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/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;
