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}

webrtc/DEPS

@@ -9,11 +9,7 @@ include_rules = [
   "+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 @@ include_rules = [
   "+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",
   ],

webrtc/api/BUILD.gn

@@ -19,6 +19,25 @@ group("api") {
   ]
 }
 
+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 @@ source_set("libjingle_peerconnection") {
   }
 
   deps = [
+    ":call_api",
     "../call",
     "../media",
     "../pc",

webrtc/api/api.gyp

@@ -94,10 +94,27 @@
     }],
   ],  # 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',
       ],

webrtc/api/call/DEPS

@@ -0,0 +1,4 @@
+include_rules = [
+  "+webrtc/modules/audio_coding/codecs",
+]
+

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 @@ class AudioReceiveStream {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_RECEIVE_STREAM_H_
+#endif  // WEBRTC_API_CALL_AUDIO_RECEIVE_STREAM_H_

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 @@ class AudioSendStream {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_SEND_STREAM_H_
+#endif  // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_

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 @@ class AudioSinkInterface {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_SINK_H_
+#endif  // WEBRTC_API_CALL_AUDIO_SINK_H_

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 @@ class AudioState : public rtc::RefCountInterface {
 };
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_STATE_H_
+#endif  // WEBRTC_API_CALL_AUDIO_STATE_H_

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"
 

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"

webrtc/audio/BUILD.gn

@@ -31,6 +31,7 @@ source_set("audio") {
 
   deps = [
     "..:webrtc_common",
+    "../api:call_api",
     "../system_wrappers",
     "../voice_engine",
   ]

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"

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"

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"

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"

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',

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"
 

webrtc/call/BUILD.gn

@@ -28,6 +28,7 @@ source_set("call") {
   deps = [
     "..:rtc_event_log",
     "..:webrtc_common",
+    "../api:call_api",
     "../audio",
     "../modules/congestion_controller",
     "../modules/rtp_rtcp",

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"

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"

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',

webrtc/media/BUILD.gn

@@ -204,6 +204,7 @@ source_set("rtc_media") {
 
   deps += [
     "..:webrtc_common",
+    "../api:call_api",
     "../base:rtc_base_approved",
     "../call",
     "../libjingle/xmllite",

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"

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"

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"

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"

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"

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"

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',
           ],

webrtc/pc/BUILD.gn

@@ -49,6 +49,7 @@ source_set("rtc_pc") {
   ]
 
   deps = [
+    "../api:call_api",
     "../base:rtc_base",
     "../media",
   ]

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"

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"

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"

webrtc/voice_engine/BUILD.gn

@@ -92,6 +92,7 @@ source_set("voice_engine") {
     ":level_indicator",
     "..:rtc_event_log",
     "..:webrtc_common",
+    "../api:call_api",
     "../base:rtc_base_approved",
     "../common_audio",
     "../modules/audio_coding",

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"

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"
 

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',

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',

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',