Fix issue 2203561: Sholes: audio playing out of earpiece.

Create a new IAudioTrack interface to AudioFlinger when start() fails due to a broken pipe error.
Do the same if start fails due to the same error after time out in obtainBuffer().
Do not indicate that the AudioTrack is started to AudioPolicyManager if IAudioTrack start fails.
This avoids that an AudioTrack keeps a dead IAudioTrack after a media server crash.

Same modifications for AudioRecord.

Add a flag to ToneGenerator indicating that the callback thread can call Java. Without it, when the media server crashes and restarts, the AudioSystem error callback will crash in JNI if the IAudiotrack is created from AudioTrack callback thread.

1 files changed, 103 insertions, 60 deletions

diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 4b9d272..8529a8e 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -122,11 +122,6 @@ status_t AudioTrack::set(
         return INVALID_OPERATION;
     }
 
-    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
-    if (audioFlinger == 0) {
-       LOGE("Could not get audioflinger");
-       return NO_INIT;
-    }
     int afSampleRate;
     if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
         return NO_INIT;
@@ -217,28 +212,16 @@ status_t AudioTrack::set(
         }
     }
 
-    // create the track
-    status_t status;
-    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
-                                                      streamType,
-                                                      sampleRate,
-                                                      format,
-                                                      channelCount,
-                                                      frameCount,
-                                                      ((uint16_t)flags) << 16,
-                                                      sharedBuffer,
-                                                      output,
-                                                      &status);
+    mVolume[LEFT] = 1.0f;
+    mVolume[RIGHT] = 1.0f;
+    // create the IAudioTrack
+    status_t status = createTrack(streamType, sampleRate, format, channelCount,
+                                  frameCount, flags, sharedBuffer, output);
 
-    if (track == 0) {
-        LOGE("AudioFlinger could not create track, status: %d", status);
+    if (status != NO_ERROR) {
         return status;
     }
-    sp<IMemory> cblk = track->getCblk();
-    if (cblk == 0) {
-        LOGE("Could not get control block");
-        return NO_INIT;
-    }
+
     if (cbf != 0) {
         mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
         if (mAudioTrackThread == 0) {
@@ -249,22 +232,6 @@ status_t AudioTrack::set(
 
     mStatus = NO_ERROR;
 
-    mAudioTrack = track;
-    mCblkMemory = cblk;
-    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
-    mCblk->out = 1;
-    // Update buffer size in case it has been limited by AudioFlinger during track creation
-    mFrameCount = mCblk->frameCount;
-    if (sharedBuffer == 0) {
-        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
-    } else {
-        mCblk->buffers = sharedBuffer->pointer();
-         // Force buffer full condition as data is already present in shared memory
-        mCblk->stepUser(mFrameCount);
-    }
-    mCblk->volume[0] = mCblk->volume[1] = 0x1000;
-    mVolume[LEFT] = 1.0f;
-    mVolume[RIGHT] = 1.0f;
     mStreamType = streamType;
     mFormat = format;
     mChannels = channels;
@@ -351,16 +318,27 @@ void AudioTrack::start()
      }
 
     if (android_atomic_or(1, &mActive) == 0) {
-        AudioSystem::startOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
-        mNewPosition = mCblk->server + mUpdatePeriod;
-        mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
-        mCblk->waitTimeMs = 0;
-        if (t != 0) {
-           t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
+        audio_io_handle_t output = AudioTrack::getOutput();
+        status_t status = mAudioTrack->start();
+        if (status == DEAD_OBJECT) {
+            LOGV("start() dead IAudioTrack: creating a new one");
+            status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
+                                 mFrameCount, mFlags, mSharedBuffer, output);
+        }
+        if (status == NO_ERROR) {
+            AudioSystem::startOutput(output, (AudioSystem::stream_type)mStreamType);
+            mNewPosition = mCblk->server + mUpdatePeriod;
+            mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
+            mCblk->waitTimeMs = 0;
+            if (t != 0) {
+               t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
+            } else {
+                setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+            }
         } else {
-            setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+            LOGV("start() failed");
+            android_atomic_and(~1, &mActive);
         }
-        mAudioTrack->start();
     }
 
     if (t != 0) {
@@ -617,10 +595,67 @@ audio_io_handle_t AudioTrack::getOutput()
 
 // -------------------------------------------------------------------------
 
+status_t AudioTrack::createTrack(
+        int streamType,
+        uint32_t sampleRate,
+        int format,
+        int channelCount,
+        int frameCount,
+        uint32_t flags,
+        const sp<IMemory>& sharedBuffer,
+        audio_io_handle_t output)
+{
+    status_t status;
+    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
+    if (audioFlinger == 0) {
+       LOGE("Could not get audioflinger");
+       return NO_INIT;
+    }
+
+    sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
+                                                      streamType,
+                                                      sampleRate,
+                                                      format,
+                                                      channelCount,
+                                                      frameCount,
+                                                      ((uint16_t)flags) << 16,
+                                                      sharedBuffer,
+                                                      output,
+                                                      &status);
+
+    if (track == 0) {
+        LOGE("AudioFlinger could not create track, status: %d", status);
+        return status;
+    }
+    sp<IMemory> cblk = track->getCblk();
+    if (cblk == 0) {
+        LOGE("Could not get control block");
+        return NO_INIT;
+    }
+    mAudioTrack.clear();
+    mAudioTrack = track;
+    mCblkMemory.clear();
+    mCblkMemory = cblk;
+    mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
+    mCblk->out = 1;
+    // Update buffer size in case it has been limited by AudioFlinger during track creation
+    mFrameCount = mCblk->frameCount;
+    if (sharedBuffer == 0) {
+        mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+    } else {
+        mCblk->buffers = sharedBuffer->pointer();
+         // Force buffer full condition as data is already present in shared memory
+        mCblk->stepUser(mFrameCount);
+    }
+
+    mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
+
+    return NO_ERROR;
+}
+
 status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
 {
     int active;
-    int timeout = 0;
     status_t result;
     audio_track_cblk_t* cblk = mCblk;
     uint32_t framesReq = audioBuffer->frameCount;
@@ -632,17 +667,20 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
     uint32_t framesAvail = cblk->framesAvailable();
 
     if (framesAvail == 0) {
-        Mutex::Autolock _l(cblk->lock);
+        cblk->lock.lock();
         goto start_loop_here;
         while (framesAvail == 0) {
             active = mActive;
             if (UNLIKELY(!active)) {
                 LOGV("Not active and NO_MORE_BUFFERS");
+                cblk->lock.unlock();
                 return NO_MORE_BUFFERS;
             }
-            if (UNLIKELY(!waitCount))
+            if (UNLIKELY(!waitCount)) {
+                cblk->lock.unlock();
                 return WOULD_BLOCK;
-            timeout = 0;
+            }
+
             result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
             if (__builtin_expect(result!=NO_ERROR, false)) {
                 cblk->waitTimeMs += waitTimeMs;
@@ -654,14 +692,23 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
                                 "user=%08x, server=%08x", this, cblk->user, cblk->server);
                         //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
                         cblk->lock.unlock();
-                        mAudioTrack->start();
+                        result = mAudioTrack->start();
+                        if (result == DEAD_OBJECT) {
+                            LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
+                            result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
+                                                 mFrameCount, mFlags, mSharedBuffer, getOutput());
+                            if (result == NO_ERROR) {
+                                cblk = mCblk;
+                                cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+                            }
+                        }
                         cblk->lock.lock();
-                        timeout = 1;
                     }
                     cblk->waitTimeMs = 0;
                 }
 
                 if (--waitCount == 0) {
+                    cblk->lock.unlock();
                     return TIMED_OUT;
                 }
             }
@@ -669,6 +716,7 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
         start_loop_here:
             framesAvail = cblk->framesAvailable_l();
         }
+        cblk->lock.unlock();
     }
 
     cblk->waitTimeMs = 0;
@@ -684,11 +732,6 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
         framesReq = bufferEnd - u;
     }
 
-    LOGW_IF(timeout,
-        "*** SERIOUS WARNING *** obtainBuffer() timed out "
-        "but didn't need to be locked. We recovered, but "
-        "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
-
     audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
     audioBuffer->channelCount = mChannelCount;
     audioBuffer->frameCount = framesReq;
@@ -991,7 +1034,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
         // Mark that we have read the first buffer so that next time stepUser() is called
         // we switch to normal obtainBuffer() timeout period
         if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
-            bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1;
+            bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
         }
         // It is possible that we receive a flush()
         // while the mixer is processing a block: in this case,