AAudio: Add sample rate conversion to MMAP
This CL adds sample rate conversion to the MMAP path in AAudio.
AAudio no longer needs to use AudioFlinger's sample rate converter for
MMAP streams. This effectively reduces the latency from 100ms to 30ms
on speaker, 3.5mm, and USB if MMAP is enabled and sample rate conversion
is needed.
A sample rate converter is added to the AAudio flowgraph. Since input
and output sample rates are different for the flowgraph, a variable
number of frames must be used. 8 frames are passed through the flowgraph
at any given point of times and the output of the flowgraph pulls as
much data as there exists in the flowgraph.
Burst sizes, capacity, latency, frames read, and frames written are all
changed reflect the sample rate of the app.
Flowgraph was also added to input MMAP streams.
Bug: 219533889
Test: OboeTester Test Input/Output with a variety of sample rates and formats
Test: atest AAudioTests
Change-Id: I9b6a2694921ea558fb91a231f956f638d3342837
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 7f228c7..2ef6fe5 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -45,6 +45,8 @@
#define OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS (3 * AAUDIO_NANOS_PER_MILLISECOND)
#define INPUT_ESTIMATED_HARDWARE_OFFSET_NANOS (-1 * AAUDIO_NANOS_PER_MILLISECOND)
+#define AAUDIO_MAX_OPEN_ATTEMPTS 10
+
using namespace android; // TODO just import names needed
using namespace aaudio; // TODO just import names needed
@@ -77,14 +79,23 @@
{AUDIO_FORMAT_PCM_24_BIT_PACKED, AUDIO_FORMAT_PCM_16_BIT}
};
-audio_format_t getNextFormatToTry(audio_format_t curFormat, audio_format_t returnedFromAPM) {
- if (returnedFromAPM != AUDIO_FORMAT_DEFAULT) {
- return returnedFromAPM;
- }
+audio_format_t getNextFormatToTry(audio_format_t curFormat) {
const auto it = NEXT_FORMAT_TO_TRY.find(curFormat);
- return it != NEXT_FORMAT_TO_TRY.end() ? it->second : AUDIO_FORMAT_DEFAULT;
+ return it != NEXT_FORMAT_TO_TRY.end() ? it->second : curFormat;
}
+struct configComp {
+ bool operator() (const audio_config_base_t& lhs, const audio_config_base_t& rhs) const {
+ if (lhs.sample_rate != rhs.sample_rate) {
+ return lhs.sample_rate < rhs.sample_rate;
+ } else if (lhs.channel_mask != rhs.channel_mask) {
+ return lhs.channel_mask < rhs.channel_mask;
+ } else {
+ return lhs.format < rhs.format;
+ }
+ }
+};
+
} // namespace
aaudio_result_t AAudioServiceEndpointMMAP::open(const aaudio::AAudioStreamRequest &request) {
@@ -101,60 +112,66 @@
legacy2aidl_pid_t_int32_t(IPCThreadState::self()->getCallingPid()));
audio_format_t audioFormat = getFormat();
- std::set<audio_format_t> formatsTried;
- while (true) {
- if (formatsTried.find(audioFormat) != formatsTried.end()) {
+ int32_t sampleRate = getSampleRate();
+ if (sampleRate == AAUDIO_UNSPECIFIED) {
+ sampleRate = AAUDIO_SAMPLE_RATE_DEFAULT;
+ }
+
+ const aaudio_direction_t direction = getDirection();
+ audio_config_base_t config;
+ config.format = audioFormat;
+ config.sample_rate = sampleRate;
+ config.channel_mask = AAudio_getChannelMaskForOpen(
+ getChannelMask(), getSamplesPerFrame(), direction == AAUDIO_DIRECTION_INPUT);
+
+ std::set<audio_config_base_t, configComp> configsTried;
+ int32_t numberOfAttempts = 0;
+ while (numberOfAttempts < AAUDIO_MAX_OPEN_ATTEMPTS) {
+ if (configsTried.find(config) != configsTried.end()) {
// APM returning something that has already tried.
- ALOGW("Have already tried to open with format=%#x, but failed before", audioFormat);
+ ALOGW("Have already tried to open with format=%#x and sr=%d, but failed before",
+ config.format, config.sample_rate);
break;
}
- formatsTried.insert(audioFormat);
+ configsTried.insert(config);
- audio_format_t nextFormatToTry = AUDIO_FORMAT_DEFAULT;
- result = openWithFormat(audioFormat, &nextFormatToTry);
+ audio_config_base_t previousConfig = config;
+ result = openWithConfig(&config);
if (result != AAUDIO_ERROR_UNAVAILABLE) {
// Return if it is successful or there is an error that is not
// AAUDIO_ERROR_UNAVAILABLE happens.
- ALOGI("Opened format=%#x with result=%d", audioFormat, result);
+ ALOGI("Opened format=%#x sr=%d, with result=%d", previousConfig.format,
+ previousConfig.sample_rate, result);
break;
}
- nextFormatToTry = getNextFormatToTry(audioFormat, nextFormatToTry);
- ALOGD("%s() %#x failed, perhaps due to format. Try again with %#x",
- __func__, audioFormat, nextFormatToTry);
- audioFormat = nextFormatToTry;
- if (audioFormat == AUDIO_FORMAT_DEFAULT) {
- // Nothing else to try
- break;
+ // Try other formats if the config from APM is the same as our current config.
+ // Some HALs may report its format support incorrectly.
+ if ((previousConfig.format == config.format) &&
+ (previousConfig.sample_rate == config.sample_rate)) {
+ config.format = getNextFormatToTry(config.format);
}
+
+ ALOGD("%s() %#x %d failed, perhaps due to format or sample rate. Try again with %#x %d",
+ __func__, previousConfig.format, previousConfig.sample_rate, config.format,
+ config.sample_rate);
+ numberOfAttempts++;
}
return result;
}
-aaudio_result_t AAudioServiceEndpointMMAP::openWithFormat(
- audio_format_t audioFormat, audio_format_t* nextFormatToTry) {
+aaudio_result_t AAudioServiceEndpointMMAP::openWithConfig(
+ audio_config_base_t* config) {
aaudio_result_t result = AAUDIO_OK;
- audio_config_base_t config;
+ audio_config_base_t currentConfig = *config;
audio_port_handle_t deviceId;
const audio_attributes_t attributes = getAudioAttributesFrom(this);
deviceId = mRequestedDeviceId;
- // Fill in config
- config.format = audioFormat;
-
- int32_t aaudioSampleRate = getSampleRate();
- if (aaudioSampleRate == AAUDIO_UNSPECIFIED) {
- aaudioSampleRate = AAUDIO_SAMPLE_RATE_DEFAULT;
- }
- config.sample_rate = aaudioSampleRate;
-
const aaudio_direction_t direction = getDirection();
- config.channel_mask = AAudio_getChannelMaskForOpen(
- getChannelMask(), getSamplesPerFrame(), direction == AAUDIO_DIRECTION_INPUT);
-
if (direction == AAUDIO_DIRECTION_OUTPUT) {
mHardwareTimeOffsetNanos = OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at DAC later
@@ -177,11 +194,11 @@
// Open HAL stream. Set mMmapStream
ALOGD("%s trying to open MMAP stream with format=%#x, "
"sample_rate=%u, channel_mask=%#x, device=%d",
- __func__, config.format, config.sample_rate,
- config.channel_mask, deviceId);
+ __func__, config->format, config->sample_rate,
+ config->channel_mask, deviceId);
const status_t status = MmapStreamInterface::openMmapStream(streamDirection,
&attributes,
- &config,
+ config,
mMmapClient,
&deviceId,
&sessionId,
@@ -195,9 +212,9 @@
// not match the hardware.
ALOGD("%s() - openMmapStream() returned status=%d, suggested format=%#x, sample_rate=%u, "
"channel_mask=%#x",
- __func__, status, config.format, config.sample_rate, config.channel_mask);
- *nextFormatToTry = config.format != audioFormat ? config.format
- : *nextFormatToTry;
+ __func__, status, config->format, config->sample_rate, config->channel_mask);
+ // Keep the channel mask of the current config
+ config->channel_mask = currentConfig.channel_mask;
return AAUDIO_ERROR_UNAVAILABLE;
}
@@ -217,7 +234,7 @@
setSessionId(actualSessionId);
ALOGD("%s(format = 0x%X) deviceId = %d, sessionId = %d",
- __func__, audioFormat, getDeviceId(), getSessionId());
+ __func__, config->format, getDeviceId(), getSessionId());
// Create MMAP/NOIRQ buffer.
result = createMmapBuffer();
@@ -227,11 +244,11 @@
// Get information about the stream and pass it back to the caller.
setChannelMask(AAudioConvert_androidToAAudioChannelMask(
- config.channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
- AAudio_isChannelIndexMask(config.channel_mask)));
+ config->channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
+ AAudio_isChannelIndexMask(config->channel_mask)));
- setFormat(config.format);
- setSampleRate(config.sample_rate);
+ setFormat(config->format);
+ setSampleRate(config->sample_rate);
setHardwareSampleRate(getSampleRate());
setHardwareFormat(getFormat());
setHardwareSamplesPerFrame(AAudioConvert_channelMaskToCount(getChannelMask()));