Remove chromium clang style errors affecting sdk/android/media_jni
Bug: webrtc:163
Change-Id: I1e98174817ca032ee13f9a6a386803382843389d
Reviewed-on: https://webrtc-review.googlesource.com/67360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22796}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 1a2ae4b..fddea97 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -189,6 +189,7 @@
rtc_source_set("audio_options_api") {
visibility = [ "*" ]
sources = [
+ "audio_options.cc",
"audio_options.h",
]
diff --git a/api/audio_options.cc b/api/audio_options.cc
new file mode 100644
index 0000000..c196d7d
--- /dev/null
+++ b/api/audio_options.cc
@@ -0,0 +1,18 @@
+/*
+ * Copyright (c) 2018 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.
+ */
+
+#include "api/audio_options.h"
+
+namespace cricket {
+
+AudioOptions::AudioOptions() = default;
+AudioOptions::~AudioOptions() = default;
+
+} // namespace cricket
diff --git a/api/audio_options.h b/api/audio_options.h
index 8d2880b..5d69842 100644
--- a/api/audio_options.h
+++ b/api/audio_options.h
@@ -23,6 +23,8 @@
// We are moving all of the setting of options to structs like this,
// but some things currently still use flags.
struct AudioOptions {
+ AudioOptions();
+ ~AudioOptions();
void SetAll(const AudioOptions& change) {
SetFrom(&echo_cancellation, change.echo_cancellation);
#if defined(WEBRTC_IOS)
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 58238af..3d6f32f 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -10,10 +10,15 @@
rtc_source_set("call_interfaces") {
sources = [
+ "audio_receive_stream.cc",
"audio_receive_stream.h",
"audio_send_stream.h",
+ "audio_state.cc",
"audio_state.h",
"call.h",
+ "call_config.cc",
+ "call_config.h",
+ "flexfec_receive_stream.cc",
"flexfec_receive_stream.h",
"syncable.cc",
"syncable.h",
@@ -32,6 +37,8 @@
"../api:transport_api",
"../api/audio:audio_mixer_api",
"../api/audio_codecs:audio_codecs_api",
+ "../modules/audio_device:audio_device",
+ "../modules/audio_processing:audio_processing",
"../modules/audio_processing:audio_processing_statistics",
"../rtc_base:audio_format_to_string",
"../rtc_base:rtc_base",
diff --git a/call/audio_receive_stream.cc b/call/audio_receive_stream.cc
new file mode 100644
index 0000000..c3c2ac7
--- /dev/null
+++ b/call/audio_receive_stream.cc
@@ -0,0 +1,24 @@
+/*
+ * Copyright (c) 2018 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.
+ */
+
+#include "call/audio_receive_stream.h"
+
+namespace webrtc {
+
+AudioReceiveStream::Stats::Stats() = default;
+AudioReceiveStream::Stats::~Stats() = default;
+
+AudioReceiveStream::Config::Config() = default;
+AudioReceiveStream::Config::~Config() = default;
+
+AudioReceiveStream::Config::Rtp::Rtp() = default;
+AudioReceiveStream::Config::Rtp::~Rtp() = default;
+
+} // namespace webrtc
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 54a4d9b..3c0e58d 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -32,6 +32,8 @@
class AudioReceiveStream {
public:
struct Stats {
+ Stats();
+ ~Stats();
uint32_t remote_ssrc = 0;
int64_t bytes_rcvd = 0;
uint32_t packets_rcvd = 0;
@@ -71,10 +73,16 @@
};
struct Config {
+ Config();
+ ~Config();
+
std::string ToString() const;
// Receive-stream specific RTP settings.
struct Rtp {
+ Rtp();
+ ~Rtp();
+
std::string ToString() const;
// Synchronization source (stream identifier) to be received.
diff --git a/call/audio_state.cc b/call/audio_state.cc
new file mode 100644
index 0000000..725d27f
--- /dev/null
+++ b/call/audio_state.cc
@@ -0,0 +1,18 @@
+/*
+ * Copyright (c) 2018 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.
+ */
+
+#include "call/audio_state.h"
+
+namespace webrtc {
+
+AudioState::Config::Config() = default;
+AudioState::Config::~Config() = default;
+
+} // namespace webrtc
diff --git a/call/audio_state.h b/call/audio_state.h
index e947beb..a85cd86 100644
--- a/call/audio_state.h
+++ b/call/audio_state.h
@@ -11,13 +11,13 @@
#define CALL_AUDIO_STATE_H_
#include "api/audio/audio_mixer.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
-class AudioDeviceModule;
-class AudioProcessing;
class AudioTransport;
// AudioState holds the state which must be shared between multiple instances of
@@ -25,6 +25,9 @@
class AudioState : public rtc::RefCountInterface {
public:
struct Config {
+ Config();
+ ~Config();
+
// The audio mixer connected to active receive streams. One per
// AudioState.
rtc::scoped_refptr<AudioMixer> audio_mixer;
@@ -65,7 +68,7 @@
static rtc::scoped_refptr<AudioState> Create(
const AudioState::Config& config);
- virtual ~AudioState() {}
+ ~AudioState() override {}
};
} // namespace webrtc
diff --git a/call/call.h b/call/call.h
index 8630815..d2971be 100644
--- a/call/call.h
+++ b/call/call.h
@@ -15,12 +15,9 @@
#include <string>
#include <vector>
-#include "api/fec_controller.h"
-#include "api/rtcerror.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
-#include "call/audio_state.h"
-#include "call/bitrate_constraints.h"
+#include "call/call_config.h"
#include "call/flexfec_receive_stream.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "call/video_receive_stream.h"
@@ -29,14 +26,10 @@
#include "rtc_base/bitrateallocationstrategy.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/networkroute.h"
-#include "rtc_base/platform_file.h"
#include "rtc_base/socket.h"
namespace webrtc {
-class AudioProcessing;
-class RtcEventLog;
-
enum class MediaType {
ANY,
AUDIO,
@@ -60,33 +53,6 @@
virtual ~PacketReceiver() {}
};
-struct CallConfig {
- explicit CallConfig(RtcEventLog* event_log) : event_log(event_log) {
- RTC_DCHECK(event_log);
- }
-
- RTC_DEPRECATED static constexpr int kDefaultStartBitrateBps = 300000;
-
- // Bitrate config used until valid bitrate estimates are calculated. Also
- // used to cap total bitrate used. This comes from the remote connection.
- BitrateConstraints bitrate_config;
-
- // AudioState which is possibly shared between multiple calls.
- // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
- rtc::scoped_refptr<AudioState> audio_state;
-
- // Audio Processing Module to be used in this call.
- // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
- AudioProcessing* audio_processing = nullptr;
-
- // RtcEventLog to use for this call. Required.
- // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
- RtcEventLog* event_log = nullptr;
-
- // FecController to use for this call.
- FecControllerFactoryInterface* fec_controller_factory = nullptr;
-};
-
// A Call instance can contain several send and/or receive streams. All streams
// are assumed to have the same remote endpoint and will share bitrate estimates
// etc.
diff --git a/call/call_config.cc b/call/call_config.cc
new file mode 100644
index 0000000..ca5fb60
--- /dev/null
+++ b/call/call_config.cc
@@ -0,0 +1,20 @@
+/*
+ * Copyright (c) 2018 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.
+ */
+
+#include "call/call_config.h"
+
+namespace webrtc {
+
+CallConfig::CallConfig(RtcEventLog* event_log) : event_log(event_log) {
+ RTC_DCHECK(event_log);
+}
+CallConfig::~CallConfig() = default;
+
+} // namespace webrtc
diff --git a/call/call_config.h b/call/call_config.h
new file mode 100644
index 0000000..421b524
--- /dev/null
+++ b/call/call_config.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (c) 2018 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.
+ */
+#ifndef CALL_CALL_CONFIG_H_
+#define CALL_CALL_CONFIG_H_
+
+#include "api/fec_controller.h"
+#include "api/rtcerror.h"
+#include "call/audio_state.h"
+#include "call/bitrate_constraints.h"
+#include "rtc_base/platform_file.h"
+
+namespace webrtc {
+
+class AudioProcessing;
+class RtcEventLog;
+
+struct CallConfig {
+ explicit CallConfig(RtcEventLog* event_log);
+ ~CallConfig();
+
+ RTC_DEPRECATED static constexpr int kDefaultStartBitrateBps = 300000;
+
+ // Bitrate config used until valid bitrate estimates are calculated. Also
+ // used to cap total bitrate used. This comes from the remote connection.
+ BitrateConstraints bitrate_config;
+
+ // AudioState which is possibly shared between multiple calls.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ rtc::scoped_refptr<AudioState> audio_state;
+
+ // Audio Processing Module to be used in this call.
+ // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
+ AudioProcessing* audio_processing = nullptr;
+
+ // RtcEventLog to use for this call. Required.
+ // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
+ RtcEventLog* event_log = nullptr;
+
+ // FecController to use for this call.
+ FecControllerFactoryInterface* fec_controller_factory = nullptr;
+};
+
+} // namespace webrtc
+
+#endif // CALL_CALL_CONFIG_H_
diff --git a/call/flexfec_receive_stream.cc b/call/flexfec_receive_stream.cc
new file mode 100644
index 0000000..86c0006
--- /dev/null
+++ b/call/flexfec_receive_stream.cc
@@ -0,0 +1,21 @@
+/*
+ * Copyright (c) 2018 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.
+ */
+
+#include "call/flexfec_receive_stream.h"
+
+namespace webrtc {
+
+FlexfecReceiveStream::Config::Config(Transport* rtcp_send_transport)
+ : rtcp_send_transport(rtcp_send_transport) {
+ RTC_DCHECK(rtcp_send_transport);
+}
+FlexfecReceiveStream::Config::~Config() = default;
+
+} // namespace webrtc
diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h
index 98ce351..19f945e 100644
--- a/call/flexfec_receive_stream.h
+++ b/call/flexfec_receive_stream.h
@@ -36,10 +36,8 @@
};
struct Config {
- explicit Config(Transport* rtcp_send_transport)
- : rtcp_send_transport(rtcp_send_transport) {
- RTC_DCHECK(rtcp_send_transport);
- }
+ explicit Config(Transport* rtcp_send_transport);
+ ~Config();
std::string ToString() const;
diff --git a/media/BUILD.gn b/media/BUILD.gn
index ffc4883..c2d6149 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -83,6 +83,7 @@
"base/codec.h",
"base/cryptoparams.h",
"base/device.h",
+ "base/mediachannel.cc",
"base/mediachannel.h",
"base/mediaconstants.cc",
"base/mediaconstants.h",
diff --git a/media/base/mediachannel.cc b/media/base/mediachannel.cc
new file mode 100644
index 0000000..019739d
--- /dev/null
+++ b/media/base/mediachannel.cc
@@ -0,0 +1,86 @@
+/*
+ * Copyright (c) 2018 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.
+ */
+
+#include "media/base/mediachannel.h"
+
+namespace cricket {
+
+VideoOptions::VideoOptions() = default;
+VideoOptions::~VideoOptions() = default;
+
+void MediaChannel::SetInterface(NetworkInterface* iface) {
+ rtc::CritScope cs(&network_interface_crit_);
+ network_interface_ = iface;
+ SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
+}
+
+rtc::DiffServCodePoint MediaChannel::PreferredDscp() const {
+ return rtc::DSCP_DEFAULT;
+}
+
+int MediaChannel::GetRtpSendTimeExtnId() const {
+ return -1;
+}
+
+MediaSenderInfo::MediaSenderInfo() = default;
+MediaSenderInfo::~MediaSenderInfo() = default;
+
+MediaReceiverInfo::MediaReceiverInfo() = default;
+MediaReceiverInfo::~MediaReceiverInfo() = default;
+
+VoiceSenderInfo::VoiceSenderInfo() = default;
+VoiceSenderInfo::~VoiceSenderInfo() = default;
+
+VoiceReceiverInfo::VoiceReceiverInfo() = default;
+VoiceReceiverInfo::~VoiceReceiverInfo() = default;
+
+VideoSenderInfo::VideoSenderInfo() = default;
+VideoSenderInfo::~VideoSenderInfo() = default;
+
+VideoReceiverInfo::VideoReceiverInfo() = default;
+VideoReceiverInfo::~VideoReceiverInfo() = default;
+
+VoiceMediaInfo::VoiceMediaInfo() = default;
+VoiceMediaInfo::~VoiceMediaInfo() = default;
+
+VideoMediaInfo::VideoMediaInfo() = default;
+VideoMediaInfo::~VideoMediaInfo() = default;
+
+DataMediaInfo::DataMediaInfo() = default;
+DataMediaInfo::~DataMediaInfo() = default;
+
+AudioSendParameters::AudioSendParameters() = default;
+AudioSendParameters::~AudioSendParameters() = default;
+
+std::map<std::string, std::string> AudioSendParameters::ToStringMap() const {
+ auto params = RtpSendParameters<AudioCodec>::ToStringMap();
+ params["options"] = options.ToString();
+ return params;
+}
+
+VideoSendParameters::VideoSendParameters() = default;
+VideoSendParameters::~VideoSendParameters() = default;
+
+std::map<std::string, std::string> VideoSendParameters::ToStringMap() const {
+ auto params = RtpSendParameters<VideoCodec>::ToStringMap();
+ params["conference_mode"] = (conference_mode ? "yes" : "no");
+ return params;
+}
+
+DataMediaChannel::DataMediaChannel() = default;
+DataMediaChannel::DataMediaChannel(const MediaConfig& config)
+ : MediaChannel(config) {}
+DataMediaChannel::~DataMediaChannel() = default;
+
+bool DataMediaChannel::GetStats(DataMediaInfo* info) {
+ return true;
+}
+
+} // namespace cricket
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 7695079..57cbb3e 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -94,6 +94,9 @@
// We are moving all of the setting of options to structs like this,
// but some things currently still use flags.
struct VideoOptions {
+ VideoOptions();
+ ~VideoOptions();
+
void SetAll(const VideoOptions& change) {
SetFrom(&video_noise_reduction, change.video_noise_reduction);
SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
@@ -176,17 +179,11 @@
explicit MediaChannel(const MediaConfig& config)
: enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
- virtual ~MediaChannel() {}
+ ~MediaChannel() override {}
// Sets the abstract interface class for sending RTP/RTCP data.
- virtual void SetInterface(NetworkInterface *iface) {
- rtc::CritScope cs(&network_interface_crit_);
- network_interface_ = iface;
- SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
- }
- virtual rtc::DiffServCodePoint PreferredDscp() const {
- return rtc::DSCP_DEFAULT;
- }
+ virtual void SetInterface(NetworkInterface* iface);
+ virtual rtc::DiffServCodePoint PreferredDscp() const;
// Called when a RTP packet is received.
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) = 0;
@@ -217,9 +214,7 @@
virtual bool RemoveRecvStream(uint32_t ssrc) = 0;
// Returns the absoulte sendtime extension id value from media channel.
- virtual int GetRtpSendTimeExtnId() const {
- return -1;
- }
+ virtual int GetRtpSendTimeExtnId() const;
// Base method to send packet using NetworkInterface.
bool SendPacket(rtc::CopyOnWriteBuffer* packet,
@@ -294,6 +289,8 @@
};
struct MediaSenderInfo {
+ MediaSenderInfo();
+ ~MediaSenderInfo();
void add_ssrc(const SsrcSenderInfo& stat) {
local_stats.push_back(stat);
}
@@ -339,6 +336,8 @@
};
struct MediaReceiverInfo {
+ MediaReceiverInfo();
+ ~MediaReceiverInfo();
void add_ssrc(const SsrcReceiverInfo& stat) {
local_stats.push_back(stat);
}
@@ -383,6 +382,8 @@
};
struct VoiceSenderInfo : public MediaSenderInfo {
+ VoiceSenderInfo();
+ ~VoiceSenderInfo();
int ext_seqnum = 0;
int jitter_ms = 0;
int audio_level = 0;
@@ -404,6 +405,8 @@
};
struct VoiceReceiverInfo : public MediaReceiverInfo {
+ VoiceReceiverInfo();
+ ~VoiceReceiverInfo();
int ext_seqnum = 0;
int jitter_ms = 0;
int jitter_buffer_ms = 0;
@@ -447,6 +450,8 @@
};
struct VideoSenderInfo : public MediaSenderInfo {
+ VideoSenderInfo();
+ ~VideoSenderInfo();
std::vector<SsrcGroup> ssrc_groups;
// TODO(hbos): Move this to |VideoMediaInfo::send_codecs|?
std::string encoder_implementation_name;
@@ -473,6 +478,8 @@
};
struct VideoReceiverInfo : public MediaReceiverInfo {
+ VideoReceiverInfo();
+ ~VideoReceiverInfo();
std::vector<SsrcGroup> ssrc_groups;
// TODO(hbos): Move this to |VideoMediaInfo::receive_codecs|?
std::string decoder_implementation_name;
@@ -547,6 +554,8 @@
typedef std::map<int, webrtc::RtpCodecParameters> RtpCodecParametersMap;
struct VoiceMediaInfo {
+ VoiceMediaInfo();
+ ~VoiceMediaInfo();
void Clear() {
senders.clear();
receivers.clear();
@@ -560,6 +569,8 @@
};
struct VideoMediaInfo {
+ VideoMediaInfo();
+ ~VideoMediaInfo();
void Clear() {
senders.clear();
receivers.clear();
@@ -577,6 +588,8 @@
};
struct DataMediaInfo {
+ DataMediaInfo();
+ ~DataMediaInfo();
void Clear() {
senders.clear();
receivers.clear();
@@ -636,14 +649,12 @@
};
struct AudioSendParameters : RtpSendParameters<AudioCodec> {
+ AudioSendParameters();
+ ~AudioSendParameters() override;
AudioOptions options;
protected:
- std::map<std::string, std::string> ToStringMap() const override {
- auto params = RtpSendParameters<AudioCodec>::ToStringMap();
- params["options"] = options.ToString();
- return params;
- }
+ std::map<std::string, std::string> ToStringMap() const override;
};
struct AudioRecvParameters : RtpParameters<AudioCodec> {
@@ -654,7 +665,7 @@
VoiceMediaChannel() {}
explicit VoiceMediaChannel(const MediaConfig& config)
: MediaChannel(config) {}
- virtual ~VoiceMediaChannel() {}
+ ~VoiceMediaChannel() override {}
virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
@@ -702,6 +713,8 @@
// TODO(deadbeef): Rename to VideoSenderParameters, since they're intended to
// encapsulate all the parameters needed for a video RtpSender.
struct VideoSendParameters : RtpSendParameters<VideoCodec> {
+ VideoSendParameters();
+ ~VideoSendParameters() override;
// Use conference mode? This flag comes from the remote
// description's SDP line 'a=x-google-flag:conference', copied over
// by VideoChannel::SetRemoteContent_w, and ultimately used by
@@ -711,11 +724,7 @@
bool conference_mode = false;
protected:
- std::map<std::string, std::string> ToStringMap() const override {
- auto params = RtpSendParameters<VideoCodec>::ToStringMap();
- params["conference_mode"] = (conference_mode ? "yes" : "no");
- return params;
- }
+ std::map<std::string, std::string> ToStringMap() const override;
};
// TODO(deadbeef): Rename to VideoReceiverParameters, since they're intended to
@@ -728,7 +737,7 @@
VideoMediaChannel() {}
explicit VideoMediaChannel(const MediaConfig& config)
: MediaChannel(config) {}
- virtual ~VideoMediaChannel() {}
+ ~VideoMediaChannel() override {}
virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
@@ -837,21 +846,21 @@
class DataMediaChannel : public MediaChannel {
public:
- DataMediaChannel() {}
- explicit DataMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
- virtual ~DataMediaChannel() {}
+ DataMediaChannel();
+ explicit DataMediaChannel(const MediaConfig& config);
+ ~DataMediaChannel() override;
virtual bool SetSendParameters(const DataSendParameters& params) = 0;
virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
// TODO(pthatcher): Implement this.
- virtual bool GetStats(DataMediaInfo* info) { return true; }
+ virtual bool GetStats(DataMediaInfo* info);
virtual bool SetSend(bool send) = 0;
virtual bool SetReceive(bool receive) = 0;
- virtual void OnNetworkRouteChanged(const std::string& transport_name,
- const rtc::NetworkRoute& network_route) {}
+ void OnNetworkRouteChanged(const std::string& transport_name,
+ const rtc::NetworkRoute& network_route) override {}
virtual bool SendData(
const SendDataParams& params,
diff --git a/media/base/mediaengine.cc b/media/base/mediaengine.cc
index d85e127..d40f765 100644
--- a/media/base/mediaengine.cc
+++ b/media/base/mediaengine.cc
@@ -12,6 +12,9 @@
namespace cricket {
+RtpCapabilities::RtpCapabilities() = default;
+RtpCapabilities::~RtpCapabilities() = default;
+
webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
webrtc::RtpParameters parameters;
webrtc::RtpEncodingParameters encoding;
diff --git a/media/base/mediaengine.h b/media/base/mediaengine.h
index 920ed85..b832174 100644
--- a/media/base/mediaengine.h
+++ b/media/base/mediaengine.h
@@ -39,6 +39,8 @@
namespace cricket {
struct RtpCapabilities {
+ RtpCapabilities();
+ ~RtpCapabilities();
std::vector<webrtc::RtpExtension> header_extensions;
};
diff --git a/media/base/streamparams.cc b/media/base/streamparams.cc
index 2efa0d0..ac09bfb 100644
--- a/media/base/streamparams.cc
+++ b/media/base/streamparams.cc
@@ -38,6 +38,9 @@
return found != nullptr;
}
+MediaStreams::MediaStreams() = default;
+MediaStreams::~MediaStreams() = default;
+
bool MediaStreams::GetAudioStream(
const StreamSelector& selector, StreamParams* stream) {
return GetStream(audio_, selector, stream);
@@ -100,6 +103,16 @@
return ost.str();
}
+SsrcGroup::SsrcGroup(const std::string& usage,
+ const std::vector<uint32_t>& ssrcs)
+ : semantics(usage), ssrcs(ssrcs) {}
+SsrcGroup::SsrcGroup(const SsrcGroup&) = default;
+SsrcGroup::SsrcGroup(SsrcGroup&&) = default;
+SsrcGroup::~SsrcGroup() = default;
+
+SsrcGroup& SsrcGroup::operator=(const SsrcGroup&) = default;
+SsrcGroup& SsrcGroup::operator=(SsrcGroup&&) = default;
+
bool SsrcGroup::has_semantics(const std::string& semantics_in) const {
return (semantics == semantics_in && ssrcs.size() > 0);
}
@@ -113,6 +126,13 @@
return ost.str();
}
+StreamParams::StreamParams() = default;
+StreamParams::StreamParams(const StreamParams&) = default;
+StreamParams::StreamParams(StreamParams&&) = default;
+StreamParams::~StreamParams() = default;
+StreamParams& StreamParams::operator=(const StreamParams&) = default;
+StreamParams& StreamParams::operator=(StreamParams&&) = default;
+
std::string StreamParams::ToString() const {
std::ostringstream ost;
ost << "{";
diff --git a/media/base/streamparams.h b/media/base/streamparams.h
index 52f1918..6523430 100644
--- a/media/base/streamparams.h
+++ b/media/base/streamparams.h
@@ -43,8 +43,12 @@
extern const char kSimSsrcGroupSemantics[];
struct SsrcGroup {
- SsrcGroup(const std::string& usage, const std::vector<uint32_t>& ssrcs)
- : semantics(usage), ssrcs(ssrcs) {}
+ SsrcGroup(const std::string& usage, const std::vector<uint32_t>& ssrcs);
+ SsrcGroup(const SsrcGroup&);
+ SsrcGroup(SsrcGroup&&);
+ ~SsrcGroup();
+ SsrcGroup& operator=(const SsrcGroup&);
+ SsrcGroup& operator=(SsrcGroup&&);
bool operator==(const SsrcGroup& other) const {
return (semantics == other.semantics && ssrcs == other.ssrcs);
@@ -62,6 +66,13 @@
};
struct StreamParams {
+ StreamParams();
+ StreamParams(const StreamParams&);
+ StreamParams(StreamParams&&);
+ ~StreamParams();
+ StreamParams& operator=(const StreamParams&);
+ StreamParams& operator=(StreamParams&&);
+
static StreamParams CreateLegacy(uint32_t ssrc) {
StreamParams stream;
stream.ssrcs.push_back(ssrc);
@@ -216,7 +227,8 @@
// See https://code.google.com/p/webrtc/issues/detail?id=4107
struct MediaStreams {
public:
- MediaStreams() {}
+ MediaStreams();
+ ~MediaStreams();
void CopyFrom(const MediaStreams& sources);
bool empty() const {
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index ff5a9fb..a08cae7 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -141,11 +141,6 @@
if (rtc_enable_android_aaudio) {
deps += [ ":aaudio_audio_device_jni" ]
}
-
- 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" ]
- }
}
rtc_source_set("audio_device_base_jni") {
@@ -477,14 +472,6 @@
"../../modules/audio_device:audio_device",
"../../modules/audio_processing:audio_processing",
]
-
- if (is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [
- "//build/config/clang:extra_warnings",
- "//build/config/clang:find_bad_constructs",
- ]
- }
}
rtc_static_library("null_media_jni") {
@@ -495,14 +482,6 @@
deps = [
":base_jni",
]
-
- if (is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [
- "//build/config/clang:extra_warnings",
- "//build/config/clang:find_bad_constructs",
- ]
- }
}
generate_jni("generated_peerconnection_jni") {
@@ -1167,14 +1146,6 @@
"native_api/codecs/wrapper.h",
]
- if (is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [
- "//build/config/clang:extra_warnings",
- "//build/config/clang:find_bad_constructs",
- ]
- }
-
deps = [
":base_jni",
":video_jni",
