8000 refactor: Create a separate directory to store the third_party librar… · strogo/flutter-webrtc@7b17d65 · GitHub
[go: up one dir, main page]

Skip to content

Commit 7b17d65

Browse files
committed
refactor: Create a separate directory to store the third_party libraries.
1 parent aa5300a commit 7b17d65

40 files changed

+1274
-1274
lines changed

windows/third_party/libwebrtc/include/base/fixed_size_function.h renamed to third_party/libwebrtc/include/base/fixed_size_function.h

Lines changed: 300 additions & 300 deletions
Large diffs are not rendered by default.
Lines changed: 21 additions & 21 deletions
Original file line numberDiff line numberDiff line change
@@ -1,21 +1,21 @@
1-
#ifndef LIB_WEBRTC_HXX
2-
#define LIB_WEBRTC_HXX
3-
4-
#include "rtc_peerconnection_factory.h"
5-
#include "rtc_types.h"
6-
7-
namespace libwebrtc {
8-
9-
class LibWebRTC {
10-
public:
11-
LIB_WEBRTC_API static bool Initialize();
12-
13-
LIB_WEBRTC_API static scoped_refptr<RTCPeerConnectionFactory>
14-
CreateRTCPeerConnectionFactory();
15-
16-
LIB_WEBRTC_API static void Terminate();
17-
};
18-
19-
} // namespace libwebrtc
20-
21-
#endif // LIB_WEBRTC_HXX
1+
#ifndef LIB_WEBRTC_HXX
2+
#define LIB_WEBRTC_HXX
3+
4+
#include "rtc_peerconnection_factory.h"
5+
#include "rtc_types.h"
6+
7+
namespace libwebrtc {
8+
9+
class LibWebRTC {
10+
public:
11+
LIB_WEBRTC_API static bool Initialize();
12+
13+
LIB_WEBRTC_API static scoped_refptr<RTCPeerConnectionFactory>
14+
CreateRTCPeerConnectionFactory();
15+
16+
LIB_WEBRTC_API static void Terminate();
17+
};
18+
19+
} // namespace libwebrtc
20+
21+
#endif // LIB_WEBRTC_HXX
Original file line numberDiff line numberDiff line change
@@ -1,39 +1,39 @@
1-
#ifndef LIB_WEBRTC_RTC_AUDIO_DEVICE_HXX
2-
#define LIB_WEBRTC_RTC_AUDIO_DEVICE_HXX
3-
4-
#include "rtc_types.h"
5-
6-
namespace libwebrtc {
7-
8-
class RTCAudioDevice : public RefCountInterface {
9-
public:
10-
static const int kAdmMaxDeviceNameSize = 128;
11-
static const int kAdmMaxFileNameSize = 512;
12-
static const int kAdmMaxGuidSize = 128;
13-
14-
public:
15-
// Device enumeration
16-
virtual int16_t PlayoutDevices() = 0;
17-
18-
virtual int16_t RecordingDevices() = 0;
19-
20-
virtual int32_t PlayoutDeviceName(uint16_t index,
21-
char name[kAdmMaxDeviceNameSize],
22-
char guid[kAdmMaxGuidSize]) = 0;
23-
24-
virtual int32_t RecordingDeviceName(uint16_t index,
25-
char name[kAdmMaxDeviceNameSize],
26-
char guid[kAdmMaxGuidSize]) = 0;
27-
28-
// Device selection
29-
virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
30-
31-
virtual int32_t SetRecordingDevice(uint16_t index) = 0;
32-
33-
protected:
34-
virtual ~RTCAudioDevice() {}
35-
};
36-
37-
} // namespace libwebrtc
38-
39-
#endif // LIB_WEBRTC_RTC_AUDIO_DEVICE_HXX
1+
#ifndef LIB_WEBRTC_RTC_AUDIO_DEVICE_HXX
2+
#define LIB_WEBRTC_RTC_AUDIO_DEVICE_HXX
3+
4+
#include "rtc_types.h"
5+
6+
namespace libwebrtc {
7+
8+
class RTCAudioDevice : public RefCountInterface {
9+
public:
10+
static const int kAdmMaxDeviceNameSize = 128;
11+
static const int kAdmMaxFileNameSize = 512;
12+
static const int kAdmMaxGuidSize = 128;
13+
14+
public:
15+
// Device enumeration
16+
virtual int16_t PlayoutDevices() = 0;
17+
18+
virtual int16_t RecordingDevices() = 0;
19+
20+
virtual int32_t PlayoutDeviceName(uint16_t index,
21+
char name[kAdmMaxDeviceNameSize],
22+
char guid[kAdmMaxGuidSize]) = 0;
23+
24+
virtual int32_t RecordingDeviceName(uint16_t index,
25+
char name[kAdmMaxDeviceNameSize],
26+
char guid[kAdmMaxGuidSize]) = 0;
27+
28+
// Device selection
29+
virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
30+
31+
virtual int32_t SetRecordingDevice(uint16_t index) = 0;
32+
33+
protected:
34+
virtual ~RTCAudioDevice() {}
35+
};
36+
37+
} // namespace libwebrtc
38+
39+
#endif // LIB_WEBRTC_RTC_AUDIO_DEVICE_HXX
Original file line numberDiff line numberDiff line change
@@ -1,49 +1,49 @@
1-
#ifndef AUDIO_FRAME_HXX
2-
#define AUDIO_FRAME_HXX
3-
4-
#include "media_manager_types.h"
5-
6-
namespace b2bua {
7-
8-
class AudioFrame {
9-
public:
10-
MEDIA_MANAGER_API static AudioFrame* Create();
11-
MEDIA_MANAGER_API static AudioFrame* Create(int id,
12-
uint32_t timestamp,
13-
const int16_t* data,
14-
size_t samples_per_channel,
15-
int sample_rate_hz,
16-
size_t num_channels = 1);
17-
18-
virtual void Release() = 0;
19-
20-
public:
21-
virtual void UpdateFrame(int id,
22-
uint32_t timestamp,
23-
const int16_t* data,
24-
size_t samples_per_channel,
25-
int sample_rate_hz,
26-
size_t num_channels = 1) = 0;
27-
28-
virtual void CopyFrom(const AudioFrame& src) = 0;
29-
30-
virtual void Add(const AudioFrame& frame_to_add) = 0;
31-
32-
virtual void Mute() = 0;
33-
34-
virtual const int16_t* data() = 0;
35-
36-
virtual size_t samples_per_channel() = 0;
37-
38-
virtual int sample_rate_hz() = 0;
39-
40-
virtual size_t num_channels() = 0;
41-
42-
virtual uint32_t timestamp() = 0;
43-
44-
virtual int id() = 0;
45-
};
46-
47-
}; // namespace b2bua
48-
49-
#endif
1+
#ifndef AUDIO_FRAME_HXX
2+
#define AUDIO_FRAME_HXX
3+
4+
#include "media_manager_types.h"
5+
6+
namespace b2bua {
7+
8+
class AudioFrame {
9+
public:
10+
MEDIA_MANAGER_API static AudioFrame* Create();
11+
MEDIA_MANAGER_API static AudioFrame* Create(int id,
12+
uint32_t timestamp,
13+
const int16_t* data,
14+
size_t samples_per_channel,
15+
int sample_rate_hz,
16+
size_t num_channels = 1);
17+
18+
virtual void Release() = 0;
19+
20+
public:
21+
virtual void UpdateFrame(int id,
22+
uint32_t timestamp,
23+
const int16_t* data,
24+
size_t samples_per_channel,
25+
int sample_rate_hz,
26+
size_t num_channels = 1) = 0;
27+
28+
virtual void CopyFrom(const AudioFrame& src) = 0;
29+
30+
virtual void Add(const AudioFrame& frame_to_add) = 0;
31+
32+
virtual void Mute() = 0;
33+
34+
virtual const int16_t* data() = 0;
35+
36+
virtual size_t samples_per_channel() = 0;
37+
38+
virtual int sample_rate_hz() = 0;
39+
40+
virtual size_t num_channels() = 0;
41+
42+
virtual uint32_t timestamp() = 0;
43+
44+
virtual int id() = 0;
45+
};
46+
47+
}; // namespace b2bua
48+
49+
#endif
Original file line numberDiff line numberDiff line change
@@ -1,15 +1,15 @@
1-
#ifndef LIB_WEBRTC_RTC_AUDIO_SOURCE_HXX
2-
#define LIB_WEBRTC_RTC_AUDIO_SOURCE_HXX
3-
4-
#include "rtc_types.h"
5-
6-
namespace libwebrtc {
7-
8-
class RTCAudioSource : public RefCountInterface {
9-
protected:
10-
virtual ~RTCAudioSource() {}
11-
};
12-
13-
} // namespace libwebrtc
14-
15-
#endif // LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
1+
#ifndef LIB_WEBRTC_RTC_AUDIO_SOURCE_HXX
2+
#define LIB_WEBRTC_RTC_AUDIO_SOURCE_HXX
3+
4+
#include "rtc_types.h"
5+
6+
namespace libwebrtc {
7+
8+
class RTCAudioSource : public RefCountInterface {
9+
protected:
10+
virtual ~RTCAudioSource() {}
11+
};
12+
13+
} // namespace libwebrtc
14+
15+
#endif // LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
Original file line numberDiff line numberDiff line change
@@ -1,15 +1,15 @@
1-
#ifndef LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
2-
#define LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
3-
4-
#include "rtc_media_track.h"
5-
#include "rtc_types.h"
6-
7-
namespace libwebrtc {
8-
9-
class RTCAudioTrack : public RTCMediaTrack {
10-
protected:
11-
virtual ~RTCAudioTrack() {}
12-
};
13-
} // namespace libwebrtc
14-
15-
#endif // LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
1+
#ifndef LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
2+
#define LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
3+
4+
#include "rtc_media_track.h"
5+
#include "rtc_types.h"
6+
7+
namespace libwebrtc {
8+
9+
class RTCAudioTrack : public RTCMediaTrack {
10+
protected:
11+
virtual ~RTCAudioTrack() {}
12+
};
13+
} // namespace libwebrtc
14+
15+
#endif // LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
Original file line numberDiff line numberDiff line change
@@ -1,57 +1,57 @@
1-
#ifndef LIB_WEBRTC_RTC_DATA_CHANNEL_HXX
2-
#define LIB_WEBRTC_RTC_DATA_CHANNEL_HXX
3-
4-
#include "rtc_types.h"
5-
6-
namespace libwebrtc {
7-
8-
enum RTCDataChannelState {
9-
RTCDataChannelConnecting,
10-
RTCDataChannelOpen,
11-
RTCDataChannelClosing,
12-
RTCDataChannelClosed,
13-
};
14-
15-
struct RTCDataChannelInit {
16-
bool ordered = true;
17-
bool reliable = true;
18-
int maxRetransmitTime = -1;
19-
int maxRetransmits = -1;
20-
string protocol = {"sctp"}; // sctp | quic
21-
bool negotiated = false;
22-
int id = 0;
23-
};
24-
25-
class RTCDataChannelObserver {
26-
public:
27-
virtual void OnStateChange(RTCDataChannelState state) = 0;
28-
29-
virtual void OnMessage(const char* buffer, int length, bool binary) = 0;
30-
31-
protected:
32-
virtual ~RTCDataChannelObserver() = default;
33-
};
34-
35-
class RTCDataChannel : public RefCountInterface {
36-
public:
37-
virtual void Send(const string data, bool binary = false) = 0;
38-
39-
virtual void Close() = 0;
40-
41-
virtual void RegisterObserver(RTCDataChannelObserver* observer) = 0;
42-
43-
virtual void UnregisterObserver() = 0;
44-
45-
virtual const string label() const = 0;
46-
47-
virtual int id() const = 0;
48-
49-
virtual RTCDataChannelState state() = 0;
50-
51-
protected:
52-
virtual ~RTCDataChannel() {}
53-
};
54-
55-
} // namespace libwebrtc
56-
57-
#endif // LIB_WEBRTC_RTC_DATA_CHANNEL_HXX
1+
#ifndef LIB_WEBRTC_RTC_DATA_CHANNEL_HXX
2+
#define LIB_WEBRTC_RTC_DATA_CHANNEL_HXX
3+
4+
#include "rtc_types.h"
5+
6+
namespace libwebrtc {
7+
8+
enum RTCDataChannelState {
9+
RTCDataChannelConnecting,
10+
RTCDataChannelOpen,
11+
RTCDataChannelClosing,
12+
RTCDataChannelClosed,
13+
};
14+
15+
struct RTCDataChannelInit {
16+
bool ordered = true;
17+
bool reliable = true;
18+
int maxRetransmitTime = -1;
19+
int maxRetransmits = -1;
20+
string protocol = {"sctp"}; // sctp | quic
21+
bool negotiated = false;
22+
int id = 0;
23+
};
24+
25+
class RTCDataChannelObserver {
26+
public:
27+
virtual void OnStateChange(RTCDataChannelState state) = 0;
28+
29+
virtual void OnMessage(const char* buffer, int length, bool binary) = 0;
30+
31+
protected:
32+
virtual ~RTCDataChannelObserver() = default;
33+
};
34+
35+
class RTCDataChannel : public RefCountInterface {
36+
public:
37+
virtual void Send(const string data, bool binary = false) = 0;
38+
39+
virtual void Close() = 0;
40+
41+
virtual void RegisterObserver(RTCDataChannelObserver* observer) = 0;
42+
43+
virtual void UnregisterObserver() = 0;
44+
45+
virtual const string label() const = 0;
46+
47+
virtual int id() const = 0;
48+
49+
virtual RTCDataChannelState state() = 0;
50+
51+
protected:
52+
virtual ~RTCDataChannel() {}
53+
};
54+
55+
} // namespace libwebrtc
56+
57+
#endif // LIB_WEBRTC_RTC_DATA_CHANNEL_HXX

0 commit comments

Comments
 (0)
0