Rename neteq4 folder to neteq
Keep the old neteq4/audio_decoder_unittests.isolate while waiting for
a hard-coded reference to change.
This CL effectively reverts r6257 "Rename neteq4 folder to neteq".
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6367 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq/neteq.gypi b/webrtc/modules/audio_coding/neteq/neteq.gypi
new file mode 100644
index 0000000..ccdc9f5
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -0,0 +1,234 @@
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+{
+ 'variables': {
+ 'codecs': [
+ 'G711',
+ 'G722',
+ 'PCM16B',
+ 'iLBC',
+ 'iSAC',
+ 'iSACFix',
+ 'CNG',
+ ],
+ 'neteq_defines': [],
+ 'conditions': [
+ ['include_opus==1', {
+ 'codecs': ['webrtc_opus',],
+ 'neteq_defines': ['WEBRTC_CODEC_OPUS',],
+ }],
+ ],
+ 'neteq_dependencies': [
+ '<@(codecs)',
+ '<(DEPTH)/third_party/opus/opus.gyp:opus',
+ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ ],
+ },
+ 'targets': [
+ {
+ 'target_name': 'neteq',
+ 'type': 'static_library',
+ 'dependencies': [
+ '<@(neteq_dependencies)',
+ ],
+ 'defines': [
+ '<@(neteq_defines)',
+ ],
+ 'include_dirs': [
+ # Need Opus header files for the audio classifier.
+ '<(DEPTH)/third_party/opus/src/celt',
+ '<(DEPTH)/third_party/opus/src/src',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ # Need Opus header files for the audio classifier.
+ '<(DEPTH)/third_party/opus/src/celt',
+ '<(DEPTH)/third_party/opus/src/src',
+ ],
+ },
+ 'export_dependent_settings': [
+ '<(DEPTH)/third_party/opus/opus.gyp:opus',
+ ],
+ 'sources': [
+ 'interface/audio_decoder.h',
+ 'interface/neteq.h',
+ 'accelerate.cc',
+ 'accelerate.h',
+ 'audio_classifier.cc',
+ 'audio_classifier.h',
+ 'audio_decoder_impl.cc',
+ 'audio_decoder_impl.h',
+ 'audio_decoder.cc',
+ 'audio_multi_vector.cc',
+ 'audio_multi_vector.h',
+ 'audio_vector.cc',
+ 'audio_vector.h',
+ 'background_noise.cc',
+ 'background_noise.h',
+ 'buffer_level_filter.cc',
+ 'buffer_level_filter.h',
+ 'comfort_noise.cc',
+ 'comfort_noise.h',
+ 'decision_logic.cc',
+ 'decision_logic.h',
+ 'decision_logic_fax.cc',
+ 'decision_logic_fax.h',
+ 'decision_logic_normal.cc',
+ 'decision_logic_normal.h',
+ 'decoder_database.cc',
+ 'decoder_database.h',
+ 'defines.h',
+ 'delay_manager.cc',
+ 'delay_manager.h',
+ 'delay_peak_detector.cc',
+ 'delay_peak_detector.h',
+ 'dsp_helper.cc',
+ 'dsp_helper.h',
+ 'dtmf_buffer.cc',
+ 'dtmf_buffer.h',
+ 'dtmf_tone_generator.cc',
+ 'dtmf_tone_generator.h',
+ 'expand.cc',
+ 'expand.h',
+ 'merge.cc',
+ 'merge.h',
+ 'neteq_impl.cc',
+ 'neteq_impl.h',
+ 'neteq.cc',
+ 'statistics_calculator.cc',
+ 'statistics_calculator.h',
+ 'normal.cc',
+ 'normal.h',
+ 'packet_buffer.cc',
+ 'packet_buffer.h',
+ 'payload_splitter.cc',
+ 'payload_splitter.h',
+ 'post_decode_vad.cc',
+ 'post_decode_vad.h',
+ 'preemptive_expand.cc',
+ 'preemptive_expand.h',
+ 'random_vector.cc',
+ 'random_vector.h',
+ 'rtcp.cc',
+ 'rtcp.h',
+ 'sync_buffer.cc',
+ 'sync_buffer.h',
+ 'timestamp_scaler.cc',
+ 'timestamp_scaler.h',
+ 'time_stretch.cc',
+ 'time_stretch.h',
+ ],
+ },
+ ], # targets
+ 'conditions': [
+ ['include_tests==1', {
+ 'includes': ['neteq_tests.gypi',],
+ 'targets': [
+ {
+ 'target_name': 'audio_decoder_unittests',
+ 'type': '<(gtest_target_type)',
+ 'dependencies': [
+ '<@(codecs)',
+ '<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
+ '<(webrtc_root)/test/test.gyp:test_support_main',
+ ],
+ 'defines': [
+ 'AUDIO_DECODER_UNITTEST',
+ 'WEBRTC_CODEC_G722',
+ 'WEBRTC_CODEC_ILBC',
+ 'WEBRTC_CODEC_ISACFX',
+ 'WEBRTC_CODEC_ISAC',
+ 'WEBRTC_CODEC_PCM16',
+ '<@(neteq_defines)',
+ ],
+ 'sources': [
+ 'audio_decoder_impl.cc',
+ 'audio_decoder_impl.h',
+ 'audio_decoder_unittest.cc',
+ 'audio_decoder.cc',
+ 'interface/audio_decoder.h',
+ ],
+ 'conditions': [
+ # TODO(henrike): remove build_with_chromium==1 when the bots are
+ # using Chromium's buildbots.
+ ['build_with_chromium==1 and OS=="android"', {
+ 'dependencies': [
+ '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
+ ],
+ }],
+ ],
+ }, # audio_decoder_unittests
+
+ {
+ 'target_name': 'neteq_unittest_tools',
+ 'type': 'static_library',
+ 'dependencies': [
+ 'rtp_rtcp',
+ ],
+ 'direct_dependent_settings': {
+ 'include_dirs': [
+ 'tools',
+ ],
+ },
+ 'include_dirs': [
+ 'tools',
+ ],
+ 'sources': [
+ 'tools/audio_loop.cc',
+ 'tools/audio_loop.h',
+ 'tools/input_audio_file.cc',
+ 'tools/input_audio_file.h',
+ 'tools/packet.cc',
+ 'tools/packet.h',
+ 'tools/packet_source.h',
+ 'tools/rtp_file_source.cc',
+ 'tools/rtp_file_source.h',
+ 'tools/rtp_generator.cc',
+ 'tools/rtp_generator.h',
+ ],
+ }, # neteq_unittest_tools
+ ], # targets
+ 'conditions': [
+ # TODO(henrike): remove build_with_chromium==1 when the bots are using
+ # Chromium's buildbots.
+ ['build_with_chromium==1 and OS=="android"', {
+ 'targets': [
+ {
+ 'target_name': 'audio_decoder_unittests_apk_target',
+ 'type': 'none',
+ 'dependencies': [
+ '<(apk_tests_path):audio_decoder_unittests_apk',
+ ],
+ },
+ ],
+ }],
+ ['test_isolation_mode != "noop"', {
+ 'targets': [
+ {
+ 'target_name': 'audio_decoder_unittests_run',
+ 'type': 'none',
+ 'dependencies': [
+ 'audio_decoder_unittests',
+ ],
+ 'includes': [
+ '../../../build/isolate.gypi',
+ 'audio_decoder_unittests.isolate',
+ ],
+ 'sources': [
+ 'audio_decoder_unittests.isolate',
+ ],
+ },
+ ],
+ }],
+ ],
+ }], # include_tests
+ ], # conditions
+}