When passing a size_t via binder, use 64-bits unconditionally
64-bits is almost always over-kill. But it's easier and cleaner to change
the binder code to be accurate, than to rename all the the API parameter
types to be 32-bit.
Bug: 12381724
Change-Id: Ib8f198d814a2027760ef24e9e3feacee21a973b1
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 86ff8bd..e02107f 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -106,7 +106,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
- data.writeInt32(frameCount);
+ data.writeInt64(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
if (sharedBuffer != 0) {
@@ -163,7 +163,7 @@
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
- data.writeInt32(frameCount);
+ data.writeInt64(frameCount);
track_flags_t lFlags = flags != NULL ? *flags : (track_flags_t) TRACK_DEFAULT;
data.writeInt32(lFlags);
data.writeInt32((int32_t) tid);
@@ -228,7 +228,7 @@
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32((int32_t) output);
remote()->transact(FRAME_COUNT, data, &reply);
- return reply.readInt32();
+ return reply.readInt64();
}
virtual uint32_t latency(audio_io_handle_t output) const
@@ -378,7 +378,7 @@
data.writeInt32(format);
data.writeInt32(channelMask);
remote()->transact(GET_INPUTBUFFERSIZE, data, &reply);
- return reply.readInt32();
+ return reply.readInt64();
}
virtual audio_io_handle_t openOutput(audio_module_handle_t module,
@@ -726,7 +726,7 @@
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
remote()->transact(GET_PRIMARY_OUTPUT_FRAME_COUNT, data, &reply);
- return reply.readInt32();
+ return reply.readInt64();
}
virtual status_t setLowRamDevice(bool isLowRamDevice)
@@ -754,7 +754,7 @@
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
- size_t frameCount = data.readInt32();
+ size_t frameCount = data.readInt64();
track_flags_t flags = (track_flags_t) data.readInt32();
bool haveSharedBuffer = data.readInt32() != 0;
sp<IMemory> buffer;
@@ -791,7 +791,7 @@
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
- size_t frameCount = data.readInt32();
+ size_t frameCount = data.readInt64();
track_flags_t flags = (track_flags_t) data.readInt32();
pid_t tid = (pid_t) data.readInt32();
int sessionId = data.readInt32();
@@ -817,7 +817,7 @@
} break;
case FRAME_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( frameCount((audio_io_handle_t) data.readInt32()) );
+ reply->writeInt64( frameCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case LATENCY: {
@@ -916,7 +916,7 @@
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
audio_channel_mask_t channelMask = data.readInt32();
- reply->writeInt32( getInputBufferSize(sampleRate, format, channelMask) );
+ reply->writeInt64( getInputBufferSize(sampleRate, format, channelMask) );
return NO_ERROR;
} break;
case OPEN_OUTPUT: {
@@ -1119,7 +1119,7 @@
} break;
case GET_PRIMARY_OUTPUT_FRAME_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(getPrimaryOutputFrameCount());
+ reply->writeInt64(getPrimaryOutputFrameCount());
return NO_ERROR;
} break;
case SET_LOW_RAM_DEVICE: {