Move audio_frame_operations to the utility module.
TBR=henrika@webrtc.org BUG=issue551 TEST=voe_auto_test, webrtc_utility_unittest, trybots Review URL: https://webrtc-codereview.appspot.com/599006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2318 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -13,6 +13,7 @@
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'type': '<(library)',
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'dependencies': [
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'audio_processing',
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'webrtc_utility',
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'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
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],
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'include_dirs': [
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@ -14,7 +14,7 @@
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#include "audio_processing.h"
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#include "critical_section_wrapper.h"
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#include "map_wrapper.h"
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#include "voice_engine/main/source/audio_frame_operations.h"
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#include "modules/utility/interface/audio_frame_operations.h"
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#include "trace.h"
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namespace webrtc {
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@ -18,8 +18,8 @@ namespace webrtc {
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class AudioFrame;
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// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
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// Change reference parameters to pointers. Move out of VoE to a common place.
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// Consider using a namespace rather than class.
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// Change reference parameters to pointers. Consider using a namespace rather
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// than a class.
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class AudioFrameOperations {
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public:
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static int MonoToStereo(AudioFrame& frame);
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@ -29,10 +29,12 @@
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],
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},
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'sources': [
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'../interface/audio_frame_operations.h'
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'../interface/file_player.h',
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'../interface/file_recorder.h',
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'../interface/process_thread.h',
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'../interface/rtp_dump.h',
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'audio_frame_operations.cc',
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'coder.cc',
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'coder.h',
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'file_player_impl.cc',
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@ -77,6 +79,7 @@
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'<(webrtc_root)/../test/test.gyp:test_support_main',
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],
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'sources': [
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'audio_frame_operations_unittest.cc',
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'file_player_unittest.cc',
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],
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}, # webrtc_utility_unittests
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@ -54,8 +54,6 @@
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'../interface/voe_rtp_rtcp.h',
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'../interface/voe_video_sync.h',
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'../interface/voe_volume_control.h',
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'audio_frame_operations.cc',
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'audio_frame_operations.h',
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'channel.cc',
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'channel.h',
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'channel_manager.cc',
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@ -141,7 +139,6 @@
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'../interface',
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],
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'sources': [
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'audio_frame_operations_unittest.cc',
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'channel_unittest.cc',
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],
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},
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