Update for Froyo

Add new input stream method getInputFramesLost().
Add new output stream method getInputFramesLost().

3 files changed, 47 insertions, 2 deletions

diff --git a/AudioHardwareALSA.h b/AudioHardwareALSA.h
index d2a9bec..e08fade 100644
--- a/AudioHardwareALSA.h
+++ b/AudioHardwareALSA.h
@@ -199,8 +199,15 @@ public:
         return ALSAStreamOps::getParameters(keys);
     }
 
+    // return the number of audio frames written by the audio dsp to DAC since
+    // the output has exited standby
+    virtual status_t    getRenderPosition(uint32_t *dspFrames);
+
     status_t            open(int mode);
     status_t            close();
+
+private:
+    uint32_t            mFrameCount;
 };
 
 class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps
@@ -248,12 +255,21 @@ public:
         return ALSAStreamOps::getParameters(keys);
     }
 
+    // Return the amount of input frames lost in the audio driver since the last call of this function.
+    // Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call.
+    // Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers.
+    // Unit: the number of input audio frames
+    virtual unsigned int  getInputFramesLost() const;
+
     status_t            setAcousticParams(void* params);
 
     status_t            open(int mode);
     status_t            close();
 
 private:
+    void                resetFramesLost();
+
+    unsigned int        mFramesLost;
     AudioSystem::audio_in_acoustics mAcoustics;
 };
 
diff --git a/AudioStreamInALSA.cpp b/AudioStreamInALSA.cpp
index 970dafd..32528a7 100644
--- a/AudioStreamInALSA.cpp
+++ b/AudioStreamInALSA.cpp
@@ -40,6 +40,7 @@ AudioStreamInALSA::AudioStreamInALSA(AudioHardwareALSA *parent,
         alsa_handle_t *handle,
         AudioSystem::audio_in_acoustics audio_acoustics) :
     ALSAStreamOps(parent, handle),
+    mFramesLost(0),
     mAcoustics(audio_acoustics)
 {
     acoustic_device_t *aDev = acoustics();
@@ -143,6 +144,21 @@ status_t AudioStreamInALSA::standby()
     return NO_ERROR;
 }
 
+void AudioStreamInALSA::resetFramesLost()
+{
+    AutoMutex lock(mLock);
+    mFramesLost = 0;
+}
+
+unsigned int AudioStreamInALSA::getInputFramesLost() const
+{
+    unsigned int count = mFramesLost;
+    // Stupid interface wants us to have a side effect of clearing the count
+    // but is defined as a const to prevent such a thing.
+    ((AudioStreamInALSA *)this)->resetFramesLost();
+    return count;
+}
+
 status_t AudioStreamInALSA::setAcousticParams(void *params)
 {
     AutoMutex lock(mLock);
diff --git a/AudioStreamOutALSA.cpp b/AudioStreamOutALSA.cpp
index 781afbb..b295676 100644
--- a/AudioStreamOutALSA.cpp
+++ b/AudioStreamOutALSA.cpp
@@ -47,7 +47,8 @@ static const int DEFAULT_SAMPLE_RATE = ALSA_DEFAULT_SAMPLE_RATE;
 // ----------------------------------------------------------------------------
 
 AudioStreamOutALSA::AudioStreamOutALSA(AudioHardwareALSA *parent, alsa_handle_t *handle) :
-    ALSAStreamOps(parent, handle)
+    ALSAStreamOps(parent, handle),
+    mFrameCount(0)
 {
 }
 
@@ -109,8 +110,10 @@ ssize_t AudioStreamOutALSA::write(const void *buffer, size_t bytes)
                 if (n) return static_cast<ssize_t>(n);
             }
         }
-        else
+        else {
+            mFrameCount += n;
             sent += static_cast<ssize_t>(snd_pcm_frames_to_bytes(mHandle->handle, n));
+        }
 
     } while (mHandle->handle && sent < bytes);
 
@@ -155,6 +158,8 @@ status_t AudioStreamOutALSA::standby()
         mPowerLock = false;
     }
 
+    mFrameCount = 0;
+
     return NO_ERROR;
 }
 
@@ -166,4 +171,12 @@ uint32_t AudioStreamOutALSA::latency() const
     return USEC_TO_MSEC (mHandle->latency);
 }
 
+// return the number of audio frames written by the audio dsp to DAC since
+// the output has exited standby
+status_t AudioStreamOutALSA::getRenderPosition(uint32_t *dspFrames)
+{
+    *dspFrames = mFrameCount;
+    return NO_ERROR;
+}
+
 }       // namespace android