commit cf5f6d2825d9a8430a291042ca9c6f68e5b666d0
author Phil Burk <philburk@google.com> Fri May 26 12:35:07 2017 -0700
committer Phil Burk <philburk@google.com> Fri May 26 15:38:09 2017 -0700
tree 01266d4a2fdf5ccce8067b583be1f3a0ac9b81f0
parent 433ecd85ef10284a71c4ea57279f53427fb9983d

aaudio: reduce logspam, improve critical logs

This will make AAudio less annoying and easier to debug.

Bug: 62080950
Test: look in logcat
Change-Id: Id7ae26a212f83ea8be0b285bd20334eb48607be8
Signed-off-by: Phil Burk <philburk@google.com>

media/libaaudio/src/legacy/AudioStreamTrack.cpp

diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 9c433cd..e566332 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AudioStreamTrack"
+#define LOG_TAG "AAudio"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
 
@@ -58,15 +58,11 @@
         return result;
     }
 
-    ALOGD("AudioStreamTrack::open = %p", this);
-
     // Try to create an AudioTrack
     // Use stereo if unspecified.
     int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
                               ? 2 : getSamplesPerFrame();
     audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(samplesPerFrame);
-    ALOGD("AudioStreamTrack::open(), samplesPerFrame = %d, channelMask = 0x%08x",
-            samplesPerFrame, channelMask);
 
     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
     aaudio_performance_mode_t perfMode = getPerformanceMode();
@@ -87,8 +83,7 @@
             break;
     }
 
-    int32_t frameCount = builder.getBufferCapacity();
-    ALOGD("AudioStreamTrack::open(), requested buffer capacity %d", frameCount);
+    size_t frameCount = (size_t)builder.getBufferCapacity();
 
     int32_t notificationFrames = 0;
 
@@ -118,7 +113,8 @@
     }
     mCallbackBufferSize = builder.getFramesPerDataCallback();
 
-    ALOGD("AudioStreamTrack::open(), notificationFrames = %d", notificationFrames);
+    ALOGD("AudioStreamTrack::open(), request notificationFrames = %d, frameCount = %u",
+          notificationFrames, (uint)frameCount);
     mAudioTrack = new AudioTrack(
             (audio_stream_type_t) AUDIO_STREAM_MUSIC,
             getSampleRate(),
@@ -135,7 +131,6 @@
 
     // Did we get a valid track?
     status_t status = mAudioTrack->initCheck();
-    ALOGD("AudioStreamTrack::open(), initCheck() returned %d", status);
     if (status != NO_ERROR) {
         close();
         ALOGE("AudioStreamTrack::open(), initCheck() returned %d", status);
@@ -144,11 +139,16 @@
 
     // Get the actual values from the AudioTrack.
     setSamplesPerFrame(mAudioTrack->channelCount());
-    setSampleRate(mAudioTrack->getSampleRate());
     aaudio_audio_format_t aaudioFormat =
             AAudioConvert_androidToAAudioDataFormat(mAudioTrack->format());
     setFormat(aaudioFormat);
 
+    int32_t actualSampleRate = mAudioTrack->getSampleRate();
+    ALOGW_IF(actualSampleRate != getSampleRate(),
+             "AudioStreamTrack::open() sampleRate changed from %d to %d",
+             getSampleRate(), actualSampleRate);
+    setSampleRate(actualSampleRate);
+
     // We may need to pass the data through a block size adapter to guarantee constant size.
     if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) {
         int callbackSizeBytes = getBytesPerFrame() * mCallbackBufferSize;