Sample rate 0 means a route-dependent rate
Remove check for primary _output_ [sic] sampling rate for fast capture.
Clean up AudioRecord handling of frame count and sample rate.
Clean up AudioTrack handling of notification period updates.
Make AudioRecord and AudioTrack more similar in order of operation, comments, and whitespace.
Bug: 25641253
Bug: 21019153
Change-Id: I24a6677945987fc39a9bf93f70357e4bc7410f98
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 36c1d10..8e02900 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1192,6 +1192,7 @@
mSampleRate = mAfSampleRate;
mOriginalSampleRate = mAfSampleRate;
}
+
// Client can only express a preference for FAST. Server will perform additional tests.
if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
bool useCaseAllowed =
@@ -1207,7 +1208,7 @@
// sample rates must also match
bool fastAllowed = useCaseAllowed && (mSampleRate == mAfSampleRate);
if (!fastAllowed) {
- ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d,"
+ ALOGW("AUDIO_OUTPUT_FLAG_FAST denied by client; transfer %d, "
"track %u Hz, output %u Hz",
mTransfer, mSampleRate, mAfSampleRate);
// once denied, do not request again if IAudioTrack is re-created
@@ -1215,13 +1216,6 @@
}
}
- // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
- // n = 1 fast track with single buffering; nBuffering is ignored
- // n = 2 fast track with double buffering
- // n = 2 normal track, (including those with sample rate conversion)
- // n >= 3 very high latency or very small notification interval (unused).
- const uint32_t nBuffering = 2;
-
mNotificationFramesAct = mNotificationFramesReq;
size_t frameCount = mReqFrameCount;
@@ -1320,6 +1314,7 @@
// AudioFlinger now owns the reference to the I/O handle,
// so we are no longer responsible for releasing it.
+ // FIXME compare to AudioRecord
sp<IMemory> iMem = track->getCblk();
if (iMem == 0) {
ALOGE("Could not get control block");
@@ -1383,14 +1378,25 @@
//return NO_INIT;
}
}
- // Make sure that application is notified with sufficient margin before underrun
+
+ // Make sure that application is notified with sufficient margin before underrun.
+ // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where
+ // n = 1 fast track with single buffering; nBuffering is ignored
+ // n = 2 fast track with double buffering
+ // n = 2 normal track, (including those with sample rate conversion)
+ // n >= 3 very high latency or very small notification interval (unused).
+ // FIXME Move the computation from client side to server side,
+ // and allow nBuffering to be larger than 1 for OpenSL ES, like it can be for Java.
if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) {
- // Theoretically double-buffering is not required for fast tracks,
- // due to tighter scheduling. But in practice, to accommodate kernels with
- // scheduling jitter, and apps with computation jitter, we use double-buffering
- // for fast tracks just like normal streaming tracks.
- if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount / nBuffering) {
- mNotificationFramesAct = frameCount / nBuffering;
+ size_t maxNotificationFrames = frameCount;
+ if (!(trackFlags & IAudioFlinger::TRACK_FAST)) {
+ const uint32_t nBuffering = 2;
+ maxNotificationFrames /= nBuffering;
+ }
+ if (mNotificationFramesAct == 0 || mNotificationFramesAct > maxNotificationFrames) {
+ ALOGW("Client adjusted notificationFrames from %u to %zu for frameCount %zu",
+ mNotificationFramesAct, maxNotificationFrames, frameCount);
+ mNotificationFramesAct = (uint32_t) maxNotificationFrames;
}
}