Use audio_channel_mask_t more consistently

In IAudioFlinger::createTrack() and IAudioFlinger::openRecord(),
declare input parameter to use correct type audio_channel_mask_t.

In IAudioFlinger::getInputBufferSize(), input parameter is now channel mask
instead of channel count.

Remove unused IAudioFlinger::channelCount(audio_io_handle_t).

In AudioRecord::getMinFrameCount() and AudioSystem::getInputBufferSize(),
input parameter is channel mask instead of channel count.

Change-Id: Ib2f1c29bea70f016b3cfce83942ba292190ac965

4 files changed, 27 insertions, 21 deletions

diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 0f17bbd..f8813c9 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -39,18 +39,18 @@ status_t AudioRecord::getMinFrameCount(
         int* frameCount,
         uint32_t sampleRate,
         audio_format_t format,
-        int channelCount)
+        audio_channel_mask_t channelMask)
 {
     size_t size = 0;
-    if (AudioSystem::getInputBufferSize(sampleRate, format, channelCount, &size)
+    if (AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size)
             != NO_ERROR) {
         ALOGE("AudioSystem could not query the input buffer size.");
         return NO_INIT;
     }
 
     if (size == 0) {
-        ALOGE("Unsupported configuration: sampleRate %d, format %d, channelCount %d",
-            sampleRate, format, channelCount);
+        ALOGE("Unsupported configuration: sampleRate %d, format %d, channelMask %#x",
+            sampleRate, format, channelMask);
         return BAD_VALUE;
     }
 
@@ -58,6 +58,7 @@ status_t AudioRecord::getMinFrameCount(
     size <<= 1;
 
     if (audio_is_linear_pcm(format)) {
+        int channelCount = popcount(channelMask);
         size /= channelCount * audio_bytes_per_sample(format);
     }
 
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 7c29c87..9c270c8 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -41,7 +41,7 @@ DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSyst
 // Cached values for recording queries, all protected by gLock
 uint32_t AudioSystem::gPrevInSamplingRate = 16000;
 audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
-int AudioSystem::gPrevInChannelCount = 1;
+audio_channel_mask_t AudioSystem::gPrevInChannelMask = AUDIO_CHANNEL_IN_MONO;
 size_t AudioSystem::gInBuffSize = 0;
 
 
@@ -334,25 +334,25 @@ status_t AudioSystem::getLatency(audio_io_handle_t output,
     return NO_ERROR;
 }
 
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
-    size_t* buffSize)
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+        audio_channel_mask_t channelMask, size_t* buffSize)
 {
     gLock.lock();
     // Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
     size_t inBuffSize = gInBuffSize;
     if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
-        || (channelCount != gPrevInChannelCount)) {
+        || (channelMask != gPrevInChannelMask)) {
         gLock.unlock();
         const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
         if (af == 0) {
             return PERMISSION_DENIED;
         }
-        inBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+        inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
         gLock.lock();
         // save the request params
         gPrevInSamplingRate = sampleRate;
         gPrevInFormat = format;
-        gPrevInChannelCount = channelCount;
+        gPrevInChannelMask = channelMask;
 
         gInBuffSize = inBuffSize;
     }
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index e8dd438..27f6b45 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -32,7 +32,7 @@ enum {
     CREATE_TRACK = IBinder::FIRST_CALL_TRANSACTION,
     OPEN_RECORD,
     SAMPLE_RATE,
-    CHANNEL_COUNT,
+    CHANNEL_COUNT,  // obsolete
     FORMAT,
     FRAME_COUNT,
     LATENCY,
@@ -86,7 +86,7 @@ public:
                                 audio_stream_type_t streamType,
                                 uint32_t sampleRate,
                                 audio_format_t format,
-                                uint32_t channelMask,
+                                audio_channel_mask_t channelMask,
                                 int frameCount,
                                 track_flags_t flags,
                                 const sp<IMemory>& sharedBuffer,
@@ -182,6 +182,7 @@ public:
         return reply.readInt32();
     }
 
+#if 0
     virtual int channelCount(audio_io_handle_t output) const
     {
         Parcel data, reply;
@@ -190,6 +191,7 @@ public:
         remote()->transact(CHANNEL_COUNT, data, &reply);
         return reply.readInt32();
     }
+#endif
 
     virtual audio_format_t format(audio_io_handle_t output) const
     {
@@ -347,13 +349,14 @@ public:
         remote()->transact(REGISTER_CLIENT, data, &reply);
     }
 
-    virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
+    virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format,
+            audio_channel_mask_t channelMask) const
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.writeInt32(sampleRate);
         data.writeInt32(format);
-        data.writeInt32(channelCount);
+        data.writeInt32(channelMask);
         remote()->transact(GET_INPUTBUFFERSIZE, data, &reply);
         return reply.readInt32();
     }
@@ -698,7 +701,7 @@ status_t BnAudioFlinger::onTransact(
             int streamType = data.readInt32();
             uint32_t sampleRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
-            int channelCount = data.readInt32();
+            audio_channel_mask_t channelMask = data.readInt32();
             size_t bufferCount = data.readInt32();
             track_flags_t flags = (track_flags_t) data.readInt32();
             sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
@@ -708,7 +711,7 @@ status_t BnAudioFlinger::onTransact(
             status_t status;
             sp<IAudioTrack> track = createTrack(pid,
                     (audio_stream_type_t) streamType, sampleRate, format,
-                    channelCount, bufferCount, flags, buffer, output, tid, &sessionId, &status);
+                    channelMask, bufferCount, flags, buffer, output, tid, &sessionId, &status);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(track->asBinder());
@@ -720,13 +723,13 @@ status_t BnAudioFlinger::onTransact(
             audio_io_handle_t input = (audio_io_handle_t) data.readInt32();
             uint32_t sampleRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
-            int channelCount = data.readInt32();
+            audio_channel_mask_t channelMask = data.readInt32();
             size_t bufferCount = data.readInt32();
             track_flags_t flags = (track_flags_t) data.readInt32();
             int sessionId = data.readInt32();
             status_t status;
             sp<IAudioRecord> record = openRecord(pid, input,
-                    sampleRate, format, channelCount, bufferCount, flags, &sessionId, &status);
+                    sampleRate, format, channelMask, bufferCount, flags, &sessionId, &status);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(record->asBinder());
@@ -737,11 +740,13 @@ status_t BnAudioFlinger::onTransact(
             reply->writeInt32( sampleRate((audio_io_handle_t) data.readInt32()) );
             return NO_ERROR;
         } break;
+#if 0
         case CHANNEL_COUNT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             reply->writeInt32( channelCount((audio_io_handle_t) data.readInt32()) );
             return NO_ERROR;
         } break;
+#endif
         case FORMAT: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             reply->writeInt32( format((audio_io_handle_t) data.readInt32()) );
@@ -846,8 +851,8 @@ status_t BnAudioFlinger::onTransact(
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             uint32_t sampleRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
-            int channelCount = data.readInt32();
-            reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
+            audio_channel_mask_t channelMask = data.readInt32();
+            reply->writeInt32( getInputBufferSize(sampleRate, format, channelMask) );
             return NO_ERROR;
         } break;
         case OPEN_OUTPUT: {
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 572927a..ed142a4 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -62,7 +62,7 @@ AudioSource::AudioSource(
     status_t status = AudioRecord::getMinFrameCount(&minFrameCount,
                                            sampleRate,
                                            AUDIO_FORMAT_PCM_16_BIT,
-                                           channelCount);
+                                           audio_channel_in_mask_from_count(channelCount));
     if (status == OK) {
         // make sure that the AudioRecord callback never returns more than the maximum
         // buffer size