Use audio_format_t consistently, continued
Was int or uint32_t.
When AudioFlinger::format can't determine the correct format,
return INVALID rather than DEFAULT.
Init mFormat to INVALID rather than DEFAULT in the constructor.
Subclass constructors will set mFormat to the correct value.
Change-Id: I9b62640aa107d24d2d27925f5563d0d7407d1b73
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index eef551c..0d442ef 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -84,7 +84,7 @@
pid_t pid,
audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -131,7 +131,7 @@
pid_t pid,
int input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -188,13 +188,13 @@
return reply.readInt32();
}
- virtual uint32_t format(int output) const
+ virtual audio_format_t format(int output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(output);
remote()->transact(FORMAT, data, &reply);
- return reply.readInt32();
+ return (audio_format_t) reply.readInt32();
}
virtual size_t frameCount(int output) const
@@ -343,7 +343,7 @@
remote()->transact(REGISTER_CLIENT, data, &reply);
}
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -356,7 +356,7 @@
virtual int openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags)
@@ -364,7 +364,7 @@
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
@@ -382,7 +382,7 @@
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -430,14 +430,14 @@
virtual int openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t acoustics)
{
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -452,7 +452,7 @@
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -678,7 +678,7 @@
pid_t pid = data.readInt32();
int streamType = data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
@@ -699,7 +699,7 @@
pid_t pid = data.readInt32();
int input = data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
@@ -825,7 +825,7 @@
case GET_INPUTBUFFERSIZE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
return NO_ERROR;
@@ -834,7 +834,7 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
uint32_t latency = data.readInt32();
uint32_t flags = data.readInt32();
@@ -879,7 +879,7 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
uint32_t acoutics = data.readInt32();