C++11 in-class member initialization in Call configs.
BUG=
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1166263004.
Cr-Commit-Position: refs/heads/master@{#9416}
diff --git a/webrtc/call.h b/webrtc/call.h
index ac11794..d1a45c8 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -73,54 +73,43 @@
kNetworkDown,
};
struct Config {
+ Config() = delete;
explicit Config(newapi::Transport* send_transport)
- : send_transport(send_transport),
- voice_engine(NULL),
- overuse_callback(NULL) {}
+ : send_transport(send_transport) {}
static const int kDefaultStartBitrateBps;
// TODO(solenberg): Need to add media type to the interface for outgoing
// packets too.
- newapi::Transport* send_transport;
+ newapi::Transport* send_transport = nullptr;
// VoiceEngine used for audio/video synchronization for this Call.
- VoiceEngine* voice_engine;
+ VoiceEngine* voice_engine = nullptr;
// Callback for overuse and normal usage based on the jitter of incoming
- // captured frames. 'NULL' disables the callback.
- LoadObserver* overuse_callback;
+ // captured frames. 'nullptr' disables the callback.
+ LoadObserver* overuse_callback = nullptr;
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used.
struct BitrateConfig {
- BitrateConfig()
- : min_bitrate_bps(0),
- start_bitrate_bps(kDefaultStartBitrateBps),
- max_bitrate_bps(-1) {}
- int min_bitrate_bps;
- int start_bitrate_bps;
- int max_bitrate_bps;
+ int min_bitrate_bps = 0;
+ int start_bitrate_bps = kDefaultStartBitrateBps;
+ int max_bitrate_bps = -1;
} bitrate_config;
struct AudioConfig {
- AudioDeviceModule* audio_device_manager;
- AudioProcessing* audio_processing;
- VoiceEngineObserver* voice_engine_observer;
+ AudioDeviceModule* audio_device_manager = nullptr;
+ AudioProcessing* audio_processing = nullptr;
+ VoiceEngineObserver* voice_engine_observer = nullptr;
} audio_config;
};
struct Stats {
- Stats()
- : send_bandwidth_bps(0),
- recv_bandwidth_bps(0),
- pacer_delay_ms(0),
- rtt_ms(-1) {}
-
- int send_bandwidth_bps;
- int recv_bandwidth_bps;
- int64_t pacer_delay_ms;
- int64_t rtt_ms;
+ int send_bandwidth_bps = 0;
+ int recv_bandwidth_bps = 0;
+ int64_t pacer_delay_ms = 0;
+ int64_t rtt_ms = -1;
};
static Call* Create(const Call::Config& config);
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index 5e5ece5..a8003c1 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -36,32 +36,27 @@
// TODO(mflodman) Move all these settings to VideoDecoder and move the
// declaration to common_types.h.
struct Decoder {
- Decoder()
- : decoder(NULL),
- payload_type(0),
- is_renderer(false),
- expected_delay_ms(0) {}
std::string ToString() const;
// The actual decoder instance.
- VideoDecoder* decoder;
+ VideoDecoder* decoder = nullptr;
// Received RTP packets with this payload type will be sent to this decoder
// instance.
- int payload_type;
+ int payload_type = 0;
// Name of the decoded payload (such as VP8). Maps back to the depacketizer
// used to unpack incoming packets.
std::string payload_name;
// 'true' if the decoder handles rendering as well.
- bool is_renderer;
+ bool is_renderer = false;
// The expected delay for decoding and rendering, i.e. the frame will be
// delivered this many milliseconds, if possible, earlier than the ideal
// render time.
// Note: Ignored if 'renderer' is false.
- int expected_delay_ms;
+ int expected_delay_ms = 0;
};
struct Stats {
@@ -90,13 +85,6 @@
};
struct Config {
- Config()
- : renderer(NULL),
- render_delay_ms(10),
- audio_channel_id(-1),
- pre_decode_callback(NULL),
- pre_render_callback(NULL),
- target_delay_ms(0) {}
std::string ToString() const;
// Decoders for every payload that we can receive.
@@ -104,32 +92,25 @@
// Receive-stream specific RTP settings.
struct Rtp {
- Rtp()
- : remote_ssrc(0),
- local_ssrc(0),
- rtcp_mode(newapi::kRtcpCompound),
- remb(false) {}
std::string ToString() const;
// Synchronization source (stream identifier) to be received.
- uint32_t remote_ssrc;
+ uint32_t remote_ssrc = 0;
// Sender SSRC used for sending RTCP (such as receiver reports).
- uint32_t local_ssrc;
+ uint32_t local_ssrc = 0;
// See RtcpMode for description.
- newapi::RtcpMode rtcp_mode;
+ newapi::RtcpMode rtcp_mode = newapi::kRtcpCompound;
// Extended RTCP settings.
struct RtcpXr {
- RtcpXr() : receiver_reference_time_report(false) {}
-
// True if RTCP Receiver Reference Time Report Block extension
// (RFC 3611) should be enabled.
- bool receiver_reference_time_report;
+ bool receiver_reference_time_report = false;
} rtcp_xr;
// See draft-alvestrand-rmcat-remb for information.
- bool remb;
+ bool remb = false;
// See NackConfig for description.
NackConfig nack;
@@ -140,13 +121,11 @@
// RTX settings for incoming video payloads that may be received. RTX is
// disabled if there's no config present.
struct Rtx {
- Rtx() : ssrc(0), payload_type(0) {}
-
// SSRCs to use for the RTX streams.
- uint32_t ssrc;
+ uint32_t ssrc = 0;
// Payload type to use for the RTX stream.
- int payload_type;
+ int payload_type = 0;
};
// Map from video RTP payload type -> RTX config.
@@ -157,33 +136,33 @@
std::vector<RtpExtension> extensions;
} rtp;
- // VideoRenderer will be called for each decoded frame. 'NULL' disables
+ // VideoRenderer will be called for each decoded frame. 'nullptr' disables
// rendering of this stream.
- VideoRenderer* renderer;
+ VideoRenderer* renderer = nullptr;
// Expected delay needed by the renderer, i.e. the frame will be delivered
// this many milliseconds, if possible, earlier than the ideal render time.
// Only valid if 'renderer' is set.
- int render_delay_ms;
+ int render_delay_ms = 10;
// Audio channel corresponding to this video stream, used for audio/video
// synchronization. 'audio_channel_id' is ignored if no VoiceEngine is set
// when creating the VideoEngine instance. '-1' disables a/v sync.
- int audio_channel_id;
+ int audio_channel_id = -1;
// Called for each incoming video frame, i.e. in encoded state. E.g. used
// when
- // saving the stream to a file. 'NULL' disables the callback.
- EncodedFrameObserver* pre_decode_callback;
+ // saving the stream to a file. 'nullptr' disables the callback.
+ EncodedFrameObserver* pre_decode_callback = nullptr;
// Called for each decoded frame. E.g. used when adding effects to the
// decoded
- // stream. 'NULL' disables the callback.
- I420FrameCallback* pre_render_callback;
+ // stream. 'nullptr' disables the callback.
+ I420FrameCallback* pre_render_callback = nullptr;
// Target delay in milliseconds. A positive value indicates this stream is
// used for streaming instead of a real-time call.
- int target_delay_ms;
+ int target_delay_ms = 0;
};
virtual void Start() = 0;
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index 4dc9a6b..4c4960a 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -52,56 +52,38 @@
};
struct Stats {
- Stats()
- : input_frame_rate(0),
- encode_frame_rate(0),
- avg_encode_time_ms(0),
- encode_usage_percent(0),
- target_media_bitrate_bps(0),
- media_bitrate_bps(0),
- suspended(false) {}
- int input_frame_rate;
- int encode_frame_rate;
- int avg_encode_time_ms;
- int encode_usage_percent;
- int target_media_bitrate_bps;
- int media_bitrate_bps;
- bool suspended;
+ int input_frame_rate = 0;
+ int encode_frame_rate = 0;
+ int avg_encode_time_ms = 0;
+ int encode_usage_percent = 0;
+ int target_media_bitrate_bps = 0;
+ int media_bitrate_bps = 0;
+ bool suspended = false;
std::map<uint32_t, StreamStats> substreams;
};
struct Config {
- Config()
- : pre_encode_callback(NULL),
- post_encode_callback(NULL),
- local_renderer(NULL),
- render_delay_ms(0),
- target_delay_ms(0),
- suspend_below_min_bitrate(false) {}
std::string ToString() const;
struct EncoderSettings {
- EncoderSettings() : payload_type(-1), encoder(NULL) {}
-
std::string ToString() const;
std::string payload_name;
- int payload_type;
+ int payload_type = -1;
// Uninitialized VideoEncoder instance to be used for encoding. Will be
// initialized from inside the VideoSendStream.
- VideoEncoder* encoder;
+ VideoEncoder* encoder = nullptr;
} encoder_settings;
static const size_t kDefaultMaxPacketSize = 1500 - 40; // TCP over IPv4.
struct Rtp {
- Rtp() : max_packet_size(kDefaultMaxPacketSize) {}
std::string ToString() const;
std::vector<uint32_t> ssrcs;
// Max RTP packet size delivered to send transport from VideoEngine.
- size_t max_packet_size;
+ size_t max_packet_size = kDefaultMaxPacketSize;
// RTP header extensions to use for this send stream.
std::vector<RtpExtension> extensions;
@@ -115,13 +97,12 @@
// Settings for RTP retransmission payload format, see RFC 4588 for
// details.
struct Rtx {
- Rtx() : payload_type(-1) {}
std::string ToString() const;
// SSRCs to use for the RTX streams.
std::vector<uint32_t> ssrcs;
// Payload type to use for the RTX stream.
- int payload_type;
+ int payload_type = -1;
} rtx;
// RTCP CNAME, see RFC 3550.
@@ -129,30 +110,30 @@
} rtp;
// Called for each I420 frame before encoding the frame. Can be used for
- // effects, snapshots etc. 'NULL' disables the callback.
- I420FrameCallback* pre_encode_callback;
+ // effects, snapshots etc. 'nullptr' disables the callback.
+ I420FrameCallback* pre_encode_callback = nullptr;
- // Called for each encoded frame, e.g. used for file storage. 'NULL'
+ // Called for each encoded frame, e.g. used for file storage. 'nullptr'
// disables the callback.
- EncodedFrameObserver* post_encode_callback;
+ EncodedFrameObserver* post_encode_callback = nullptr;
// Renderer for local preview. The local renderer will be called even if
- // sending hasn't started. 'NULL' disables local rendering.
- VideoRenderer* local_renderer;
+ // sending hasn't started. 'nullptr' disables local rendering.
+ VideoRenderer* local_renderer = nullptr;
// Expected delay needed by the renderer, i.e. the frame will be delivered
// this many milliseconds, if possible, earlier than expected render time.
// Only valid if |local_renderer| is set.
- int render_delay_ms;
+ int render_delay_ms = 0;
// Target delay in milliseconds. A positive value indicates this stream is
// used for streaming instead of a real-time call.
- int target_delay_ms;
+ int target_delay_ms = 0;
// True if the stream should be suspended when the available bitrate fall
// below the minimum configured bitrate. If this variable is false, the
// stream may send at a rate higher than the estimated available bitrate.
- bool suspend_below_min_bitrate;
+ bool suspend_below_min_bitrate = false;
};
// Gets interface used to insert captured frames. Valid as long as the
