Change references of Q19.12 to Q4.27 for clarity

Change-Id: I5beb7daf6ff9bc123ff3582f7c294edcaf8652f6
Signed-off-by: Andy Hung <hunga@google.com>

5 files changed, 10 insertions, 9 deletions

diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index a1783fe..2d67efb 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1181,7 +1181,7 @@ void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
             }
             switch (t1.mMixerFormat) {
             case AUDIO_FORMAT_PCM_FLOAT:
-                memcpy_to_float_from_q19_12(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
+                memcpy_to_float_from_q4_27(reinterpret_cast<float *>(out), outTemp, BLOCKSIZE * 2);
                 out += BLOCKSIZE * 2; // output is 2 floats/frame.
                 break;
             case AUDIO_FORMAT_PCM_16_BIT:
@@ -1274,7 +1274,7 @@ void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
         }
         switch (t1.mMixerFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
-            memcpy_to_float_from_q19_12(reinterpret_cast<float*>(out), outTemp, numFrames*2);
+            memcpy_to_float_from_q4_27(reinterpret_cast<float*>(out), outTemp, numFrames*2);
             break;
         case AUDIO_FORMAT_PCM_16_BIT:
             ditherAndClamp(out, outTemp, numFrames);
@@ -1330,8 +1330,8 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
                 in += 2;
                 int32_t l = mulRL(1, rl, vrl);
                 int32_t r = mulRL(0, rl, vrl);
-                *fout++ = float_from_q19_12(l);
-                *fout++ = float_from_q19_12(r);
+                *fout++ = float_from_q4_27(l);
+                *fout++ = float_from_q4_27(r);
                 // Note: In case of later int16_t sink output,
                 // conversion and clamping is done by memcpy_to_i16_from_float().
             } while (--outFrames);
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 0592855..b84567e 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -63,7 +63,7 @@ public:
     // A mono provider delivers a sequence of samples.
     // A stereo provider delivers a sequence of interleaved pairs of samples.
     // Multi-channel providers are not supported.
-    // In either case, 'out' holds interleaved pairs of fixed-point signed Q19.12.
+    // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
     // That is, for a mono provider, there is an implicit up-channeling.
     // Since this method accumulates, the caller is responsible for clearing 'out' initially.
     // FIXME assumes provider is always successful; it should return the actual frame count.
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 3ec9889..6fc06d8 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -66,7 +66,7 @@ private:
            // updated by RecordThread::readInputParameters_l()
             AudioResampler                      *mResampler;
 
-            // interleaved stereo pairs of fixed-point signed Q19.12
+            // interleaved stereo pairs of fixed-point Q4.27
             int32_t                             *mRsmpOutBuffer;
             // current allocated frame count for the above, which may be larger than needed
             size_t                              mRsmpOutFrameCount;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 65e9eec..feedd89 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -4954,7 +4954,7 @@ reacquire_wakelock:
                     // ditherAndClamp() works as long as all buffers returned by
                     // activeTrack->getNextBuffer() are 32 bit aligned which should be always true.
                     if (activeTrack->mChannelCount == 1) {
-                        // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t
+                        // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t
                         ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer,
                                 framesOut);
                         // the resampler always outputs stereo samples:
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 3ab3ba9..7b7cdac 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -352,8 +352,9 @@ int main(int argc, char* argv[]) {
     size_t out_frames = output_size/8;
 
     /* set volume precision to 12 bits, so the volume scale is 1<<12.
-     * This means the "integer" part fits in the Q19.12 precision
-     * representation of output int32_t.
+     * The output int32_t is represented as Q4.27, with 4 bits of guard
+     * followed by the int16_t Q.15 portion, and then 12 trailing bits of
+     * additional precision.
      *
      * Generally 0 < volumePrecision <= 14 (due to the limits of
      * int16_t values for Volume). volumePrecision cannot be 0 due