Move remaining traces of VoiceEngine
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}diff --git a/.gn b/.gn
index 5e3c6ca..e1ed9ae 100644
--- a/.gn
+++ b/.gn
@@ -42,7 +42,6 @@
"//system_wrappers/*",
"//test/*",
"//video/*",
- "//voice_engine/*",
"//third_party/libyuv/*",
]
diff --git a/BUILD.gn b/BUILD.gn
index 4e6f01b..bb5c953 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -63,7 +63,6 @@
"video:screenshare_loopback",
"video:sv_loopback",
"video:video_loopback",
- "voice_engine:voice_engine_unittests",
]
if (is_android) {
deps += [
@@ -342,7 +341,6 @@
"sdk",
"system_wrappers:system_wrappers_default",
"video",
- "voice_engine",
]
if (build_with_mozilla) {
diff --git a/WATCHLISTS b/WATCHLISTS
index ed0a614..b6a5703 100644
--- a/WATCHLISTS
+++ b/WATCHLISTS
@@ -23,8 +23,7 @@
'filepath': '^[^/]*$|^webrtc/[^/]*$|^webrtc/build/.*',
},
'documented_interfaces': {
- 'filepath': '^webrtc/[^/]*\.h$|'\
- 'webrtc/voice_engine/include/.*',
+ 'filepath': '^webrtc/[^/]*\.h$',
},
'build_files': {
'filepath': '\.gyp$|\.gypi$|Android\.mk$',
@@ -50,9 +49,6 @@
'video': {
'filepath': 'webrtc/video/.*',
},
- 'voice_engine': {
- 'filepath': 'webrtc/voice_engine/.*',
- },
'common_audio': {
'filepath': 'webrtc/common_audio/.*',
},
@@ -126,20 +122,11 @@
'call': ['mflodman@webrtc.org',
'solenberg@webrtc.org',
'stefan@webrtc.org'],
- 'media_engine': ['solenberg@webrtc.org'],
'video': ['mflodman@webrtc.org',
'stefan@webrtc.org',
'video-team@agora.io',
'yujie.mao@webrtc.org',
'zhengzhonghou@agora.io'],
- 'voice_engine': ['alessiob@webrtc.org',
- 'andrew@webrtc.org',
- 'audio-team@agora.io',
- 'henrika@webrtc.org',
- 'henrik.lundin@webrtc.org',
- 'minyue@webrtc.org',
- 'peah@webrtc.org',
- 'solenberg@webrtc.org'],
'video_capture': ['mflodman@webrtc.org',
'perkj@webrtc.org',
'sdk-team@agora.io',
diff --git a/api/DEPS b/api/DEPS
index a537633..5ac116c 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -7,10 +7,6 @@
]
specific_include_rules = {
- "peerconnection_jni\.cc": [
- "+voice_engine",
- ],
-
# TODO(ossu): Remove this exception when {builtin_,}audio_encoder_factory.h
# has moved to api/.
"peerconnectioninterface\.h": [
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index d3af836..2eee20b 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -14,6 +14,8 @@
rtc_static_library("audio") {
sources = [
+ "audio_level.cc",
+ "audio_level.h",
"audio_receive_stream.cc",
"audio_receive_stream.h",
"audio_send_stream.cc",
@@ -22,11 +24,19 @@
"audio_state.h",
"audio_transport_impl.cc",
"audio_transport_impl.h",
+ "channel.cc",
+ "channel.h",
+ "channel_proxy.cc",
+ "channel_proxy.h",
"conversion.h",
"null_audio_poller.cc",
"null_audio_poller.h",
+ "remix_resample.cc",
+ "remix_resample.h",
"time_interval.cc",
"time_interval.h",
+ "transport_feedback_packet_loss_tracker.cc",
+ "transport_feedback_packet_loss_tracker.h",
]
if (!build_with_chromium && is_clang) {
@@ -36,15 +46,23 @@
deps = [
"..:webrtc_common",
+ "../api:array_view",
"../api:audio_mixer_api",
"../api:call_api",
+ "../api:libjingle_peerconnection_api",
"../api:optional",
+ "../api:transport_api",
"../api/audio_codecs:audio_codecs_api",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../call:bitrate_allocator",
"../call:call_interfaces",
"../call:rtp_interfaces",
"../common_audio",
+ "../logging:rtc_event_log_api",
+ "../modules:module_api",
+ "../modules/audio_coding",
+ "../modules/audio_coding:audio_format_conversion",
+ "../modules/audio_coding:audio_network_adaptor_config",
"../modules/audio_coding:cng",
"../modules/audio_device",
"../modules/audio_processing",
@@ -53,14 +71,16 @@
"../modules/pacing:pacing",
"../modules/remote_bitrate_estimator:remote_bitrate_estimator",
"../modules/rtp_rtcp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
+ "../modules/utility",
"../rtc_base:checks",
+ "../rtc_base:rate_limiter",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_task_queue",
"../system_wrappers",
"../system_wrappers:field_trial_api",
- "../voice_engine",
- "../voice_engine:audio_level",
+ "../system_wrappers:metrics_api",
"utility:audio_frame_operations",
]
}
@@ -94,7 +114,10 @@
"audio_send_stream_tests.cc",
"audio_send_stream_unittest.cc",
"audio_state_unittest.cc",
+ "mock_voe_channel_proxy.h",
+ "remix_resample_unittest.cc",
"time_interval_unittest.cc",
+ "transport_feedback_packet_loss_tracker_unittest.cc",
]
deps = [
":audio",
@@ -104,7 +127,9 @@
"../call:mock_rtp_interfaces",
"../call:rtp_interfaces",
"../call:rtp_receiver",
+ "../common_audio",
"../logging:mocks",
+ "../modules:module_api",
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing:audio_processing_statistics",
@@ -116,6 +141,7 @@
"../modules/pacing:pacing",
"../modules/rtp_rtcp:mock_rtp_rtcp",
"../modules/rtp_rtcp:rtp_rtcp_format",
+ "../rtc_base:checks",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:rtc_task_queue",
@@ -125,7 +151,6 @@
"../test:rtp_test_utils",
"../test:test_common",
"../test:test_support",
- "../voice_engine",
"utility:utility_tests",
"//testing/gmock",
"//testing/gtest",
diff --git a/audio/DEPS b/audio/DEPS
index 70e3346..8bb1f80 100644
--- a/audio/DEPS
+++ b/audio/DEPS
@@ -5,14 +5,15 @@
"+modules/audio_coding",
"+modules/audio_device",
"+modules/audio_mixer",
+ "+modules/audio_processing",
"+modules/audio_processing/include",
"+modules/bitrate_controller",
"+modules/congestion_controller",
"+modules/pacing",
"+modules/remote_bitrate_estimator",
"+modules/rtp_rtcp",
+ "+modules/utility",
"+system_wrappers",
- "+voice_engine",
]
specific_include_rules = {
diff --git a/voice_engine/audio_level.cc b/audio/audio_level.cc
similarity index 98%
rename from voice_engine/audio_level.cc
rename to audio/audio_level.cc
index 57b4855..fe2a240 100644
--- a/voice_engine/audio_level.cc
+++ b/audio/audio_level.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "voice_engine/audio_level.h"
+#include "audio/audio_level.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/include/module_common_types.h"
diff --git a/voice_engine/audio_level.h b/audio/audio_level.h
similarity index 91%
rename from voice_engine/audio_level.h
rename to audio/audio_level.h
index a1951ed..883641a 100644
--- a/voice_engine/audio_level.h
+++ b/audio/audio_level.h
@@ -8,12 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef VOICE_ENGINE_AUDIO_LEVEL_H_
-#define VOICE_ENGINE_AUDIO_LEVEL_H_
+#ifndef AUDIO_AUDIO_LEVEL_H_
+#define AUDIO_AUDIO_LEVEL_H_
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
-#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@@ -57,4 +56,4 @@
} // namespace voe
} // namespace webrtc
-#endif // VOICE_ENGINE_AUDIO_LEVEL_H_
+#endif // AUDIO_AUDIO_LEVEL_H_
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 294dad0..45ffe34 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -16,6 +16,7 @@
#include "api/call/audio_sink.h"
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
+#include "audio/channel_proxy.h"
#include "audio/conversion.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
@@ -24,7 +25,6 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/timeutils.h"
-#include "voice_engine/channel_proxy.h"
namespace webrtc {
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index c8318aa..fa663fe 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -15,6 +15,7 @@
#include "api/test/mock_audio_mixer.h"
#include "audio/audio_receive_stream.h"
#include "audio/conversion.h"
+#include "audio/mock_voe_channel_proxy.h"
#include "call/rtp_stream_receiver_controller.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_device/include/mock_audio_device.h"
@@ -24,7 +25,6 @@
#include "modules/rtp_rtcp/source/byte_io.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder_factory.h"
-#include "test/mock_voe_channel_proxy.h"
namespace webrtc {
namespace test {
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index d207dbd..80c2c6b 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -15,6 +15,7 @@
#include <vector>
#include "audio/audio_state.h"
+#include "audio/channel_proxy.h"
#include "audio/conversion.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
@@ -28,7 +29,6 @@
#include "rtc_base/task_queue.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
-#include "voice_engine/channel_proxy.h"
namespace webrtc {
namespace internal {
diff --git a/audio/audio_send_stream.h b/audio/audio_send_stream.h
index bab8f80..093ca46 100644
--- a/audio/audio_send_stream.h
+++ b/audio/audio_send_stream.h
@@ -15,6 +15,7 @@
#include <vector>
#include "audio/time_interval.h"
+#include "audio/transport_feedback_packet_loss_tracker.h"
#include "call/audio_send_stream.h"
#include "call/audio_state.h"
#include "call/bitrate_allocator.h"
@@ -22,7 +23,6 @@
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
namespace webrtc {
class RtcEventLog;
diff --git a/audio/audio_send_stream_unittest.cc b/audio/audio_send_stream_unittest.cc
index 4644bf4..925f93e 100644
--- a/audio/audio_send_stream_unittest.cc
+++ b/audio/audio_send_stream_unittest.cc
@@ -15,6 +15,7 @@
#include "audio/audio_send_stream.h"
#include "audio/audio_state.h"
#include "audio/conversion.h"
+#include "audio/mock_voe_channel_proxy.h"
#include "call/fake_rtp_transport_controller_send.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
@@ -32,7 +33,6 @@
#include "test/gtest.h"
#include "test/mock_audio_encoder.h"
#include "test/mock_audio_encoder_factory.h"
-#include "test/mock_voe_channel_proxy.h"
namespace webrtc {
namespace test {
diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc
index 30ffc6d..f9b0311 100644
--- a/audio/audio_transport_impl.cc
+++ b/audio/audio_transport_impl.cc
@@ -14,10 +14,10 @@
#include <memory>
#include <utility>
+#include "audio/remix_resample.h"
#include "audio/utility/audio_frame_operations.h"
#include "call/audio_send_stream.h"
#include "rtc_base/logging.h"
-#include "voice_engine/utility.h"
namespace webrtc {
diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index 8a316a5..4e6e047 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -14,6 +14,7 @@
#include <vector>
#include "api/audio/audio_mixer.h"
+#include "audio/audio_level.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
@@ -22,7 +23,6 @@
#include "rtc_base/criticalsection.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread_annotations.h"
-#include "voice_engine/audio_level.h"
namespace webrtc {
diff --git a/voice_engine/channel.cc b/audio/channel.cc
similarity index 98%
rename from voice_engine/channel.cc
rename to audio/channel.cc
index efc76bd..1799e7a 100644
--- a/voice_engine/channel.cc
+++ b/audio/channel.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "voice_engine/channel.h"
+#include "audio/channel.h"
#include <algorithm>
#include <map>
@@ -365,10 +365,7 @@
return false;
}
- uint8_t* bufferToSendPtr = (uint8_t*)data;
- size_t bufferLength = len;
-
- if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
+ if (!_transportPtr->SendRtp(data, len, options)) {
RTC_LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed";
return false;
}
@@ -383,10 +380,7 @@
return false;
}
- uint8_t* bufferToSendPtr = (uint8_t*)data;
- size_t bufferLength = len;
-
- int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
+ int n = _transportPtr->SendRtcp(data, len);
if (n < 0) {
RTC_LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed";
return false;
diff --git a/voice_engine/channel.h b/audio/channel.h
similarity index 98%
rename from voice_engine/channel.h
rename to audio/channel.h
index 3d6dd8f..c5d243c 100644
--- a/voice_engine/channel.h
+++ b/audio/channel.h
@@ -8,16 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef VOICE_ENGINE_CHANNEL_H_
-#define VOICE_ENGINE_CHANNEL_H_
+#ifndef AUDIO_CHANNEL_H_
+#define AUDIO_CHANNEL_H_
+#include <map>
#include <memory>
+#include <string>
+#include <vector>
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/call/audio_sink.h"
#include "api/call/transport.h"
#include "api/optional.h"
+#include "audio/audio_level.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_processing/rms_level.h"
@@ -29,7 +33,6 @@
#include "rtc_base/event.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_checker.h"
-#include "voice_engine/audio_level.h"
namespace rtc {
class TimestampWrapAroundHandler;
@@ -410,4 +413,4 @@
} // namespace voe
} // namespace webrtc
-#endif // VOICE_ENGINE_CHANNEL_H_
+#endif // AUDIO_CHANNEL_H_
diff --git a/voice_engine/channel_proxy.cc b/audio/channel_proxy.cc
similarity index 99%
rename from voice_engine/channel_proxy.cc
rename to audio/channel_proxy.cc
index 70564d4..cdfe090 100644
--- a/voice_engine/channel_proxy.cc
+++ b/audio/channel_proxy.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "voice_engine/channel_proxy.h"
+#include "audio/channel_proxy.h"
#include <utility>
diff --git a/voice_engine/channel_proxy.h b/audio/channel_proxy.h
similarity index 97%
rename from voice_engine/channel_proxy.h
rename to audio/channel_proxy.h
index ef81174..f5f603b 100644
--- a/voice_engine/channel_proxy.h
+++ b/audio/channel_proxy.h
@@ -8,9 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef VOICE_ENGINE_CHANNEL_PROXY_H_
-#define VOICE_ENGINE_CHANNEL_PROXY_H_
+#ifndef AUDIO_CHANNEL_PROXY_H_
+#define AUDIO_CHANNEL_PROXY_H_
+#include <map>
#include <memory>
#include <string>
#include <vector>
@@ -18,11 +19,11 @@
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_encoder.h"
#include "api/rtpreceiverinterface.h"
+#include "audio/channel.h"
#include "call/rtp_packet_sink_interface.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"
-#include "voice_engine/channel.h"
namespace webrtc {
@@ -141,4 +142,4 @@
} // namespace voe
} // namespace webrtc
-#endif // VOICE_ENGINE_CHANNEL_PROXY_H_
+#endif // AUDIO_CHANNEL_PROXY_H_
diff --git a/test/mock_voe_channel_proxy.h b/audio/mock_voe_channel_proxy.h
similarity index 95%
rename from test/mock_voe_channel_proxy.h
rename to audio/mock_voe_channel_proxy.h
index 29635ba..d512ad4 100644
--- a/test/mock_voe_channel_proxy.h
+++ b/audio/mock_voe_channel_proxy.h
@@ -8,14 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef TEST_MOCK_VOE_CHANNEL_PROXY_H_
-#define TEST_MOCK_VOE_CHANNEL_PROXY_H_
+#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
+#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
+#include <map>
+#include <memory>
#include <string>
+#include <vector>
+#include "audio/channel_proxy.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "test/gmock.h"
-#include "voice_engine/channel_proxy.h"
namespace webrtc {
namespace test {
@@ -101,4 +104,4 @@
} // namespace test
} // namespace webrtc
-#endif // TEST_MOCK_VOE_CHANNEL_PROXY_H_
+#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
diff --git a/voice_engine/utility.cc b/audio/remix_resample.cc
similarity index 98%
rename from voice_engine/utility.cc
rename to audio/remix_resample.cc
index 9398702..52a491f 100644
--- a/voice_engine/utility.cc
+++ b/audio/remix_resample.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "voice_engine/utility.h"
+#include "audio/remix_resample.h"
#include "audio/utility/audio_frame_operations.h"
#include "common_audio/resampler/include/push_resampler.h"
diff --git a/voice_engine/utility.h b/audio/remix_resample.h
similarity index 86%
rename from voice_engine/utility.h
rename to audio/remix_resample.h
index dc23e16..ddd8086 100644
--- a/voice_engine/utility.h
+++ b/audio/remix_resample.h
@@ -8,15 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-/*
- * Contains functions often used by different parts of VoiceEngine.
- */
-
-#ifndef VOICE_ENGINE_UTILITY_H_
-#define VOICE_ENGINE_UTILITY_H_
+#ifndef AUDIO_REMIX_RESAMPLE_H_
+#define AUDIO_REMIX_RESAMPLE_H_
#include "common_audio/resampler/include/push_resampler.h"
-#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
@@ -48,4 +43,4 @@
} // namespace voe
} // namespace webrtc
-#endif // VOICE_ENGINE_UTILITY_H_
+#endif // AUDIO_REMIX_RESAMPLE_H_
diff --git a/voice_engine/utility_unittest.cc b/audio/remix_resample_unittest.cc
similarity index 98%
rename from voice_engine/utility_unittest.cc
rename to audio/remix_resample_unittest.cc
index c798582..753584b 100644
--- a/voice_engine/utility_unittest.cc
+++ b/audio/remix_resample_unittest.cc
@@ -10,12 +10,12 @@
#include <math.h>
+#include "audio/remix_resample.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "modules/include/module_common_types.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/format_macros.h"
#include "test/gtest.h"
-#include "voice_engine/utility.h"
namespace webrtc {
namespace voe {
diff --git a/voice_engine/transport_feedback_packet_loss_tracker.cc b/audio/transport_feedback_packet_loss_tracker.cc
similarity index 99%
rename from voice_engine/transport_feedback_packet_loss_tracker.cc
rename to audio/transport_feedback_packet_loss_tracker.cc
index 774faf5..101b6b4 100644
--- a/voice_engine/transport_feedback_packet_loss_tracker.cc
+++ b/audio/transport_feedback_packet_loss_tracker.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
+#include "audio/transport_feedback_packet_loss_tracker.h"
#include <limits>
#include <utility>
diff --git a/voice_engine/transport_feedback_packet_loss_tracker.h b/audio/transport_feedback_packet_loss_tracker.h
similarity index 95%
rename from voice_engine/transport_feedback_packet_loss_tracker.h
rename to audio/transport_feedback_packet_loss_tracker.h
index d742078..7e73210 100644
--- a/voice_engine/transport_feedback_packet_loss_tracker.h
+++ b/audio/transport_feedback_packet_loss_tracker.h
@@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
-#define VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
+#ifndef AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
+#define AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
#include <map>
+#include <vector>
#include "api/optional.h"
#include "modules/include/module_common_types.h"
@@ -138,4 +139,4 @@
} // namespace webrtc
-#endif // VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
+#endif // AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
diff --git a/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc b/audio/transport_feedback_packet_loss_tracker_unittest.cc
similarity index 99%
rename from voice_engine/transport_feedback_packet_loss_tracker_unittest.cc
rename to audio/transport_feedback_packet_loss_tracker_unittest.cc
index 55626be..8f8fe05 100644
--- a/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc
+++ b/audio/transport_feedback_packet_loss_tracker_unittest.cc
@@ -13,12 +13,12 @@
#include <numeric>
#include <vector>
+#include "audio/transport_feedback_packet_loss_tracker.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
#include "rtc_base/checks.h"
#include "test/gmock.h"
#include "test/gtest.h"
-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
namespace webrtc {
diff --git a/call/BUILD.gn b/call/BUILD.gn
index d7ed4d9..03ed9b4 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -271,7 +271,6 @@
"../test:test_support",
"../test:video_test_common",
"../video",
- "../voice_engine",
"//testing/gtest",
]
if (!build_with_chromium && is_clang) {
diff --git a/call/DEPS b/call/DEPS
index 7622e24..307a26e 100644
--- a/call/DEPS
+++ b/call/DEPS
@@ -11,7 +11,6 @@
"+modules/rtp_rtcp",
"+modules/utility",
"+system_wrappers",
- "+voice_engine",
"+video",
]
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index f8d787c..4667498 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -98,9 +98,6 @@
Transport* rtcp_send_transport = nullptr;
- // TODO(solenberg): Remove once clients don't use it anymore.
- int voe_channel_id = -1;
-
// NetEq settings.
size_t jitter_buffer_max_packets = 50;
bool jitter_buffer_fast_accelerate = false;
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 03f32b7..908da11 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -87,9 +87,6 @@
// the entire life of the AudioSendStream and is owned by the API client.
Transport* send_transport = nullptr;
- // TODO(solenberg): Remove once clients don't use it anymore.
- int voe_channel_id = -1;
-
// Bitrate limits used for variable audio bitrate streams. Set both to -1 to
// disable audio bitrate adaptation.
// Note: This is still an experimental feature and not ready for real usage.
diff --git a/call/audio_state.h b/call/audio_state.h
index a8e57f0..9c40187 100644
--- a/call/audio_state.h
+++ b/call/audio_state.h
@@ -19,16 +19,12 @@
class AudioDeviceModule;
class AudioProcessing;
class AudioTransport;
-class VoiceEngine;
// AudioState holds the state which must be shared between multiple instances of
// webrtc::Call for audio processing purposes.
class AudioState : public rtc::RefCountInterface {
public:
struct Config {
- // TODO(solenberg): Remove once clients don't use it anymore.
- VoiceEngine* voice_engine = nullptr;
-
// The audio mixer connected to active receive streams. One per
// AudioState.
rtc::scoped_refptr<AudioMixer> audio_mixer;
diff --git a/media/BUILD.gn b/media/BUILD.gn
index b7ea5af..bfd91ce 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -317,7 +317,6 @@
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../system_wrappers:metrics_api",
- "../voice_engine",
]
}
@@ -616,7 +615,6 @@
"../test:audio_codec_mocks",
"../test:test_support",
"../test:video_test_common",
- "../voice_engine:voice_engine",
]
}
}
diff --git a/media/DEPS b/media/DEPS
index 7a266a2..99e62aa 100644
--- a/media/DEPS
+++ b/media/DEPS
@@ -14,7 +14,6 @@
"+pc",
"+sound",
"+system_wrappers",
- "+voice_engine",
"+usrsctplib",
"+third_party/libyuv",
]
diff --git a/modules/audio_mixer/DEPS b/modules/audio_mixer/DEPS
index 51608ad..de2271a 100644
--- a/modules/audio_mixer/DEPS
+++ b/modules/audio_mixer/DEPS
@@ -10,5 +10,4 @@
"+modules/rtp_rtcp",
"+modules/utility",
"+system_wrappers",
- "+voice_engine",
]
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index 6bca26d..2fe3cc2 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -155,7 +155,6 @@
"../../system_wrappers:field_trial_api",
"../../test:perf_test",
"../../test:test_support",
- "../../voice_engine",
"../bitrate_controller",
"../congestion_controller",
"../congestion_controller:delay_based_bwe",
diff --git a/modules/remote_bitrate_estimator/DEPS b/modules/remote_bitrate_estimator/DEPS
index d6a2fba..8499c25 100644
--- a/modules/remote_bitrate_estimator/DEPS
+++ b/modules/remote_bitrate_estimator/DEPS
@@ -2,9 +2,3 @@
"+logging/rtc_event_log",
"+system_wrappers",
]
-
-specific_include_rules = {
- "nada\.h": [
- "+voice_engine",
- ],
-}
diff --git a/modules/remote_bitrate_estimator/test/estimators/nada.h b/modules/remote_bitrate_estimator/test/estimators/nada.h
index 590175c..f00a6d3 100644
--- a/modules/remote_bitrate_estimator/test/estimators/nada.h
+++ b/modules/remote_bitrate_estimator/test/estimators/nada.h
@@ -24,7 +24,6 @@
#include "modules/include/module_common_types.h"
#include "modules/remote_bitrate_estimator/test/bwe.h"
#include "rtc_base/constructormagic.h"
-#include "voice_engine/channel.h"
namespace webrtc {
diff --git a/pc/DEPS b/pc/DEPS
index 1992af0..24cb242 100644
--- a/pc/DEPS
+++ b/pc/DEPS
@@ -18,7 +18,6 @@
specific_include_rules = {
"androidtestinitializer\.cc": [
"+base/android", # Allowed only for Android tests.
- "+voice_engine",
],
"srtpfilter_unittest\.cc": [
"+crypto",
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index b57f48d..fa4a8f9 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -111,7 +111,6 @@
"../../api/audio_codecs:builtin_audio_encoder_factory",
"../../modules/audio_processing:audio_processing",
"../../rtc_base:rtc_base_approved",
- "../../voice_engine:voice_engine",
]
}
diff --git a/sdk/android/src/jni/DEPS b/sdk/android/src/jni/DEPS
index db95db6..1cf4ba1 100644
--- a/sdk/android/src/jni/DEPS
+++ b/sdk/android/src/jni/DEPS
@@ -12,5 +12,4 @@
"+modules/video_coding",
"+pc",
"+system_wrappers/include",
- "+voice_engine/include/voe_base.h",
]
diff --git a/test/BUILD.gn b/test/BUILD.gn
index c058b5c..a83e8bb 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -567,7 +567,6 @@
"layer_filtering_transport.cc",
"layer_filtering_transport.h",
"mock_transport.h",
- "mock_voe_channel_proxy.h",
"null_transport.cc",
"null_transport.h",
"rtp_rtcp_observer.h",
@@ -628,7 +627,6 @@
"../system_wrappers",
"../system_wrappers:field_trial_api",
"../video",
- "../voice_engine",
"//testing/gmock",
"//testing/gtest",
]
diff --git a/test/DEPS b/test/DEPS
index 1cf56a8..71d3d25 100644
--- a/test/DEPS
+++ b/test/DEPS
@@ -16,7 +16,6 @@
"+modules/video_coding",
"+sdk",
"+system_wrappers",
- "+voice_engine",
"+third_party/libyuv",
]
diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
index 43df618..0ccabfc 100644
--- a/test/fuzzers/BUILD.gn
+++ b/test/fuzzers/BUILD.gn
@@ -416,9 +416,9 @@
]
deps = [
"../../api:array_view",
+ "../../audio",
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:rtc_base_approved",
- "../../voice_engine",
]
}
rtc_static_library("audio_processing_fuzzer_helper") {
diff --git a/test/fuzzers/DEPS b/test/fuzzers/DEPS
index cdbb566..222cd02 100644
--- a/test/fuzzers/DEPS
+++ b/test/fuzzers/DEPS
@@ -1,3 +1,3 @@
include_rules = [
- "+webrtc",
+ "+audio",
]
diff --git a/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc b/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
index 31bc163..2168d21 100644
--- a/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
+++ b/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
@@ -11,10 +11,10 @@
#include <algorithm>
#include "api/array_view.h"
+#include "audio/transport_feedback_packet_loss_tracker.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
namespace webrtc {
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 31ad45c..8502691 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -94,7 +94,6 @@
"../rtc_base:sequenced_task_checker",
"../rtc_base:weak_ptr",
"../system_wrappers",
- "../voice_engine",
]
if (!build_with_mozilla) {
@@ -130,7 +129,6 @@
"../test:test_support_test_artifacts",
"../test:video_test_common",
"../test:video_test_support",
- "../voice_engine",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
diff --git a/video/DEPS b/video/DEPS
index b94455e..288ecfd 100644
--- a/video/DEPS
+++ b/video/DEPS
@@ -15,5 +15,4 @@
"+modules/video_capture",
"+modules/video_processing",
"+system_wrappers",
- "+voice_engine",
]
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
deleted file mode 100644
index d0de9c6..0000000
--- a/voice_engine/BUILD.gn
+++ /dev/null
@@ -1,148 +0,0 @@
-# Copyright (c) 2014 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.
-
-import("../webrtc.gni")
-
-visibility = [ ":*" ]
-
-rtc_static_library("voice_engine") {
- visibility += [ "*" ]
- sources = [
- "channel.cc",
- "channel.h",
- "channel_proxy.cc",
- "channel_proxy.h",
- "transport_feedback_packet_loss_tracker.cc",
- "transport_feedback_packet_loss_tracker.h",
- "utility.cc",
- "utility.h",
- ]
-
- if (is_win) {
- cflags = [
- # TODO(kjellander): Bug 261: fix this warning.
- "/wd4373", # Virtual function override.
- ]
- }
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-
- deps = [
- ":audio_level",
- "..:webrtc_common",
- "../:typedefs",
- "../api:array_view",
- "../api:audio_mixer_api",
- "../api:call_api",
- "../api:libjingle_peerconnection_api",
- "../api:optional",
- "../api:refcountedbase",
- "../api:transport_api",
- "../api/audio_codecs:audio_codecs_api",
- "../audio/utility:audio_frame_operations",
- "../call:rtp_interfaces",
- "../common_audio",
- "../logging:rtc_event_log_api",
- "../modules:module_api",
- "../modules/audio_coding",
- "../modules/audio_coding:audio_format_conversion",
- "../modules/audio_coding:audio_network_adaptor_config",
- "../modules/audio_device",
- "../modules/audio_processing",
- "../modules/bitrate_controller",
- "../modules/media_file",
- "../modules/pacing",
- "../modules/rtp_rtcp",
- "../modules/rtp_rtcp:rtp_rtcp_format",
- "../modules/utility",
- "../rtc_base:checks",
- "../rtc_base:rate_limiter",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_task_queue",
- "../system_wrappers",
- "../system_wrappers:field_trial_api",
- "../system_wrappers:metrics_api",
- ]
-}
-
-rtc_static_library("audio_level") {
- visibility += [
- ":voice_engine",
- "../audio:audio",
- ]
- sources = [
- "audio_level.cc",
- "audio_level.h",
- ]
-
- deps = [
- "..:webrtc_common",
- "../:typedefs",
- "../common_audio",
- "../modules:module_api",
- "../rtc_base:rtc_base_approved",
- ]
-}
-
-if (rtc_include_tests) {
- rtc_test("voice_engine_unittests") {
- visibility += webrtc_default_visibility
- deps = [
- ":voice_engine",
- "../api/audio_codecs:builtin_audio_decoder_factory",
- "../common_audio",
- "../modules:module_api",
- "../modules/audio_coding",
- "../modules/audio_device",
- "../modules/audio_processing",
- "../modules/media_file",
- "../modules/rtp_rtcp:rtp_rtcp_format",
- "../modules/utility",
- "../modules/video_capture:video_capture",
- "../rtc_base:checks",
- "../rtc_base:rtc_base_approved",
- "../rtc_base:rtc_base_tests_utils",
- "../system_wrappers",
- "../test:test_common",
- "../test:test_main",
- "../test:video_test_common",
- "//testing/gmock",
- "//testing/gtest",
- ]
-
- if (is_android) {
- deps += [ "//testing/android/native_test:native_test_native_code" ]
- shard_timeout = 900
- }
-
- sources = [
- "transport_feedback_packet_loss_tracker_unittest.cc",
- "utility_unittest.cc",
- ]
-
- data = [
- "../resources/utility/encapsulated_pcm16b_8khz.wav",
- "../resources/utility/encapsulated_pcmu_8khz.wav",
- ]
-
- if (is_win) {
- cflags = [
- # TODO(kjellander): Bug 261: fix this warning.
- "/wd4373", # Virtual function override.
- ]
- }
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
- }
-}
diff --git a/voice_engine/DEPS b/voice_engine/DEPS
deleted file mode 100644
index c8e9a1c..0000000
--- a/voice_engine/DEPS
+++ /dev/null
@@ -1,14 +0,0 @@
-include_rules = [
- "+audio/utility/audio_frame_operations.h",
- "+call",
- "+common_audio",
- "+logging/rtc_event_log",
- "+modules/audio_coding",
- "+modules/audio_device",
- "+modules/audio_processing",
- "+modules/media_file",
- "+modules/pacing",
- "+modules/rtp_rtcp",
- "+modules/utility",
- "+system_wrappers",
-]
diff --git a/voice_engine/OWNERS b/voice_engine/OWNERS
deleted file mode 100644
index 0430ede..0000000
--- a/voice_engine/OWNERS
+++ /dev/null
@@ -1,10 +0,0 @@
-henrikg@webrtc.org
-henrika@webrtc.org
-niklas.enbom@webrtc.org
-solenberg@webrtc.org
-
-
-# These are for the common case of adding or renaming files. If you're doing
-# structural changes, please get a review from a reviewer in this file.
-per-file *.gn=*
-per-file *.gni=*