av: Various fixes and cleanupskitkat

Change-Id: I82225d8cd3e2b3015e6c0e1dd9ddc5ec8abab525

5 files changed, 15 insertions, 6 deletions

diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index ee40689d2e..c1ec876412 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -203,17 +203,21 @@ status_t AudioRecord::set(
         ALOGE("Invalid format %d", format);
         return BAD_VALUE;
     }
-#ifdef QCOM_HARDWARE
+#if defined(QCOM_HARDWARE) && !defined(QCOM_DIRECTTRACK)
     if (format != AUDIO_FORMAT_PCM_16_BIT &&
            !audio_is_compress_voip_format(format) &&
            !audio_is_compress_capture_format(format)) {
 #else
+#ifndef QCOM_DIRECTTRACK
     // Temporary restriction: AudioFlinger currently supports 16-bit PCM only
     if (format != AUDIO_FORMAT_PCM_16_BIT) {
 #endif
+#endif
+#ifndef QCOM_DIRECTTRACK
         ALOGE("Format %d is not supported", format);
         return BAD_VALUE;
     }
+#endif
 
     mFormat = format;
 
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index c94f8724c2..8b97020eae 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -410,6 +410,7 @@ status_t AudioTrack::set(
     }
 #endif
 
+#ifndef QCOM_DIRECTTRACK
     if (audio_is_linear_pcm(format)) {
         mFrameSize = channelCount * audio_bytes_per_sample(format);
         mFrameSizeAF = channelCount * sizeof(int16_t);
@@ -417,6 +418,7 @@ status_t AudioTrack::set(
         mFrameSize = sizeof(uint8_t);
         mFrameSizeAF = sizeof(uint8_t);
     }
+#endif
 
     audio_io_handle_t output = AudioSystem::getOutput(
                                     streamType,
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1e7a8818b7..b8d8aab33f 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1590,11 +1590,11 @@ sp<IAudioRecord> AudioFlinger::openRecord(
         goto Exit;
     }
 #else
-#if defined(QCOM_HARDWARE)
+#ifdef QCOM_HARDWARE
     if (format != AUDIO_FORMAT_PCM_16_BIT &&
             !audio_is_compress_voip_format(format) &&
             !audio_is_compress_capture_format(format)) {
-#else 
+#else
     if (format != AUDIO_FORMAT_PCM_16_BIT) {
 #endif
         ALOGE("openRecord() invalid format %d", format);
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 32efa0affc..a23d031284 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -540,8 +540,9 @@ bool AudioPolicyService::isStreamActiveRemotely(audio_stream_type_t stream, uint
     }
     Mutex::Autolock _l(mLock);
     return mpAudioPolicy->is_stream_active_remotely(mpAudioPolicy, stream, inPastMs);
-#endif
+#else
     return 0;
+#endif
 }
 
 bool AudioPolicyService::isSourceActive(audio_source_t source) const
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index c92eeb257c..f93bd1dada 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -3853,7 +3853,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prep
                     tracksToRemove->add(track);
                     // indicate to client process that the track was disabled because of underrun;
                     // it will then automatically call start() when data is available
-#if defined(QCOM_HARDWARE) && !defined(QCOM_DIRECTTRACK)
+#ifdef QCOM_HARDWARE
                     android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
 #endif
                 } else if (last) {
@@ -5680,17 +5680,19 @@ void AudioFlinger::RecordThread::readInputParameters()
     mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
     mChannelCount = (uint16_t)getInputChannelCount(mChannelMask);
     mFormat = mInput->stream->common.get_format(&mInput->stream->common);
-#ifdef QCOM_HARDWARE
+#if defined(QCOM_HARDWARE) && !defined(QCOM_DIRECTTRACK)
     if (mFormat != AUDIO_FORMAT_PCM_16_BIT &&
             !audio_is_compress_voip_format(mFormat) &&
             !audio_is_compress_capture_format(mFormat)) {
         ALOGE("HAL format %d not supported", mFormat);
     }
 #else
+#ifndef QCOM_DIRECTTRACK
     if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
         ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
     }
 #endif
+#endif
     mFrameSize = audio_stream_frame_size(&mInput->stream->common);
     mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
     mFrameCount = mBufferSize / mFrameSize;