Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock.

The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).

The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.

AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.

AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.

AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.

Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee

1 files changed, 24 insertions, 11 deletions

diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ab5ac64..cd47fdf 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -32,6 +32,18 @@ namespace android {
 #define MAX_RUN_TIMEOUT_MS      1000
 #define WAIT_PERIOD_MS          10
 
+#define CBLK_UNDERRUN_MSK       0x0001
+#define CBLK_UNDERRUN_ON        0x0001  // underrun (out) or overrrun (in) indication
+#define CBLK_UNDERRUN_OFF       0x0000  // no underrun
+#define CBLK_DIRECTION_MSK      0x0002
+#define CBLK_DIRECTION_OUT      0x0002  // this cblk is for an AudioTrack
+#define CBLK_DIRECTION_IN       0x0000  // this cblk is for an AudioRecord
+#define CBLK_FORCEREADY_MSK     0x0004
+#define CBLK_FORCEREADY_ON      0x0004  // track is considered ready immediately by AudioFlinger
+#define CBLK_FORCEREADY_OFF     0x0000  // track is ready when buffer full
+#define CBLK_INVALID_MSK        0x0008
+#define CBLK_INVALID_ON         0x0008  // track buffer is invalidated by AudioFlinger: must be re-created
+#define CBLK_INVALID_OFF        0x0000
 
 struct audio_track_cblk_t
 {
@@ -44,12 +56,12 @@ struct audio_track_cblk_t
     volatile    uint32_t    server;
                 uint32_t    userBase;
                 uint32_t    serverBase;
-    void*       buffers;
-    uint32_t    frameCount;
-    // Cache line boundary
-    uint32_t    loopStart;
-    uint32_t    loopEnd;
-    int         loopCount;
+                void*       buffers;
+                uint32_t    frameCount;
+                // Cache line boundary
+                uint32_t    loopStart;
+                uint32_t    loopEnd;
+                int         loopCount;
     volatile    union {
                     uint16_t    volume[2];
                     uint32_t    volumeLR;
@@ -58,15 +70,16 @@ struct audio_track_cblk_t
                 // NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
                 // 8 bit PCM data: in this case,  mCblk->frameSize is based on a sample size of
                 // 16 bit because data is converted to 16 bit before being stored in buffer
-                uint32_t    frameSize;
+
+                uint8_t     frameSize;
                 uint8_t     channelCount;
-                uint8_t     flowControlFlag; // underrun (out) or overrrun (in) indication
-                uint8_t     out;        // out equals 1 for AudioTrack and 0 for AudioRecord
-                uint8_t     forceReady;
+                uint16_t    flags;
+
                 uint16_t    bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
                 uint16_t    waitTimeMs;      // Cumulated wait time
-                // Cache line boundary (32 bytes)
 
+                uint32_t    reserved;
+                // Cache line boundary (32 bytes)
                             audio_track_cblk_t();
                 uint32_t    stepUser(uint32_t frameCount);
                 bool        stepServer(uint32_t frameCount);