Move webrtc/audio_*.h to webrtc/api/call
BUG=webrtc:5878
NOTRY=True
Review-Url: https://codereview.webrtc.org/2059703002
Cr-Commit-Position: refs/heads/master@{#13996}
diff --git a/webrtc/DEPS b/webrtc/DEPS
index 5fab15c..41e0b71 100644
--- a/webrtc/DEPS
+++ b/webrtc/DEPS
@@ -9,11 +9,7 @@
"+libyuv",
"+testing",
"-webrtc", # Has to be disabled; otherwise all dirs below will be allowed.
- # Individual headers that will be moved out of here, see webrtc:
- "+webrtc/audio_receive_stream.h",
- "+webrtc/audio_send_stream.h",
- "+webrtc/audio_sink.h",
- "+webrtc/audio_state.h",
+ # Individual headers that will be moved out of here, see webrtc:4243.
"+webrtc/call.h",
"+webrtc/common.h",
"+webrtc/common_types.h",
@@ -29,20 +25,15 @@
"+webrtc/video_send_stream.h",
"+WebRTC",
+ "+webrtc/api",
"+webrtc/base",
"+webrtc/modules/include",
"+webrtc/test",
"+webrtc/tools",
]
-# The below rules will be removed when webrtc: is fixed.
+# The below rules will be removed when webrtc:4243 is fixed.
specific_include_rules = {
- "audio_send_stream\.h": [
- "+webrtc/modules/audio_coding",
- ],
- "audio_receive_stream\.h": [
- "+webrtc/modules/audio_coding/codecs/audio_decoder_factory.h",
- ],
"video_frame\.h": [
"+webrtc/common_video",
],
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index f09c4e4..d14fe1f 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -19,6 +19,25 @@
]
}
+source_set("call_api") {
+ sources = [
+ "call/audio_receive_stream.h",
+ "call/audio_send_stream.h",
+ "call/audio_sink.h",
+ "call/audio_state.h",
+ ]
+
+ configs += [ "..:common_config" ]
+ public_configs = [ "..:common_inherited_config" ]
+
+ deps = [
+ # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+ "..:webrtc_common",
+ "../base:rtc_base_approved",
+ "../modules/audio_coding:audio_encoder_interface",
+ ]
+}
+
config("libjingle_peerconnection_warnings_config") {
# GN orders flags on a target before flags from configs. The default config
# adds these flags so to cancel them out they need to come from a config and
@@ -113,6 +132,7 @@
}
deps = [
+ ":call_api",
"../call",
"../media",
"../pc",
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index 274b87c..5b14fdf 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -95,9 +95,26 @@
], # conditions
'targets': [
{
+ 'target_name': 'call_api',
+ 'type': 'static_library',
+ 'dependencies': [
+ # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
+ '<(webrtc_root)/base/base.gyp:rtc_base_approved',
+ '<(webrtc_root)/common.gyp:webrtc_common',
+ '<(webrtc_root)/modules/modules.gyp:audio_encoder_interface',
+ ],
+ 'sources': [
+ 'call/audio_receive_stream.h',
+ 'call/audio_send_stream.h',
+ 'call/audio_sink.h',
+ 'call/audio_state.h',
+ ],
+ },
+ {
'target_name': 'libjingle_peerconnection',
'type': 'static_library',
'dependencies': [
+ ':call_api',
'<(webrtc_root)/media/media.gyp:rtc_media',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
diff --git a/webrtc/api/call/DEPS b/webrtc/api/call/DEPS
new file mode 100644
index 0000000..d1d4309
--- /dev/null
+++ b/webrtc/api/call/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+ "+webrtc/modules/audio_coding/codecs",
+]
+
diff --git a/webrtc/audio_receive_stream.h b/webrtc/api/call/audio_receive_stream.h
similarity index 96%
rename from webrtc/audio_receive_stream.h
rename to webrtc/api/call/audio_receive_stream.h
index e0e9536..096cbc7 100644
--- a/webrtc/audio_receive_stream.h
+++ b/webrtc/api/call/audio_receive_stream.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_AUDIO_RECEIVE_STREAM_H_
-#define WEBRTC_AUDIO_RECEIVE_STREAM_H_
+#ifndef WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
+#define WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
#include <map>
#include <memory>
@@ -136,4 +136,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_AUDIO_RECEIVE_STREAM_H_
+#endif // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_
diff --git a/webrtc/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h
similarity index 96%
rename from webrtc/audio_send_stream.h
rename to webrtc/api/call/audio_send_stream.h
index c3d0d33..b309f7a 100644
--- a/webrtc/audio_send_stream.h
+++ b/webrtc/api/call/audio_send_stream.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_AUDIO_SEND_STREAM_H_
-#define WEBRTC_AUDIO_SEND_STREAM_H_
+#ifndef WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
+#define WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
#include <memory>
#include <string>
@@ -116,4 +116,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_AUDIO_SEND_STREAM_H_
+#endif // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_
diff --git a/webrtc/audio_sink.h b/webrtc/api/call/audio_sink.h
similarity index 92%
rename from webrtc/audio_sink.h
rename to webrtc/api/call/audio_sink.h
index 2c932c5..e865ead 100644
--- a/webrtc/audio_sink.h
+++ b/webrtc/api/call/audio_sink.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_AUDIO_SINK_H_
-#define WEBRTC_AUDIO_SINK_H_
+#ifndef WEBRTC_API_CALL_AUDIO_SINK_H_
+#define WEBRTC_API_CALL_AUDIO_SINK_H_
#if defined(WEBRTC_POSIX) && !defined(__STDC_FORMAT_MACROS)
// Avoid conflict with format_macros.h.
@@ -50,4 +50,4 @@
} // namespace webrtc
-#endif // WEBRTC_AUDIO_SINK_H_
+#endif // WEBRTC_API_CALL_AUDIO_SINK_H_
diff --git a/webrtc/audio_state.h b/webrtc/api/call/audio_state.h
similarity index 92%
rename from webrtc/audio_state.h
rename to webrtc/api/call/audio_state.h
index fa5784c..ac91277 100644
--- a/webrtc/audio_state.h
+++ b/webrtc/api/call/audio_state.h
@@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_AUDIO_STATE_H_
-#define WEBRTC_AUDIO_STATE_H_
+#ifndef WEBRTC_API_CALL_AUDIO_STATE_H_
+#define WEBRTC_API_CALL_AUDIO_STATE_H_
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
@@ -45,4 +45,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_AUDIO_STATE_H_
+#endif // WEBRTC_API_CALL_AUDIO_STATE_H_
diff --git a/webrtc/api/remoteaudiosource.h b/webrtc/api/remoteaudiosource.h
index 4cc68f8..a67b895 100644
--- a/webrtc/api/remoteaudiosource.h
+++ b/webrtc/api/remoteaudiosource.h
@@ -14,8 +14,8 @@
#include <list>
#include <string>
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/api/notifier.h"
-#include "webrtc/audio_sink.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/pc/channel.h"
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
index f8a8f67..df4a48a 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -17,12 +17,12 @@
#include <utility>
#include <vector>
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/sctputils.h"
#include "webrtc/api/webrtcsessiondescriptionfactory.h"
-#include "webrtc/audio_sink.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index d031f69..762eb5f 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -31,6 +31,7 @@
deps = [
"..:webrtc_common",
+ "../api:call_api",
"../system_wrappers",
"../voice_engine",
]
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index ca2c08b..91237c3 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -13,7 +13,7 @@
#include <string>
#include <utility>
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/audio/audio_state.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/base/checks.h"
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 24924c9a..871d48d6 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -13,8 +13,8 @@
#include <memory>
-#include "webrtc/audio_receive_stream.h"
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_receive_stream.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index a993d5f..ec2a4db 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -13,8 +13,8 @@
#include <memory>
-#include "webrtc/audio_send_stream.h"
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_send_stream.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/call/bitrate_allocator.h"
diff --git a/webrtc/audio/audio_state.h b/webrtc/audio/audio_state.h
index 6bee3c6..31892d0 100644
--- a/webrtc/audio/audio_state.h
+++ b/webrtc/audio/audio_state.h
@@ -11,7 +11,7 @@
#ifndef WEBRTC_AUDIO_AUDIO_STATE_H_
#define WEBRTC_AUDIO_AUDIO_STATE_H_
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/audio/scoped_voe_interface.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
diff --git a/webrtc/audio/webrtc_audio.gypi b/webrtc/audio/webrtc_audio.gypi
index 9b4879a..6230057 100644
--- a/webrtc/audio/webrtc_audio.gypi
+++ b/webrtc/audio/webrtc_audio.gypi
@@ -8,6 +8,7 @@
{
'variables': {
'webrtc_audio_dependencies': [
+ '<(webrtc_root)/api/api.gyp:call_api',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
diff --git a/webrtc/call.h b/webrtc/call.h
index ec43b18..5f64590 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -13,13 +13,13 @@
#include <string>
#include <vector>
-#include "webrtc/common_types.h"
-#include "webrtc/audio_receive_stream.h"
-#include "webrtc/audio_send_stream.h"
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_receive_stream.h"
+#include "webrtc/api/call/audio_send_stream.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/base/networkroute.h"
#include "webrtc/base/platform_file.h"
#include "webrtc/base/socket.h"
+#include "webrtc/common_types.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 5b428f7..2f3ce8c 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -28,6 +28,7 @@
deps = [
"..:rtc_event_log",
"..:webrtc_common",
+ "../api:call_api",
"../audio",
"../modules/congestion_controller",
"../modules/rtp_rtcp",
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 10d435d..ab8643d 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -14,7 +14,7 @@
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/event.h"
#include "webrtc/base/logging.h"
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc
index fb6cac1..4c2562d 100644
--- a/webrtc/call/call_unittest.cc
+++ b/webrtc/call/call_unittest.cc
@@ -13,7 +13,7 @@
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/call.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_decoder_factory.h"
#include "webrtc/test/mock_voice_engine.h"
diff --git a/webrtc/common.gyp b/webrtc/common.gyp
index c9ac71c..e5d41da 100644
--- a/webrtc/common.gyp
+++ b/webrtc/common.gyp
@@ -12,7 +12,6 @@
'target_name': 'webrtc_common',
'type': 'static_library',
'sources': [
- 'audio_sink.h',
'common.cc',
'common.h',
'common_types.cc',
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index 78e7f78..df4e3d2 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -204,6 +204,7 @@
deps += [
"..:webrtc_common",
+ "../api:call_api",
"../base:rtc_base_approved",
"../call",
"../libjingle/xmllite",
diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
index dfc7a1c..4560317 100644
--- a/webrtc/media/base/fakemediaengine.h
+++ b/webrtc/media/base/fakemediaengine.h
@@ -18,7 +18,7 @@
#include <string>
#include <vector>
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/copyonwritebuffer.h"
#include "webrtc/base/networkroute.h"
#include "webrtc/base/stringutils.h"
diff --git a/webrtc/media/base/mediaengine.h b/webrtc/media/base/mediaengine.h
index 559617f..866898c 100644
--- a/webrtc/media/base/mediaengine.h
+++ b/webrtc/media/base/mediaengine.h
@@ -18,7 +18,7 @@
#include <string>
#include <vector>
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/api/rtpparameters.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/sigslotrepeater.h"
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index fdf7cf3..d0655cf 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -13,7 +13,7 @@
#include <algorithm>
#include <utility>
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/gunit.h"
#include "webrtc/media/base/rtputils.h"
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 8581d82..f723d7e 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -23,8 +23,8 @@
#include <memory>
#include <vector>
-#include "webrtc/audio_receive_stream.h"
-#include "webrtc/audio_send_stream.h"
+#include "webrtc/api/call/audio_receive_stream.h"
+#include "webrtc/api/call/audio_send_stream.h"
#include "webrtc/base/buffer.h"
#include "webrtc/call.h"
#include "webrtc/video_frame.h"
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index d0ca9ae..0169d5a 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -18,7 +18,7 @@
#include <string>
#include <vector>
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/byteorder.h"
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index b2f9f62..6508e23 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -16,7 +16,7 @@
#include <string>
#include <vector>
-#include "webrtc/audio_state.h"
+#include "webrtc/api/call/audio_state.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/networkroute.h"
diff --git a/webrtc/media/media.gyp b/webrtc/media/media.gyp
index c948226..2eb7271 100644
--- a/webrtc/media/media.gyp
+++ b/webrtc/media/media.gyp
@@ -203,6 +203,7 @@
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/api/api.gyp:call_api',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/media/media.gyp:rtc_media',
],
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index ea3e5ae..906d995 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -49,6 +49,7 @@
]
deps = [
+ "../api:call_api",
"../base:rtc_base",
"../media",
]
diff --git a/webrtc/pc/channel.cc b/webrtc/pc/channel.cc
index 98838dd..f22bdc4c 100644
--- a/webrtc/pc/channel.cc
+++ b/webrtc/pc/channel.cc
@@ -12,7 +12,7 @@
#include "webrtc/pc/channel.h"
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/checks.h"
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index 7d3bc94..6ea5419 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -18,7 +18,7 @@
#include <utility>
#include <vector>
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/asyncinvoker.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/criticalsection.h"
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index 60051a7..2a23c94 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -17,8 +17,8 @@
#include <string>
#include <utility>
-#include "webrtc/audio_receive_stream.h"
-#include "webrtc/audio_send_stream.h"
+#include "webrtc/api/call/audio_receive_stream.h"
+#include "webrtc/api/call/audio_send_stream.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/call.h"
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 8f0e9f3..ffa17ed 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -92,6 +92,7 @@
":level_indicator",
"..:rtc_event_log",
"..:webrtc_common",
+ "../api:call_api",
"../base:rtc_base_approved",
"../common_audio",
"../modules/audio_coding",
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 6fc4f34..adeb1da 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -13,7 +13,7 @@
#include <memory>
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/optional.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
diff --git a/webrtc/voice_engine/channel_proxy.cc b/webrtc/voice_engine/channel_proxy.cc
index 8c942f3..fc22180 100644
--- a/webrtc/voice_engine/channel_proxy.cc
+++ b/webrtc/voice_engine/channel_proxy.cc
@@ -12,7 +12,7 @@
#include <utility>
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/checks.h"
#include "webrtc/voice_engine/channel.h"
diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp
index 912b522..dd6de61 100644
--- a/webrtc/voice_engine/voice_engine.gyp
+++ b/webrtc/voice_engine/voice_engine.gyp
@@ -15,6 +15,7 @@
'target_name': 'voice_engine',
'type': 'static_library',
'dependencies': [
+ '<(webrtc_root)/api/api.gyp:call_api',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp
index 8a56fe7..a4c1c1d 100644
--- a/webrtc/webrtc.gyp
+++ b/webrtc/webrtc.gyp
@@ -17,9 +17,6 @@
'target_name': 'webrtc',
'type': 'static_library',
'sources': [
- 'audio_receive_stream.h',
- 'audio_send_stream.h',
- 'audio_state.h',
'call.h',
'config.h',
'transport.h',
diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi
index 6d8ff6a..93c6c88 100644
--- a/webrtc/webrtc_tests.gypi
+++ b/webrtc/webrtc_tests.gypi
@@ -385,6 +385,7 @@
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/api/api.gyp:call_api',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:video_capture',
