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

12 files changed, 96 insertions, 96 deletions

diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 44925f2..756e91d 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -68,7 +68,7 @@ public:
         };
         uint32_t    flags;
         int         channelCount;
-        int         format;
+        audio_format_t format;
         size_t      frameCount;
         size_t      size;
         union {
@@ -112,7 +112,7 @@ public:
 
      static status_t getMinFrameCount(int* frameCount,
                                       uint32_t sampleRate,
-                                      int format,
+                                      audio_format_t format,
                                       int channelCount);
 
     /* Constructs an uninitialized AudioRecord. No connection with
@@ -151,7 +151,7 @@ public:
 
                         AudioRecord(int inputSource,
                                     uint32_t sampleRate = 0,
-                                    int format          = 0,
+                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                     uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
                                     int frameCount      = 0,
                                     uint32_t flags      = 0,
@@ -177,7 +177,7 @@ public:
      * */
             status_t    set(int inputSource     = 0,
                             uint32_t sampleRate = 0,
-                            int format          = 0,
+                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
                             uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
                             int frameCount      = 0,
                             uint32_t flags      = 0,
@@ -203,7 +203,7 @@ public:
 
    /* getters, see constructor */
 
-            int         format() const;
+            audio_format_t format() const;
             int         channelCount() const;
             int         channels() const;
             uint32_t    frameCount() const;
@@ -349,7 +349,7 @@ private:
 
             bool processAudioBuffer(const sp<ClientRecordThread>& thread);
             status_t openRecord_l(uint32_t sampleRate,
-                                uint32_t format,
+                                audio_format_t format,
                                 uint32_t channelMask,
                                 int frameCount,
                                 uint32_t flags,
@@ -365,7 +365,7 @@ private:
     uint32_t                mFrameCount;
 
     audio_track_cblk_t*     mCblk;
-    uint32_t                mFormat;
+    audio_format_t          mFormat;
     uint8_t                 mChannelCount;
     uint8_t                 mInputSource;
     uint8_t                 mReserved[2];
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 49e5690..c6368fb 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -95,7 +95,7 @@ public:
 
     static bool routedToA2dpOutput(audio_stream_type_t streamType);
 
-    static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
+    static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
         size_t* buffSize);
 
     static status_t setVoiceVolume(float volume);
@@ -134,7 +134,7 @@ public:
     class OutputDescriptor {
     public:
         OutputDescriptor()
-        : samplingRate(0), format(0), channels(0), frameCount(0), latency(0)  {}
+        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0)  {}
 
         uint32_t samplingRate;
         int32_t format;
@@ -153,7 +153,7 @@ public:
     static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
     static audio_io_handle_t getOutput(audio_stream_type_t stream,
                                         uint32_t samplingRate = 0,
-                                        uint32_t format = AUDIO_FORMAT_DEFAULT,
+                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         uint32_t channels = AUDIO_CHANNEL_OUT_STEREO,
                                         audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);
     static status_t startOutput(audio_io_handle_t output,
@@ -165,7 +165,7 @@ public:
     static void releaseOutput(audio_io_handle_t output);
     static audio_io_handle_t getInput(int inputSource,
                                     uint32_t samplingRate = 0,
-                                    uint32_t format = AUDIO_FORMAT_DEFAULT,
+                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                     uint32_t channels = AUDIO_CHANNEL_IN_MONO,
                                     audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0,
                                     int sessionId = 0);
@@ -242,7 +242,7 @@ private:
     static size_t gInBuffSize;
     // previous parameters for recording buffer size queries
     static uint32_t gPrevInSamplingRate;
-    static int gPrevInFormat;
+    static audio_format_t gPrevInFormat;
     static int gPrevInChannelCount;
 
     static sp<IAudioPolicyService> gAudioPolicyService;
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 3999479..7c0d886 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -48,7 +48,7 @@ public:
                                 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,
@@ -61,7 +61,7 @@ public:
                                 pid_t pid,
                                 int input,
                                 uint32_t sampleRate,
-                                uint32_t format,
+                                audio_format_t format,
                                 uint32_t channelMask,
                                 int frameCount,
                                 uint32_t flags,
@@ -73,7 +73,7 @@ public:
      */
     virtual     uint32_t    sampleRate(int output) const = 0;
     virtual     int         channelCount(int output) const = 0;
-    virtual     uint32_t    format(int output) const = 0;
+    virtual     audio_format_t format(int output) const = 0;
     virtual     size_t      frameCount(int output) const = 0;
     virtual     uint32_t    latency(int output) const = 0;
 
@@ -109,11 +109,11 @@ public:
     virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
 
     // retrieve the audio recording buffer size
-    virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
+    virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) = 0;
 
     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) = 0;
@@ -124,7 +124,7 @@ public:
 
     virtual int openInput(uint32_t *pDevices,
                                     uint32_t *pSamplingRate,
-                                    uint32_t *pFormat,
+                                    audio_format_t *pFormat,
                                     uint32_t *pChannels,
                                     uint32_t acoustics) = 0;
     virtual status_t closeInput(int input) = 0;
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index 8ddbe0a..07d17c5 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -50,7 +50,7 @@ public:
     virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
     virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
                                         uint32_t samplingRate = 0,
-                                        uint32_t format = AUDIO_FORMAT_DEFAULT,
+                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         uint32_t channels = 0,
                                         audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT) = 0;
     virtual status_t startOutput(audio_io_handle_t output,
@@ -62,7 +62,7 @@ public:
     virtual void releaseOutput(audio_io_handle_t output) = 0;
     virtual audio_io_handle_t getInput(int inputSource,
                                     uint32_t samplingRate = 0,
-                                    uint32_t format = AUDIO_FORMAT_DEFAULT,
+                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                     uint32_t channels = 0,
                                     audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0,
                                     int audioSession = 0) = 0;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 32b5bac..5b5b076 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -48,7 +48,7 @@ namespace android {
 status_t AudioRecord::getMinFrameCount(
         int* frameCount,
         uint32_t sampleRate,
-        int format,
+        audio_format_t format,
         int channelCount)
 {
     size_t size = 0;
@@ -86,7 +86,7 @@ AudioRecord::AudioRecord()
 AudioRecord::AudioRecord(
         int inputSource,
         uint32_t sampleRate,
-        int format,
+        audio_format_t format,
         uint32_t channelMask,
         int frameCount,
         uint32_t flags,
@@ -121,7 +121,7 @@ AudioRecord::~AudioRecord()
 status_t AudioRecord::set(
         int inputSource,
         uint32_t sampleRate,
-        int format,
+        audio_format_t format,
         uint32_t channelMask,
         int frameCount,
         uint32_t flags,
@@ -148,7 +148,7 @@ status_t AudioRecord::set(
         sampleRate = DEFAULT_SAMPLE_RATE;
     }
     // these below should probably come from the audioFlinger too...
-    if (format == 0) {
+    if (format == AUDIO_FORMAT_DEFAULT) {
         format = AUDIO_FORMAT_PCM_16_BIT;
     }
     // validate parameters
@@ -248,7 +248,7 @@ uint32_t AudioRecord::latency() const
     return mLatency;
 }
 
-int AudioRecord::format() const
+audio_format_t AudioRecord::format() const
 {
     return mFormat;
 }
@@ -448,7 +448,7 @@ unsigned int AudioRecord::getInputFramesLost()
 // must be called with mLock held
 status_t AudioRecord::openRecord_l(
         uint32_t sampleRate,
-        uint32_t format,
+        audio_format_t format,
         uint32_t channelMask,
         int frameCount,
         uint32_t flags,
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 5ca868a..952d634 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -40,7 +40,7 @@ DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSyst
 
 // Cached values for recording queries, all protected by gLock
 uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-int AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
+audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
 int AudioSystem::gPrevInChannelCount = 1;
 size_t AudioSystem::gInBuffSize = 0;
 
@@ -308,7 +308,7 @@ status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t st
     return NO_ERROR;
 }
 
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
     size_t* buffSize)
 {
     gLock.lock();
@@ -572,7 +572,7 @@ audio_policy_forced_cfg_t AudioSystem::getForceUse(audio_policy_force_use_t usag
 
 audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
                                     uint32_t samplingRate,
-                                    uint32_t format,
+                                    audio_format_t format,
                                     uint32_t channels,
                                     audio_policy_output_flags_t flags)
 {
@@ -632,7 +632,7 @@ void AudioSystem::releaseOutput(audio_io_handle_t output)
 
 audio_io_handle_t AudioSystem::getInput(int inputSource,
                                     uint32_t samplingRate,
-                                    uint32_t format,
+                                    audio_format_t format,
                                     uint32_t channels,
                                     audio_in_acoustics_t acoustics,
                                     int sessionId)
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 @@ public:
                                 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 @@ public:
                                 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 @@ public:
         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 @@ public:
         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 @@ public:
 
     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 @@ public:
         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 @@ public:
         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 @@ public:
 
     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 @@ public:
         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 @@ status_t BnAudioFlinger::onTransact(
             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 @@ status_t BnAudioFlinger::onTransact(
             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 @@ status_t BnAudioFlinger::onTransact(
         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 @@ status_t BnAudioFlinger::onTransact(
             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 @@ status_t BnAudioFlinger::onTransact(
             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();
 
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index e363101..b5c857f 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -122,7 +122,7 @@ public:
     virtual audio_io_handle_t getOutput(
                                         audio_stream_type_t stream,
                                         uint32_t samplingRate,
-                                        uint32_t format,
+                                        audio_format_t format,
                                         uint32_t channels,
                                         audio_policy_output_flags_t flags)
     {
@@ -174,7 +174,7 @@ public:
     virtual audio_io_handle_t getInput(
                                     int inputSource,
                                     uint32_t samplingRate,
-                                    uint32_t format,
+                                    audio_format_t format,
                                     uint32_t channels,
                                     audio_in_acoustics_t acoustics,
                                     int audioSession)
@@ -416,7 +416,7 @@ status_t BnAudioPolicyService::onTransact(
             audio_stream_type_t stream =
                     static_cast <audio_stream_type_t>(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();
             audio_policy_output_flags_t flags =
                     static_cast <audio_policy_output_flags_t>(data.readInt32());
@@ -463,7 +463,7 @@ status_t BnAudioPolicyService::onTransact(
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             int inputSource = 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();
             audio_in_acoustics_t acoustics =
                     static_cast <audio_in_acoustics_t>(data.readInt32());
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 53b13f7..c6a9c77 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -382,7 +382,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
         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,
@@ -502,13 +502,13 @@ int AudioFlinger::channelCount(int output) const
     return thread->channelCount();
 }
 
-uint32_t AudioFlinger::format(int output) const
+audio_format_t AudioFlinger::format(int output) const
 {
     Mutex::Autolock _l(mLock);
     PlaybackThread *thread = checkPlaybackThread_l(output);
     if (thread == NULL) {
         ALOGW("format() unknown thread %d", output);
-        return 0;
+        return AUDIO_FORMAT_INVALID;
     }
     return thread->format();
 }
@@ -849,7 +849,7 @@ String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
     return String8("");
 }
 
-size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount)
 {
     status_t ret = initCheck();
     if (ret != NO_ERROR) {
@@ -990,7 +990,7 @@ void AudioFlinger::removeClient_l(pid_t pid)
 AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
     :   Thread(false),
         mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
-        mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
+        mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false),
         mDevice(device)
 {
     mDeathRecipient = new PMDeathRecipient(this);
@@ -1033,7 +1033,7 @@ int AudioFlinger::ThreadBase::channelCount() const
     return (int)mChannelCount;
 }
 
-uint32_t AudioFlinger::ThreadBase::format() const
+audio_format_t AudioFlinger::ThreadBase::format() const
 {
     return mFormat;
 }
@@ -1495,7 +1495,7 @@ sp<AudioFlinger::PlaybackThread::Track>  AudioFlinger::PlaybackThread::createTra
         const sp<AudioFlinger::Client>& client,
         audio_stream_type_t streamType,
         uint32_t sampleRate,
-        uint32_t format,
+        audio_format_t format,
         uint32_t channelMask,
         int frameCount,
         const sp<IMemory>& sharedBuffer,
@@ -2394,7 +2394,7 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l()
             reconfig = true;
         }
         if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
-            if (value != AUDIO_FORMAT_PCM_16_BIT) {
+            if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
                 status = BAD_VALUE;
             } else {
                 reconfig = true;
@@ -3233,7 +3233,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
             const wp<ThreadBase>& thread,
             const sp<Client>& client,
             uint32_t sampleRate,
-            uint32_t format,
+            audio_format_t format,
             uint32_t channelMask,
             int frameCount,
             uint32_t flags,
@@ -3395,7 +3395,7 @@ AudioFlinger::PlaybackThread::Track::Track(
             const sp<Client>& client,
             audio_stream_type_t streamType,
             uint32_t sampleRate,
-            uint32_t format,
+            audio_format_t format,
             uint32_t channelMask,
             int frameCount,
             const sp<IMemory>& sharedBuffer,
@@ -3701,7 +3701,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
             const wp<ThreadBase>& thread,
             const sp<Client>& client,
             uint32_t sampleRate,
-            uint32_t format,
+            audio_format_t format,
             uint32_t channelMask,
             int frameCount,
             uint32_t flags,
@@ -3814,7 +3814,7 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
             const wp<ThreadBase>& thread,
             DuplicatingThread *sourceThread,
             uint32_t sampleRate,
-            uint32_t format,
+            audio_format_t format,
             uint32_t channelMask,
             int frameCount)
     :   Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
@@ -4147,7 +4147,7 @@ sp<IAudioRecord> AudioFlinger::openRecord(
         pid_t pid,
         int input,
         uint32_t sampleRate,
-        uint32_t format,
+        audio_format_t format,
         uint32_t channelMask,
         int frameCount,
         uint32_t flags,
@@ -4492,7 +4492,7 @@ bool AudioFlinger::RecordThread::threadLoop()
 sp<AudioFlinger::RecordThread::RecordTrack>  AudioFlinger::RecordThread::createRecordTrack_l(
         const sp<AudioFlinger::Client>& client,
         uint32_t sampleRate,
-        int format,
+        audio_format_t format,
         int channelMask,
         int frameCount,
         uint32_t flags,
@@ -4704,7 +4704,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
         String8 keyValuePair = mNewParameters[0];
         AudioParameter param = AudioParameter(keyValuePair);
         int value;
-        int reqFormat = mFormat;
+        audio_format_t reqFormat = mFormat;
         int reqSamplingRate = mReqSampleRate;
         int reqChannelCount = mReqChannelCount;
 
@@ -4713,7 +4713,7 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
             reconfig = true;
         }
         if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
-            reqFormat = value;
+            reqFormat = (audio_format_t) value;
             reconfig = true;
         }
         if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
@@ -4924,7 +4924,7 @@ audio_stream_t* AudioFlinger::RecordThread::stream()
 
 int AudioFlinger::openOutput(uint32_t *pDevices,
                                 uint32_t *pSamplingRate,
-                                uint32_t *pFormat,
+                                audio_format_t *pFormat,
                                 uint32_t *pChannels,
                                 uint32_t *pLatencyMs,
                                 uint32_t flags)
@@ -4933,7 +4933,7 @@ int AudioFlinger::openOutput(uint32_t *pDevices,
     PlaybackThread *thread = NULL;
     mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
     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;
     audio_stream_out_t *outStream;
@@ -4956,7 +4956,7 @@ int AudioFlinger::openOutput(uint32_t *pDevices,
     if (outHwDev == NULL)
         return 0;
 
-    status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
+    status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
                                           &channels, &samplingRate, &outStream);
     ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
             outStream,
@@ -5084,17 +5084,17 @@ status_t AudioFlinger::restoreOutput(int output)
 
 int AudioFlinger::openInput(uint32_t *pDevices,
                                 uint32_t *pSamplingRate,
-                                uint32_t *pFormat,
+                                audio_format_t *pFormat,
                                 uint32_t *pChannels,
                                 uint32_t acoustics)
 {
     status_t status;
     RecordThread *thread = NULL;
     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 reqSamplingRate = samplingRate;
-    uint32_t reqFormat = format;
+    audio_format_t reqFormat = format;
     uint32_t reqChannels = channels;
     audio_stream_in_t *inStream;
     audio_hw_device_t *inHwDev;
@@ -5109,7 +5109,7 @@ int AudioFlinger::openInput(uint32_t *pDevices,
     if (inHwDev == NULL)
         return 0;
 
-    status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
+    status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
                                         &channels, &samplingRate,
                                         (audio_in_acoustics_t)acoustics,
                                         &inStream);
@@ -5129,7 +5129,7 @@ int AudioFlinger::openInput(uint32_t *pDevices,
         (samplingRate <= 2 * reqSamplingRate) &&
         (popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
         ALOGV("openInput() reopening with proposed sampling rate and channels");
-        status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
+        status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
                                             &channels, &samplingRate,
                                             (audio_in_acoustics_t)acoustics,
                                             &inStream);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 8a82bdb..d862c1d 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -72,7 +72,7 @@ public:
                                 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,
@@ -83,7 +83,7 @@ public:
 
     virtual     uint32_t    sampleRate(int output) const;
     virtual     int         channelCount(int output) const;
-    virtual     uint32_t    format(int output) const;
+    virtual     audio_format_t format(int output) const;
     virtual     size_t      frameCount(int output) const;
     virtual     uint32_t    latency(int output) const;
 
@@ -109,12 +109,12 @@ public:
 
     virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
 
-    virtual     size_t      getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
+    virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount);
     virtual     unsigned int  getInputFramesLost(int ioHandle);
 
     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);
@@ -129,7 +129,7 @@ public:
 
     virtual int openInput(uint32_t *pDevices,
                             uint32_t *pSamplingRate,
-                            uint32_t *pFormat,
+                            audio_format_t *pFormat,
                             uint32_t *pChannels,
                             uint32_t acoustics);
 
@@ -189,7 +189,7 @@ public:
                                 pid_t pid,
                                 int input,
                                 uint32_t sampleRate,
-                                uint32_t format,
+                                audio_format_t format,
                                 uint32_t channelMask,
                                 int frameCount,
                                 uint32_t flags,
@@ -315,7 +315,7 @@ private:
                                 TrackBase(const wp<ThreadBase>& thread,
                                         const sp<Client>& client,
                                         uint32_t sampleRate,
-                                        uint32_t format,
+                                        audio_format_t format,
                                         uint32_t channelMask,
                                         int frameCount,
                                         uint32_t flags,
@@ -343,7 +343,7 @@ private:
             virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
             virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
 
-            uint32_t format() const {
+            audio_format_t format() const {
                 return mFormat;
             }
 
@@ -376,7 +376,7 @@ private:
             // we don't really need a lock for these
             int                 mState;
             int                 mClientTid;
-            uint32_t            mFormat;
+            audio_format_t      mFormat;
             uint32_t            mFlags;
             int                 mSessionId;
             uint8_t             mChannelCount;
@@ -410,7 +410,7 @@ private:
                     int         type() const { return mType; }
                     uint32_t    sampleRate() const;
                     int         channelCount() const;
-                    uint32_t    format() const;
+                    audio_format_t format() const;
                     size_t      frameCount() const;
                     void        wakeUp()    { mWaitWorkCV.broadcast(); }
                     void        exit();
@@ -537,7 +537,7 @@ private:
                     uint32_t                mChannelMask;
                     uint16_t                mChannelCount;
                     size_t                  mFrameSize;
-                    uint32_t                mFormat;
+                    audio_format_t          mFormat;
                     Condition               mParamCond;
                     Vector<String8>         mNewParameters;
                     status_t                mParamStatus;
@@ -575,7 +575,7 @@ private:
                                         const sp<Client>& client,
                                         audio_stream_type_t streamType,
                                         uint32_t sampleRate,
-                                        uint32_t format,
+                                        audio_format_t format,
                                         uint32_t channelMask,
                                         int frameCount,
                                         const sp<IMemory>& sharedBuffer,
@@ -660,7 +660,7 @@ private:
                                 OutputTrack(  const wp<ThreadBase>& thread,
                                         DuplicatingThread *sourceThread,
                                         uint32_t sampleRate,
-                                        uint32_t format,
+                                        audio_format_t format,
                                         uint32_t channelMask,
                                         int frameCount);
                                 ~OutputTrack();
@@ -715,7 +715,7 @@ private:
                                     const sp<AudioFlinger::Client>& client,
                                     audio_stream_type_t streamType,
                                     uint32_t sampleRate,
-                                    uint32_t format,
+                                    audio_format_t format,
                                     uint32_t channelMask,
                                     int frameCount,
                                     const sp<IMemory>& sharedBuffer,
@@ -935,7 +935,7 @@ private:
                                 RecordTrack(const wp<ThreadBase>& thread,
                                         const sp<Client>& client,
                                         uint32_t sampleRate,
-                                        uint32_t format,
+                                        audio_format_t format,
                                         uint32_t channelMask,
                                         int frameCount,
                                         uint32_t flags,
@@ -979,7 +979,7 @@ private:
                 sp<AudioFlinger::RecordThread::RecordTrack>  createRecordTrack_l(
                         const sp<AudioFlinger::Client>& client,
                         uint32_t sampleRate,
-                        int format,
+                        audio_format_t format,
                         int channelMask,
                         int frameCount,
                         uint32_t flags,
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 7a408bc..28b1c89 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -241,7 +241,7 @@ audio_policy_forced_cfg_t AudioPolicyService::getForceUse(audio_policy_force_use
 
 audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream,
                                     uint32_t samplingRate,
-                                    uint32_t format,
+                                    audio_format_t format,
                                     uint32_t channels,
                                     audio_policy_output_flags_t flags)
 {
@@ -289,7 +289,7 @@ void AudioPolicyService::releaseOutput(audio_io_handle_t output)
 
 audio_io_handle_t AudioPolicyService::getInput(int inputSource,
                                     uint32_t samplingRate,
-                                    uint32_t format,
+                                    audio_format_t format,
                                     uint32_t channels,
                                     audio_in_acoustics_t acoustics,
                                     int audioSession)
@@ -1352,7 +1352,7 @@ extern "C" {
 static audio_io_handle_t aps_open_output(void *service,
                                              uint32_t *pDevices,
                                              uint32_t *pSamplingRate,
-                                             uint32_t *pFormat,
+                                             audio_format_t *pFormat,
                                              uint32_t *pChannels,
                                              uint32_t *pLatencyMs,
                                              audio_policy_output_flags_t flags)
@@ -1413,7 +1413,7 @@ static int aps_restore_output(void *service, audio_io_handle_t output)
 static audio_io_handle_t aps_open_input(void *service,
                                             uint32_t *pDevices,
                                             uint32_t *pSamplingRate,
-                                            uint32_t *pFormat,
+                                            audio_format_t *pFormat,
                                             uint32_t *pChannels,
                                             uint32_t acoustics)
 {
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 3693cef..9811670 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -63,7 +63,7 @@ public:
     virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
     virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
                                         uint32_t samplingRate = 0,
-                                        uint32_t format = AUDIO_FORMAT_DEFAULT,
+                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                         uint32_t channels = 0,
                                         audio_policy_output_flags_t flags =
                                             AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);
@@ -76,7 +76,7 @@ public:
     virtual void releaseOutput(audio_io_handle_t output);
     virtual audio_io_handle_t getInput(int inputSource,
                                     uint32_t samplingRate = 0,
-                                    uint32_t format = AUDIO_FORMAT_DEFAULT,
+                                    audio_format_t format = AUDIO_FORMAT_DEFAULT,
                                     uint32_t channels = 0,
                                     audio_in_acoustics_t acoustics =
                                             (audio_in_acoustics_t)0,