ReplyMQ>(1, true /*configureEventFlagWord*/),
- portConfigIt->portId, portConfigIt->format.value(),
- portConfigIt->channelMask.value(), portConfigIt->sampleRate.value().value, flags,
+ portConfigIt->format.value(), portConfigIt->channelMask.value(),
+ portConfigIt->sampleRate.value().value, flags, getNominalLatencyMs(*portConfigIt),
portConfigIt->ext.get<AudioPortExt::mix>().handle,
std::make_unique<StreamContext::DataMQ>(frameSize * in_bufferSizeFrames),
asyncCallback, outEventCallback,
@@ -359,6 +361,12 @@
return internal::getConfiguration(getType());
}
+int32_t Module::getNominalLatencyMs(const AudioPortConfig&) {
+ // Arbitrary value. Implementations must override this method to provide their actual latency.
+ static constexpr int32_t kLatencyMs = 5;
+ return kLatencyMs;
+}
+
std::vector<AudioRoute*> Module::getAudioRoutesForAudioPortImpl(int32_t portId) {
std::vector<AudioRoute*> result;
auto& routes = getConfig().routes;
@@ -965,11 +973,21 @@
return ndk::ScopedAStatus::fromExceptionCode(EX_ILLEGAL_STATE);
}
}
+ // Find the highest sample rate among mix port configs.
+ std::map<int32_t, AudioPortConfig*> sampleRates;
+ std::vector<AudioPortConfig*>& mixPortConfigs =
+ sources[0]->ext.getTag() == AudioPortExt::mix ? sources : sinks;
+ for (auto mix : mixPortConfigs) {
+ sampleRates.emplace(mix->sampleRate.value().value, mix);
+ }
*_aidl_return = in_requested;
- _aidl_return->minimumStreamBufferSizeFrames = kMinimumStreamBufferSizeFrames;
+ auto maxSampleRateIt = std::max_element(sampleRates.begin(), sampleRates.end());
+ const int32_t latencyMs = getNominalLatencyMs(*(maxSampleRateIt->second));
+ _aidl_return->minimumStreamBufferSizeFrames =
+ calculateBufferSizeFrames(latencyMs, maxSampleRateIt->first);
_aidl_return->latenciesMs.clear();
_aidl_return->latenciesMs.insert(_aidl_return->latenciesMs.end(),
- _aidl_return->sinkPortConfigIds.size(), kLatencyMs);
+ _aidl_return->sinkPortConfigIds.size(), latencyMs);
AudioPatch oldPatch{};
if (existing == patches.end()) {
_aidl_return->id = getConfig().nextPatchId++;