Skip to content

Commit bc67818

Browse files
committed
Update WebRTC impl to support latest branch_heads/66
1 parent 48b06a1 commit bc67818

34 files changed

+229
-586
lines changed

BUILD.md

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -78,7 +78,7 @@ cmake .. -DCMAKE_BUILD_TYPE=DEBUG -DBUILD_SHARED_LIBS=OFF \
7878
-DBUILD_MODULE_json=ON -DBUILD_MODULE_net=ON \
7979
-DBUILD_MODULE_socketio=ON -DBUILD_MODULE_symple=ON \
8080
-DBUILD_MODULE_util=ON -DBUILD_MODULE_webrtc=ON \
81-
-DWEBRTC_ROOT_DIR=/home/kam/sourcey/webrtcbuilds/out/webrtc-17657-02ba69d-linux-x64
81+
-DWEBRTC_ROOT_DIR=/home/kam/sourcey/webrtcbuilds/out/webrtc-22215-ab42706-linux-x64
8282

8383
## WebRTC (VS2015, Win64, Static)
8484
cmake -G "Visual Studio 14 Win64" .. -DCMAKE_BUILD_TYPE=DEBUG -DBUILD_SHARED_LIBS=OFF -DBUILD_MODULES=OFF -DBUILD_WITH_STATIC_CRT=ON -DBUILD_APPLICATIONS=ON -DBUILD_SAMPLES=ON -DBUILD_TESTS=ON -DWITH_WEBRTC=ON -DWITH_FFMPEG=ON -DBUILD_MODULE_base=ON -DBUILD_MODULE_crypto=ON -DBUILD_MODULE_http=ON -DBUILD_MODULE_json=ON -DBUILD_MODULE_av=ON -DBUILD_MODULE_net=ON -DBUILD_MODULE_socketio=ON -DBUILD_MODULE_symple=ON -DBUILD_MODULE_util=ON -DBUILD_MODULE_stun=ON -DBUILD_MODULE_turn=ON -DBUILD_MODULE_webrtc=ON -DFFMPEG_ROOT_DIR=E:/dev/vendor/ffmpeg-3.2.2-win64-dev -DWEBRTC_ROOT_DIR=E:/dev/vendor/webrtcbuilds/out/webrtc-17657-02ba69d-win-x64

circle.yml

Lines changed: 2 additions & 2 deletions
Original file line numberDiff line numberDiff line change
@@ -10,9 +10,9 @@ dependencies:
1010
- sudo apt-get install --yes alsa-base alsa-utils libasound2 libasound2-dev
1111
- sudo apt-get install --yes gcc-5 g++-5
1212
- sudo update-alternatives --install /usr/bin/gcc gcc /usr/bin/gcc-5 60 --slave /usr/bin/g++ g++ /usr/bin/g++-5
13-
- mkdir -p /tmp/webrtc-17657-02ba69d-linux-x64; curl -sSL https://github.com/sourcey/webrtc-precompiled-builds/raw/master/webrtc-17657-02ba69d-linux-x64.tar.gz | sudo tar -xzC /tmp/webrtc-17657-02ba69d-linux-x64
13+
- mkdir -p /tmp/webrtc-22215-ab42706-linux-x64; curl -sSL https://github.com/sourcey/webrtc-precompiled-builds/raw/master/webrtc-22215-ab42706-linux-x64.tar.gz | sudo tar -xzC /tmp/webrtc-22215-ab42706-linux-x64
1414
override:
15-
- mkdir build; cd build; /opt/cmake-3.4.0-Linux-x86_64/bin/cmake .. -DWITH_FFMPEG=ON -DWITH_WEBRTC=ON -DWEBRTC_ROOT_DIR=/tmp/webrtc-17657-02ba69d-linux-x64
15+
- mkdir build; cd build; /opt/cmake-3.4.0-Linux-x86_64/bin/cmake .. -DWITH_FFMPEG=ON -DWITH_WEBRTC=ON -DWEBRTC_ROOT_DIR=/tmp/webrtc-22215-ab42706-linux-x64
1616
- cd build; make; sudo make install
1717
test:
1818
override:

cmake/FindWebRTC.cmake

Lines changed: 10 additions & 5 deletions
Original file line numberDiff line numberDiff line change
@@ -33,7 +33,8 @@ set(WEBRTC_ROOT_DIR "" CACHE STRING "Where is the WebRTC root directory located?
3333
# ----------------------------------------------------------------------
3434
find_path(WEBRTC_INCLUDE_DIR
3535
NAMES
36-
webrtc/config.h
36+
# webrtc/config.h
37+
call/rtp_config.h
3738
PATHS
3839
${WEBRTC_ROOT_DIR}
3940
${WEBRTC_ROOT_DIR}/include
@@ -93,7 +94,7 @@ if(WEBRTC_INCLUDE_DIR)
9394
#unset(WEBRTC_LIBRARY CACHE)
9495
#unset(WEBRTC_LIBRARIES CACHE)
9596
#set(WEBRTC_LIBRARIES "")
96-
97+
9798
if(MSVC)
9899
set(lib_suffix "lib")
99100
else()
@@ -162,7 +163,7 @@ if(WEBRTC_INCLUDE_DIR)
162163
endif()
163164

164165
# Add vendor include directories
165-
if(WEBRTC_INCLUDE_DIR)
166+
if(WEBRTC_INCLUDE_DIR AND NOT WEBRTC_INCLUDE_DIRS)
166167
list(APPEND WEBRTC_INCLUDE_DIRS ${WEBRTC_INCLUDE_DIR} ${WEBRTC_INCLUDE_DIR}/third_party/boringssl/src/include)
167168
endif()
168169

@@ -181,9 +182,13 @@ endif()
181182
# Display status
182183
# ----------------------------------------------------------------------
183184
include(FindPackageHandleStandardArgs)
184-
find_package_handle_standard_args(WEBRTC DEFAULT_MSG WEBRTC_LIBRARIES WEBRTC_INCLUDE_DIR)
185+
message("WEBRTC_INCLUDE_DIR: ${WEBRTC_INCLUDE_DIR}")
186+
message("WEBRTC_INCLUDE_DIRS: ${WEBRTC_INCLUDE_DIRS}")
187+
188+
set(WEBRTC_FOUND TRUE)
189+
find_package_handle_standard_args(WEBRTC DEFAULT_MSG WEBRTC_LIBRARIES WEBRTC_INCLUDE_DIRS)
185190

186-
# print_module_variables(WEBRTC)
191+
print_module_variables(WEBRTC)
187192

188193
# HACK: WEBRTC_LIBRARIES and WEBRTC_DEPENDENCIES not propagating to parent scope
189194
# while the WEBRTC_DEBUG_LIBRARY and WEBRTC_RELEASE_LIBRARY vars are.

src/webrtc/deprecated/include/scy/webrtc/ffmpegvideobuffer.h

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -15,7 +15,7 @@
1515

1616
#include "scy/av/types.h"
1717

18-
#include "webrtc/media/base/videocapturer.h"
18+
#include "media/base/videocapturer.h"
1919

2020

2121
namespace scy {

src/webrtc/deprecated/include/scy/webrtc/imagesequencerecorder.h

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -15,7 +15,7 @@
1515

1616
#include "scy/av/videoencoder.h"
1717

18-
#include "webrtc/api/peerconnectioninterface.h"
18+
#include "api/peerconnectioninterface.h"
1919

2020

2121
namespace scy {

src/webrtc/deprecated/include/scy/webrtc/opencvvideocapturer.h

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -27,7 +27,7 @@
2727
#include "scy/av/mediafactory.h"
2828
#include "scy/av/videocapture.h"
2929

30-
#include "webrtc/media/base/videocapturer.h"
30+
#include "media/base/videocapturer.h"
3131

3232

3333
namespace scy {

src/webrtc/deprecated/src/imagesequencerecorder.cpp

Lines changed: 3 additions & 3 deletions
Original file line numberDiff line numberDiff line change
@@ -15,9 +15,9 @@
1515
#include "scy/filesystem.h"
1616
#include "scy/logger.h"
1717

18-
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
19-
#include "webrtc/media/engine/webrtcvideocapturerfactory.h"
20-
#include "webrtc/modules/video_capture/video_capture_factory.h"
18+
#include "common_video/libyuv/include/webrtc_libyuv.h"
19+
#include "media/engine/webrtcvideocapturerfactory.h"
20+
#include "modules/video_capture/video_capture_factory.h"
2121

2222

2323
namespace scy {

src/webrtc/include/scy/webrtc/audiopacketmodule.h

Lines changed: 39 additions & 39 deletions
Original file line numberDiff line numberDiff line change
@@ -22,13 +22,13 @@
2222
#include "scy/av/packet.h"
2323
#include "scy/packetsignal.h"
2424

25-
#include "webrtc/base/basictypes.h"
26-
#include "webrtc/base/criticalsection.h"
27-
#include "webrtc/base/messagehandler.h"
28-
#include "webrtc/base/scoped_ref_ptr.h"
29-
#include "webrtc/base/thread.h"
30-
#include "webrtc/common_types.h"
31-
#include "webrtc/modules/audio_device/include/audio_device.h"
25+
#include "rtc_base/basictypes.h"
26+
#include "rtc_base/criticalsection.h"
27+
#include "rtc_base/messagehandler.h"
28+
#include "rtc_base/scoped_ref_ptr.h"
29+
#include "rtc_base/thread.h"
30+
#include "common_types.h"
31+
#include "modules/audio_device/include/audio_device.h"
3232

3333

3434
namespace scy {
@@ -56,13 +56,13 @@ class AudioPacketModule : public webrtc::AudioDeviceModule,
5656
/// Only functions called by `Peer` are implemented, the rest do
5757
/// nothing and return success. If a function is not expected to be called
5858
/// by `Peer` an assertion is triggered if it is in fact called.
59-
int64_t TimeUntilNextProcess() override;
60-
void Process() override;
59+
// int64_t TimeUntilNextProcess() override;
60+
// void Process() override;
6161

6262
int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
6363

64-
ErrorCode LastError() const override;
65-
int32_t RegisterEventObserver(webrtc::AudioDeviceObserver* event_callback) override;
64+
// ErrorCode LastError() const override;
65+
// int32_t RegisterEventObserver(webrtc::AudioDeviceObserver* event_callback) override;
6666

6767
/// Note: Calling this method from a callback may result in deadlock.
6868
int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback) override;
@@ -99,11 +99,11 @@ class AudioPacketModule : public webrtc::AudioDeviceModule,
9999
int32_t StopRecording() override;
100100
bool Recording() const override;
101101

102-
int32_t SetAGC(bool enable) override;
103-
bool AGC() const override;
102+
// int32_t SetAGC(bool enable) override;
103+
// bool AGC() const override;
104104

105-
int32_t SetWaveOutVolume(uint16_t volume_left, uint16_t volume_right) override;
106-
int32_t WaveOutVolume(uint16_t* volume_left, uint16_t* volume_right) const override;
105+
// int32_t SetWaveOutVolume(uint16_t volume_left, uint16_t volume_right) override;
106+
// int32_t WaveOutVolume(uint16_t* volume_left, uint16_t* volume_right) const override;
107107

108108
int32_t InitSpeaker() override;
109109
bool SpeakerIsInitialized() const override;
@@ -115,15 +115,15 @@ class AudioPacketModule : public webrtc::AudioDeviceModule,
115115
int32_t SpeakerVolume(uint32_t* volume) const override;
116116
int32_t MaxSpeakerVolume(uint32_t* max_volume) const override;
117117
int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
118-
int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
118+
// int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
119119

120120
int32_t MicrophoneVolumeIsAvailable(bool* available) override;
121121
int32_t SetMicrophoneVolume(uint32_t volume) override;
122122
int32_t MicrophoneVolume(uint32_t* volume) const override;
123123
int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
124124

125125
int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
126-
int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
126+
// int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
127127

128128
int32_t SpeakerMuteIsAvailable(bool* available) override;
129129
int32_t SetSpeakerMute(bool enable) override;
@@ -133,41 +133,41 @@ class AudioPacketModule : public webrtc::AudioDeviceModule,
133133
int32_t SetMicrophoneMute(bool enable) override;
134134
int32_t MicrophoneMute(bool* enabled) const override;
135135

136-
int32_t MicrophoneBoostIsAvailable(bool* available) override;
137-
int32_t SetMicrophoneBoost(bool enable) override;
138-
int32_t MicrophoneBoost(bool* enabled) const override;
136+
// int32_t MicrophoneBoostIsAvailable(bool* available) override;
137+
// int32_t SetMicrophoneBoost(bool enable) override;
138+
// int32_t MicrophoneBoost(bool* enabled) const override;
139139

140140
int32_t StereoPlayoutIsAvailable(bool* available) const override;
141141
int32_t SetStereoPlayout(bool enable) override;
142142
int32_t StereoPlayout(bool* enabled) const override;
143143
int32_t StereoRecordingIsAvailable(bool* available) const override;
144144
int32_t SetStereoRecording(bool enable) override;
145145
int32_t StereoRecording(bool* enabled) const override;
146-
int32_t SetRecordingChannel(const ChannelType channel) override;
147-
int32_t RecordingChannel(ChannelType* channel) const override;
146+
// int32_t SetRecordingChannel(const ChannelType channel) override;
147+
// int32_t RecordingChannel(ChannelType* channel) const override;
148148

149-
int32_t SetPlayoutBuffer(const BufferType type, uint16_t size_ms = 0) override;
150-
int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
149+
// int32_t SetPlayoutBuffer(const BufferType type, uint16_t size_ms = 0) override;
150+
// int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
151151
int32_t PlayoutDelay(uint16_t* delay_ms) const override;
152-
int32_t RecordingDelay(uint16_t* delay_ms) const override;
152+
// int32_t RecordingDelay(uint16_t* delay_ms) const override;
153153

154-
int32_t CPULoad(uint16_t* load) const override;
154+
// int32_t CPULoad(uint16_t* load) const override;
155155

156-
int32_t StartRawOutputFileRecording(
157-
const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
158-
int32_t StopRawOutputFileRecording() override;
159-
int32_t StartRawInputFileRecording(
160-
const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
161-
int32_t StopRawInputFileRecording() override;
156+
// int32_t StartRawOutputFileRecording(
157+
// const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
158+
// int32_t StopRawOutputFileRecording() override;
159+
// int32_t StartRawInputFileRecording(
160+
// const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
161+
// int32_t StopRawInputFileRecording() override;
162162

163-
int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
164-
int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
165-
int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
166-
int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
163+
// int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
164+
// int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
165+
// int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
166+
// int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
167167

168-
int32_t ResetAudioDevice() override;
169-
int32_t SetLoudspeakerStatus(bool enable) override;
170-
int32_t GetLoudspeakerStatus(bool* enabled) const override;
168+
// int32_t ResetAudioDevice() override;
169+
// int32_t SetLoudspeakerStatus(bool enable) override;
170+
// int32_t GetLoudspeakerStatus(bool* enabled) const override;
171171
bool BuiltInAECIsAvailable() const override { return false; }
172172
int32_t EnableBuiltInAEC(bool enable) override { return -1; }
173173
bool BuiltInAGCIsAvailable() const override { return false; }

src/webrtc/include/scy/webrtc/fakeaudiodevicemodule.h

Lines changed: 10 additions & 48 deletions
Original file line numberDiff line numberDiff line change
@@ -13,13 +13,13 @@
1313

1414
#include "scy/base.h"
1515

16-
#include "webrtc/base/basictypes.h"
17-
#include "webrtc/base/criticalsection.h"
18-
#include "webrtc/base/messagehandler.h"
19-
#include "webrtc/base/scoped_ref_ptr.h"
20-
#include "webrtc/base/thread.h"
21-
#include "webrtc/common_types.h"
22-
#include "webrtc/modules/audio_device/include/audio_device.h"
16+
#include "rtc_base/basictypes.h"
17+
#include "rtc_base/criticalsection.h"
18+
#include "rtc_base/messagehandler.h"
19+
#include "rtc_base/scoped_ref_ptr.h"
20+
#include "rtc_base/thread.h"
21+
#include "common_types.h"
22+
#include "modules/audio_device/include/audio_device.h"
2323

2424

2525
namespace scy {
@@ -38,14 +38,11 @@ class FakeAudioDeviceModule : public webrtc::AudioDeviceModule,
3838
/// Only functions called by `Peer` are implemented, the rest do
3939
/// nothing and return success. If a function is not expected to be called
4040
/// by `Peer` an assertion is triggered if it is in fact called.
41-
int64_t TimeUntilNextProcess() override;
42-
void Process() override;
41+
// int64_t TimeUntilNextProcess() override;
42+
// void Process() override;
4343

4444
int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
4545

46-
ErrorCode LastError() const override;
47-
int32_t RegisterEventObserver(webrtc::AudioDeviceObserver* event_callback) override;
48-
4946
/// Note: Calling this method from a callback may result in deadlock.
5047
int32_t RegisterAudioCallback(webrtc::AudioTransport* audio_callback) override;
5148

@@ -81,12 +78,6 @@ class FakeAudioDeviceModule : public webrtc::AudioDeviceModule,
8178
int32_t StopRecording() override;
8279
bool Recording() const override;
8380

84-
int32_t SetAGC(bool enable) override;
85-
bool AGC() const override;
86-
87-
int32_t SetWaveOutVolume(uint16_t volume_left, uint16_t volume_right) override;
88-
int32_t WaveOutVolume(uint16_t* volume_left, uint16_t* volume_right) const override;
89-
9081
int32_t InitSpeaker() override;
9182
bool SpeakerIsInitialized() const override;
9283
int32_t InitMicrophone() override;
@@ -97,15 +88,12 @@ class FakeAudioDeviceModule : public webrtc::AudioDeviceModule,
9788
int32_t SpeakerVolume(uint32_t* volume) const override;
9889
int32_t MaxSpeakerVolume(uint32_t* max_volume) const override;
9990
int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
100-
int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
10191

10292
int32_t MicrophoneVolumeIsAvailable(bool* available) override;
10393
int32_t SetMicrophoneVolume(uint32_t volume) override;
10494
int32_t MicrophoneVolume(uint32_t* volume) const override;
105-
int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
106-
10795
int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
108-
int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
96+
int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
10997

11098
int32_t SpeakerMuteIsAvailable(bool* available) override;
11199
int32_t SetSpeakerMute(bool enable) override;
@@ -115,41 +103,15 @@ class FakeAudioDeviceModule : public webrtc::AudioDeviceModule,
115103
int32_t SetMicrophoneMute(bool enable) override;
116104
int32_t MicrophoneMute(bool* enabled) const override;
117105

118-
int32_t MicrophoneBoostIsAvailable(bool* available) override;
119-
int32_t SetMicrophoneBoost(bool enable) override;
120-
int32_t MicrophoneBoost(bool* enabled) const override;
121-
122106
int32_t StereoPlayoutIsAvailable(bool* available) const override;
123107
int32_t SetStereoPlayout(bool enable) override;
124108
int32_t StereoPlayout(bool* enabled) const override;
125109
int32_t StereoRecordingIsAvailable(bool* available) const override;
126110
int32_t SetStereoRecording(bool enable) override;
127111
int32_t StereoRecording(bool* enabled) const override;
128-
int32_t SetRecordingChannel(const ChannelType channel) override;
129-
int32_t RecordingChannel(ChannelType* channel) const override;
130112

131-
int32_t SetPlayoutBuffer(const BufferType type, uint16_t size_ms = 0) override;
132-
int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
133113
int32_t PlayoutDelay(uint16_t* delay_ms) const override;
134-
int32_t RecordingDelay(uint16_t* delay_ms) const override;
135-
136-
int32_t CPULoad(uint16_t* load) const override;
137-
138-
int32_t StartRawOutputFileRecording(
139-
const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
140-
int32_t StopRawOutputFileRecording() override;
141-
int32_t StartRawInputFileRecording(
142-
const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
143-
int32_t StopRawInputFileRecording() override;
144-
145-
int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
146-
int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
147-
int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
148-
int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
149114

150-
int32_t ResetAudioDevice() override;
151-
int32_t SetLoudspeakerStatus(bool enable) override;
152-
int32_t GetLoudspeakerStatus(bool* enabled) const override;
153115
bool BuiltInAECIsAvailable() const override { return false; }
154116
int32_t EnableBuiltInAEC(bool enable) override { return -1; }
155117
bool BuiltInAGCIsAvailable() const override { return false; }

src/webrtc/include/scy/webrtc/multiplexmediacapturer.h

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -22,7 +22,7 @@
2222
#include "scy/webrtc/audiopacketmodule.h"
2323
#include "scy/webrtc/videopacketsource.h"
2424

25-
#include "webrtc/api/peerconnectioninterface.h"
25+
#include "api/peerconnectioninterface.h"
2626

2727

2828
namespace scy {

src/webrtc/include/scy/webrtc/peer.h

Lines changed: 6 additions & 6 deletions
Original file line numberDiff line numberDiff line change
@@ -15,10 +15,10 @@
1515

1616
#include "scy/webrtc/peerfactorycontext.h"
1717

18-
#include "webrtc/api/jsep.h"
19-
#include "webrtc/api/peerconnectioninterface.h"
20-
#include "webrtc/api/test/fakeconstraints.h"
21-
#include "webrtc/p2p/client/basicportallocator.h"
18+
#include "api/jsep.h"
19+
#include "api/peerconnectioninterface.h"
20+
#include "api/test/fakeconstraints.h"
21+
#include "p2p/client/basicportallocator.h"
2222

2323

2424
namespace scy {
@@ -99,8 +99,8 @@ class Peer : public webrtc::PeerConnectionObserver,
9999
virtual void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
100100
virtual void OnFailure(const std::string& error) override;
101101

102-
virtual int AddRef() const override { return 1; }
103-
virtual int Release() const override { return 0; }
102+
virtual void AddRef() const override { return; }
103+
virtual rtc::RefCountReleaseStatus Release() const override { return rtc::RefCountReleaseStatus::kDroppedLastRef; }
104104

105105
protected:
106106
PeerManager* _manager;

src/webrtc/include/scy/webrtc/peerfactorycontext.h

Lines changed: 1 addition & 1 deletion
Original file line numberDiff line numberDiff line change
@@ -15,7 +15,7 @@
1515

1616
#include "scy/webrtc/webrtc.h"
1717

18-
#include "webrtc/pc/peerconnectionfactory.h"
18+
#include "pc/peerconnectionfactory.h"
1919

2020

2121
namespace scy {

0 commit comments

Comments
 (0)