Change AudioTrack resampling buffers from 3 to 2

Move computation of minimum AudioTrack buffer size to server
for normal streaming PCM tracks.

Use server-side computation to exactly determine requirements
for the resampler to avoid triple buffering.

This reduces latency for normal audio tracks that require resampling,
and makes things consistent with the minimum buffer size.

Change-Id: I2f2ca0e599ee20e16559bc5c5dab61ed100da16c

1 files changed, 12 insertions, 19 deletions

diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 15dd408..8fcc09c 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -172,18 +172,6 @@ static int sFastTrackMultiplier = kFastTrackMultiplier;
 // and that all "fast" AudioRecord clients read from.  In either case, the size can be small.
 static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;
 
-// Returns the source frames needed to resample to destination frames.  This is not a precise
-// value and depends on the resampler (and possibly how it handles rounding internally).
-// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which
-// may not be a true if the resampler is asynchronous.
-static inline size_t sourceFramesNeeded(
-        uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) {
-    // +1 for rounding - always do this even if matched ratio
-    // +1 for additional sample needed for interpolation
-    return srcSampleRate == dstSampleRate ? dstFramesRequired :
-            size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1);
-}
-
 // ----------------------------------------------------------------------------
 
 static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
@@ -1495,20 +1483,25 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac
                 audio_is_linear_pcm(format),
                 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask);
         *flags &= ~IAudioFlinger::TRACK_FAST;
-        // For compatibility with AudioTrack calculation, buffer depth is forced
-        // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
-        // This is probably too conservative, but legacy application code may depend on it.
-        // If you change this calculation, also review the start threshold which is related.
+      }
+    }
+    // For normal PCM streaming tracks, update minimum frame count.
+    // For compatibility with AudioTrack calculation, buffer depth is forced
+    // to be at least 2 x the normal mixer frame count and cover audio hardware latency.
+    // This is probably too conservative, but legacy application code may depend on it.
+    // If you change this calculation, also review the start threshold which is related.
+    if (!(*flags & IAudioFlinger::TRACK_FAST)
+            && audio_is_linear_pcm(format) && sharedBuffer == 0) {
         uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream);
         uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate);
         if (minBufCount < 2) {
             minBufCount = 2;
         }
-        size_t minFrameCount = mNormalFrameCount * minBufCount;
-        if (frameCount < minFrameCount) {
+        size_t minFrameCount =
+                minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate);
+        if (frameCount < minFrameCount) { // including frameCount == 0
             frameCount = minFrameCount;
         }
-      }
     }
     *pFrameCount = frameCount;